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| author | The Android Open Source Project <initial-contribution@android.com> | 2009-03-03 19:30:38 -0800 |
|---|---|---|
| committer | The Android Open Source Project <initial-contribution@android.com> | 2009-03-03 19:30:38 -0800 |
| commit | 7df30109963092559d3760c0661a020f9daf1030 (patch) | |
| tree | 5780537d7996618fb7e1754bbd27361f3ccb9286 /arm-wt-22k | |
| parent | 1b52a76815aa068a3477bfdfaf58718d41d7f9b4 (diff) | |
| download | external_sonivox-7df30109963092559d3760c0661a020f9daf1030.zip external_sonivox-7df30109963092559d3760c0661a020f9daf1030.tar.gz external_sonivox-7df30109963092559d3760c0661a020f9daf1030.tar.bz2 | |
auto import from //depot/cupcake/@135843
Diffstat (limited to 'arm-wt-22k')
116 files changed, 50475 insertions, 0 deletions
diff --git a/arm-wt-22k/Android.mk b/arm-wt-22k/Android.mk new file mode 100644 index 0000000..71fbcf8 --- /dev/null +++ b/arm-wt-22k/Android.mk @@ -0,0 +1,104 @@ +LOCAL_PATH := $(call my-dir) +include $(CLEAR_VARS) + +LOCAL_SRC_FILES = \ + lib_src/eas_chorus.c \ + lib_src/eas_chorusdata.c \ + lib_src/eas_data.c \ + lib_src/eas_dlssynth.c \ + lib_src/eas_flog.c \ + lib_src/eas_ima_tables.c \ + lib_src/eas_imaadpcm.c \ + lib_src/eas_imelody.c \ + lib_src/eas_imelodydata.c \ + lib_src/eas_math.c \ + lib_src/eas_mdls.c \ + lib_src/eas_midi.c \ + lib_src/eas_mididata.c \ + lib_src/eas_mixbuf.c \ + lib_src/eas_mixer.c \ + lib_src/eas_ota.c \ + lib_src/eas_otadata.c \ + lib_src/eas_pan.c \ + lib_src/eas_pcm.c \ + lib_src/eas_pcmdata.c \ + lib_src/eas_public.c \ + lib_src/eas_reverb.c \ + lib_src/eas_reverbdata.c \ + lib_src/eas_rtttl.c \ + lib_src/eas_rtttldata.c \ + lib_src/eas_smf.c \ + lib_src/eas_smfdata.c \ + lib_src/eas_voicemgt.c \ + lib_src/eas_wtengine.c \ + lib_src/eas_wtsynth.c \ + lib_src/eas_xmf.c \ + lib_src/eas_xmfdata.c \ + lib_src/wt_22khz.c \ + lib_src/jet.c \ + host_src/eas_config.c \ + host_src/eas_hostmm.c \ + host_src/eas_main.c \ + host_src/eas_report.c \ + host_src/eas_wave.c + +# not using these modules +# lib_src/eas_wavefile.c \ +# lib_src/eas_wavefiledata.c \ + +LOCAL_CFLAGS+= -O2 -D UNIFIED_DEBUG_MESSAGES -D EAS_WT_SYNTH \ + -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER \ + -D _XMF_PARSER -D NUM_OUTPUT_CHANNELS=2 \ + -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=64 \ + -D _8_BIT_SAMPLES -D _FILTER_ENABLED \ + -D DLS_SYNTHESIZER \ + -D _REVERB_ENABLED + +# not using these options +# -D _WAVE_PARSER +# -D _IMA_DECODER (needed for IMA-ADPCM wave files) +# -D _CHORUS_ENABLED + +LOCAL_C_INCLUDES:= \ + $(LOCAL_PATH)/host_src \ + $(LOCAL_PATH)/lib_src + +LOCAL_ARM_MODE := arm + +LOCAL_MODULE := libsonivox + +LOCAL_COPY_HEADERS_TO := libsonivox +LOCAL_COPY_HEADERS := \ + host_src/eas.h \ + host_src/eas_types.h \ + host_src/eas_reverb.h \ + host_src/jet.h + +ifeq ($(TARGET_ARCH),arm) +LOCAL_SRC_FILES+= \ + lib_src/ARM-E_filter_gnu.s \ + lib_src/ARM-E_interpolate_loop_gnu.s \ + lib_src/ARM-E_interpolate_noloop_gnu.s \ + lib_src/ARM-E_mastergain_gnu.s \ + lib_src/ARM-E_voice_gain_gnu.s + +asm_flags := \ + -I $(LOCAL_PATH)/lib_src \ + --defsym SAMPLE_RATE_22050=1 \ + --defsym STEREO_OUTPUT=1 \ + --defsym FILTER_ENABLED=1 \ + --defsym SAMPLES_8_BIT=1 + +LOCAL_CFLAGS+= -D NATIVE_EAS_KERNEL \ + $(foreach f,$(asm_flags),-Wa,"$(f)") + +LOCAL_COPY_HEADERS += lib_src/ARM_synth_constants_gnu.inc +endif + +LOCAL_SHARED_LIBRARIES := \ + libutils libcutils + +LOCAL_LDLIBS := -lpthread + +include $(BUILD_SHARED_LIBRARY) + diff --git a/arm-wt-22k/bin/arm-wt-22k b/arm-wt-22k/bin/arm-wt-22k Binary files differnew file mode 100644 index 0000000..2e54640 --- /dev/null +++ b/arm-wt-22k/bin/arm-wt-22k diff --git a/arm-wt-22k/host_src/arm-wt-22k.mak b/arm-wt-22k/host_src/arm-wt-22k.mak new file mode 100644 index 0000000..222d174 --- /dev/null +++ b/arm-wt-22k/host_src/arm-wt-22k.mak @@ -0,0 +1,25 @@ +#
+# Auto-generated sample makefile
+#
+# This makefile is intended for use with GNU make.
+# Set the paths to the tools (CC, AR, LD, etc.)
+#
+
+vpath %.c host_src
+
+CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe
+LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe
+
+%.o: %.c
+ $(CC) -c -O2 -o $@ -I host_src -D UNIFIED_DEBUG_MESSAGES -D EAS_WT_SYNTH -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _XMF_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D MMAPI_SUPPORT -D JET_INTERFACE $<
+
+%.o: %.s
+ $(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $<
+
+OBJS = eas_main.o eas_report.o eas_wave.o eas_hostmm.o eas_config.o
+
+arm-wt-22k: $(OBJS)
+ $(LD) -o $@ $(OBJS) libarm-wt-22k.a -lm
+
diff --git a/arm-wt-22k/host_src/eas.h b/arm-wt-22k/host_src/eas.h new file mode 100644 index 0000000..524601e --- /dev/null +++ b/arm-wt-22k/host_src/eas.h @@ -0,0 +1,1062 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas.h + * + * Contents and purpose: + * The public interface header for the EAS synthesizer. + * + * This header only contains declarations that are specific + * to this implementation. + * + * DO NOT MODIFY THIS FILE! + * + * Copyright Sonic Network Inc. 2005, 2006 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 852 $ + * $Date: 2007-09-04 11:43:49 -0700 (Tue, 04 Sep 2007) $ + *---------------------------------------------------------------------------- +*/ + +#ifndef _EAS_H +#define _EAS_H + +#include "eas_types.h" + +/* for C++ linkage */ +#ifdef __cplusplus +extern "C" { +#endif + +/* library version macro */ +#define MAKE_LIB_VERSION(a,b,c,d) (((((((EAS_U32) a <<8) | (EAS_U32) b) << 8) | (EAS_U32) c) << 8) | (EAS_U32) d) +#define LIB_VERSION MAKE_LIB_VERSION(3, 6, 10, 14) + +typedef struct +{ + EAS_U32 libVersion; + EAS_BOOL checkedVersion; + EAS_I32 maxVoices; + EAS_I32 numChannels; + EAS_I32 sampleRate; + EAS_I32 mixBufferSize; + EAS_BOOL filterEnabled; + EAS_U32 buildTimeStamp; + EAS_CHAR *buildGUID; +} S_EAS_LIB_CONFIG; + +/* enumerated effects module numbers for configuration */ +typedef enum +{ + EAS_MODULE_ENHANCER = 0, + EAS_MODULE_COMPRESSOR, + EAS_MODULE_REVERB, + EAS_MODULE_CHORUS, + EAS_MODULE_WIDENER, + EAS_MODULE_GRAPHIC_EQ, + EAS_MODULE_WOW, + EAS_MODULE_MAXIMIZER, + EAS_MODULE_TONECONTROLEQ, + NUM_EFFECTS_MODULES +} E_FX_MODULES; + +/* enumerated optional module numbers for configuration */ +typedef enum +{ + EAS_MODULE_MMAPI_TONE_CONTROL = 0, + EAS_MODULE_METRICS +} E_OPT_MODULES; +#define NUM_OPTIONAL_MODULES 2 + +/* enumerated audio decoders for configuration */ +typedef enum +{ + EAS_DECODER_PCM = 0, + EAS_DECODER_SMAF_ADPCM, + EAS_DECODER_IMA_ADPCM, + EAS_DECODER_7BIT_SMAF_ADPCM, + EAS_DECODER_NOT_SUPPORTED +} E_DECODER_MODULES; +#define NUM_DECODER_MODULES 4 + +/* defines for EAS_PEOpenStream flags parameter */ +#define PCM_FLAGS_STEREO 0x00000100 /* stream is stereo */ +#define PCM_FLAGS_8_BIT 0x00000001 /* 8-bit format */ +#define PCM_FLAGS_UNSIGNED 0x00000010 /* unsigned format */ +#define PCM_FLAGS_STREAMING 0x80000000 /* streaming mode */ + +/* maximum volume setting */ +#define EAS_MAX_VOLUME 100 + +/*---------------------------------------------------------------------------- + * EAS_Init() + *---------------------------------------------------------------------------- + * Purpose: + * Initialize the synthesizer library + * + * Inputs: + * polyphony - number of voices to play (dynamic memory model only) + * ppLibData - pointer to data handle variable for this instance + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Init (EAS_DATA_HANDLE *ppEASData); + +/*---------------------------------------------------------------------------- + * EAS_Config() + *---------------------------------------------------------------------------- + * Purpose: + * Returns a pointer to a structure containing the configuration options + * in this library build. + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC const S_EAS_LIB_CONFIG *EAS_Config (void); + +/*---------------------------------------------------------------------------- + * EAS_Shutdown() + *---------------------------------------------------------------------------- + * Purpose: + * Shuts down the library. Deallocates any memory associated with the + * synthesizer (dynamic memory model only) + * + * Inputs: + * pEASData - handle to data for this instance + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Shutdown (EAS_DATA_HANDLE pEASData); + +/*---------------------------------------------------------------------------- + * EAS_Render() + *---------------------------------------------------------------------------- + * Purpose: + * Parse the Midi data and render PCM audio data. + * + * Inputs: + * pEASData - buffer for internal EAS data + * pOut - output buffer pointer + * nNumRequested - requested num samples to generate + * pnNumGenerated - actual number of samples generated + * + * Outputs: + * EAS_SUCCESS if PCM data was successfully rendered + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Render (EAS_DATA_HANDLE pEASData, EAS_PCM *pOut, EAS_I32 numRequested, EAS_I32 *pNumGenerated); + +/*---------------------------------------------------------------------------- + * EAS_SetRepeat() + *---------------------------------------------------------------------------- + * Purpose: + * Set the selected stream to repeat. + * + * Inputs: + * pEASData - handle to data for this instance + * streamHandle - handle to stream + * repeatCount - repeat count (0 = no repeat, -1 = repeat forever) + * + * Outputs: + * + * Side Effects: + * + * Notes: + * 0 = no repeat + * 1 = repeat once, i.e. play through twice + * -1 = repeat forever + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 repeatCount); + +/*---------------------------------------------------------------------------- + * EAS_GetRepeat() + *---------------------------------------------------------------------------- + * Purpose: + * Gets the current repeat count for the selected stream. + * + * Inputs: + * pEASData - handle to data for this instance + * streamHandle - handle to stream + * pRrepeatCount - pointer to variable to hold repeat count + * + * Outputs: + * + * Side Effects: + * + * Notes: + * 0 = no repeat + * 1 = repeat once, i.e. play through twice + * -1 = repeat forever + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pRepeatCount); + +/*---------------------------------------------------------------------------- + * EAS_SetPlaybackRate() + *---------------------------------------------------------------------------- + * Purpose: + * Set the playback rate. + * + * Inputs: + * pEASData - handle to data for this instance + * streamHandle - handle to stream + * rate - rate (28-bit fractional amount) + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPlaybackRate (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U32 rate); +#define MAX_PLAYBACK_RATE (EAS_U32)(1L << 29) +#define MIN_PLAYBACK_RATE (EAS_U32)(1L << 27) + +/*---------------------------------------------------------------------------- + * EAS_SetTransposition) + *---------------------------------------------------------------------------- + * Purpose: + * Sets the key tranposition for the synthesizer. Transposes all + * melodic instruments by the specified amount. Range is limited + * to +/-12 semitones. + * + * Inputs: + * pEASData - handle to data for this instance + * streamHandle - handle to stream + * transposition - +/-12 semitones + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetTransposition (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 transposition); +#define MAX_TRANSPOSE 12 + +/*---------------------------------------------------------------------------- + * EAS_SetSynthPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Set the polyphony of the synthesizer. Value must be >= 1 and <= the + * maximum number of voices. This function will pin the polyphony + * at those limits + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * synthNum - synthesizer number (0 = onboard, 1 = DSP) + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 polyphonyCount); + +/*---------------------------------------------------------------------------- + * EAS_GetSynthPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current polyphony setting of the synthesizer + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * synthNum - synthesizer number (0 = onboard, 1 = DSP) + * pPolyphonyCount - pointer to variable to receive polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 *pPolyphonyCount); + +/*---------------------------------------------------------------------------- + * EAS_SetPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Set the polyphony of the stream. Value must be >= 1 and <= the + * maximum number of voices. This function will pin the polyphony + * at those limits + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 polyphonyCount); + +/*---------------------------------------------------------------------------- + * EAS_GetPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current polyphony setting of the stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * pPolyphonyCount - pointer to variable to receive polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPolyphonyCount); + +/*---------------------------------------------------------------------------- + * EAS_SetPriority() + *---------------------------------------------------------------------------- + * Purpose: + * Set the priority of the stream. Determines which stream's voices + * are stolen when there are insufficient voices for all notes. + * Value must be in the range of 1-255, lower values are higher + * priority. The default priority is 50. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 priority); + +/*---------------------------------------------------------------------------- + * EAS_GetPriority() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current priority setting of the stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * pPriority - pointer to variable to receive priority + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPriority); + +/*---------------------------------------------------------------------------- + * EAS_SetVolume() + *---------------------------------------------------------------------------- + * Purpose: + * Set the master volume for the mixer. The default volume setting is + * 90 (-10 dB). The volume range is 0 to 100 in 1dB increments. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * volume - the desired master volume + * + * Outputs: + * + * + * Side Effects: + * overrides any previously set master volume from sysex + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 volume); + +/*---------------------------------------------------------------------------- + * EAS_GetVolume() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the master volume for the mixer in 1dB increments. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * volume - the desired master volume + * + * Outputs: + * + * + * Side Effects: + * overrides any previously set master volume from sysex + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_I32 EAS_GetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_SetMaxLoad() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the maximum workload the parsers will do in a single call to + * EAS_Render. The units are currently arbitrary, but should correlate + * well to the actual CPU cycles consumed. The primary effect is to + * reduce the occasional peaks in CPU cycles consumed when parsing + * dense parts of a MIDI score. Setting maxWorkLoad to zero disables + * the workload limiting function. + * + * Inputs: + * pEASData - handle to data for this instance + * maxLoad - the desired maximum workload + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetMaxLoad (EAS_DATA_HANDLE pEASData, EAS_I32 maxLoad); + +/*---------------------------------------------------------------------------- + * EAS_SetMaxPCMStreams() + *---------------------------------------------------------------------------- + * Sets the maximum number of PCM streams allowed in parsers that + * use PCM streaming. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * maxNumStreams - maximum number of PCM streams + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetMaxPCMStreams (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 maxNumStreams); + +/*---------------------------------------------------------------------------- + * EAS_OpenFile() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a file for audio playback. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * locator - pointer to filename or other locating information + * pStreamHandle - pointer to stream handle variable + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_OpenFile (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *pStreamHandle); + +#ifdef MMAPI_SUPPORT +/*---------------------------------------------------------------------------- + * EAS_MMAPIToneControl() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a ToneControl file for audio playback. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * locator - pointer to filename or other locating information + * pStreamHandle - pointer to stream handle variable + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MMAPIToneControl (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *pStreamHandle); + +/*---------------------------------------------------------------------------- + * EAS_GetWaveFmtChunk + *---------------------------------------------------------------------------- + * Helper function to retrieve WAVE file fmt chunk for MMAPI + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * streamHandle - stream handle + * pFmtChunk - pointer to pointer to FMT chunk data + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetWaveFmtChunk (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_VOID_PTR *ppFmtChunk); +#endif + +/*---------------------------------------------------------------------------- + * EAS_GetFileType + *---------------------------------------------------------------------------- + * Returns the file type (see eas_types.h for enumerations) + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * streamHandle - stream handle + * pFileType - pointer to variable to receive file type + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetFileType (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pFileType); + +/*---------------------------------------------------------------------------- + * EAS_ParseMetaData() + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * playLength - pointer to variable to store the play length (in msecs) + * + * Outputs: + * + * + * Side Effects: + * - resets the parser to the start of the file + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_ParseMetaData (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pPlayLength); + +/*---------------------------------------------------------------------------- + * EAS_Prepare() + *---------------------------------------------------------------------------- + * Purpose: + * Prepares the synthesizer to play the file or stream. Parses the first + * frame of data from the file and arms the synthesizer. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Prepare (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_State() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the state of an audio file or stream. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_State (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_STATE *pState); + +/*---------------------------------------------------------------------------- + * EAS_RegisterMetaDataCallback() + *---------------------------------------------------------------------------- + * Purpose: + * Registers a metadata callback function for parsed metadata. To + * de-register the callback, call this function again with parameter + * cbFunc set to NULL. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * cbFunc - pointer to host callback function + * metaDataBuffer - pointer to metadata buffer + * metaDataBufSize - maximum size of the metadata buffer + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_RegisterMetaDataCallback ( + EAS_DATA_HANDLE pEASData, + EAS_HANDLE streamHandle, + EAS_METADATA_CBFUNC cbFunc, + char *metaDataBuffer, + EAS_I32 metaDataBufSize, + EAS_VOID_PTR pUserData); + +/*---------------------------------------------------------------------------- + * EAS_GetNoteCount () + *---------------------------------------------------------------------------- + * Returns the total number of notes played in this stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * pNoteCount - pointer to variable to receive note count + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetNoteCount (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pNoteCount); + +/*---------------------------------------------------------------------------- + * EAS_CloseFile() + *---------------------------------------------------------------------------- + * Purpose: + * Closes an audio file or stream. Playback should have either paused or + * completed (EAS_State returns EAS_PAUSED or EAS_STOPPED). + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_CloseFile (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_OpenMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a raw MIDI stream allowing the host to route MIDI cable data directly to the synthesizer + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * pStreamHandle - pointer to variable to hold file or stream handle + * streamHandle - open stream or NULL for new synthesizer instance + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_OpenMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE *pStreamHandle, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_WriteMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Send data to the MIDI stream device + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - stream handle + * pBuffer - pointer to buffer + * count - number of bytes to write + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_WriteMIDIStream(EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U8 *pBuffer, EAS_I32 count); + +/*---------------------------------------------------------------------------- + * EAS_CloseMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Closes a raw MIDI stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_CloseMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_Locate() + *---------------------------------------------------------------------------- + * Purpose: + * Locate into the file associated with the handle. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file handle + * milliseconds - playback offset from start of file in milliseconds + * + * Outputs: + * + * + * Side Effects: + * the actual offset will be quantized to the closest update period, typically + * a resolution of 5.9ms. Notes that are started prior to this time will not + * sound. Any notes currently playing will be shut off. + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Locate (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 milliseconds, EAS_BOOL offset); + +/*---------------------------------------------------------------------------- + * EAS_GetRenderTime() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current playback offset + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * Gets the render time clock in msecs. + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetRenderTime (EAS_DATA_HANDLE pEASData, EAS_I32 *pTime); + +/*---------------------------------------------------------------------------- + * EAS_GetLocation() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current playback offset + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file handle + * + * Outputs: + * The offset in milliseconds from the start of the current sequence, quantized + * to the nearest update period. Actual resolution is typically 5.9 ms. + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetLocation (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_I32 *pTime); + +/*---------------------------------------------------------------------------- + * EAS_Pause() + *---------------------------------------------------------------------------- + * Purpose: + * Pauses the playback of the data associated with this handle. The audio + * is gracefully ramped down to prevent clicks and pops. It may take several + * buffers of audio before the audio is muted. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Pause (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_Resume() + *---------------------------------------------------------------------------- + * Purpose: + * Resumes the playback of the data associated with this handle. The audio + * is gracefully ramped up to prevent clicks and pops. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Resume (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle); + +/*---------------------------------------------------------------------------- + * EAS_GetParameter() + *---------------------------------------------------------------------------- + * Purpose: + * Set the parameter of a module. See E_MODULES for a list of modules + * and the header files of the modules for a list of parameters. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * module - enumerated module number + * param - enumerated parameter number + * pValue - pointer to variable to receive parameter value + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 *pValue); + +/*---------------------------------------------------------------------------- + * EAS_SetParameter() + *---------------------------------------------------------------------------- + * Purpose: + * Set the parameter of a module. See E_MODULES for a list of modules + * and the header files of the modules for a list of parameters. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * handle - file or stream handle + * module - enumerated module number + * param - enumerated parameter number + * value - new parameter value + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 value); + +#ifdef _METRICS_ENABLED +/*---------------------------------------------------------------------------- + * EAS_MetricsReport() + *---------------------------------------------------------------------------- + * Purpose: + * Displays the current metrics through the EAS_Report interface. + * + * Inputs: + * pEASData - instance data handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MetricsReport (EAS_DATA_HANDLE pEASData); + +/*---------------------------------------------------------------------------- + * EAS_MetricsReset() + *---------------------------------------------------------------------------- + * Purpose: + * Displays the current metrics through the EAS_Report interface. + * + * Inputs: + * pEASData - instance data handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MetricsReset (EAS_DATA_HANDLE pEASData); +#endif + +/*---------------------------------------------------------------------------- + * EAS_SetSoundLibrary() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the location of the sound library. + * + * Inputs: + * pEASData - instance data handle + * streamHandle - file or stream handle + * pSoundLib - pointer to sound library + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetSoundLibrary (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_SNDLIB_HANDLE pSndLib); + +/*---------------------------------------------------------------------------- + * EAS_SetHeaderSearchFlag() + *---------------------------------------------------------------------------- + * By default, when EAS_OpenFile is called, the parsers check the + * first few bytes of the file looking for a specific header. Some + * mobile devices may add a header to the start of a file, which + * will prevent the parser from recognizing the file. If the + * searchFlag is set to EAS_TRUE, the parser will search the entire + * file looking for the header. This may enable EAS to recognize + * some files that it would ordinarily reject. The negative is that + * it make take slightly longer to process the EAS_OpenFile request. + * + * Inputs: + * pEASData - instance data handle + * searchFlag - search flag (EAS_TRUE or EAS_FALSE) + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetHeaderSearchFlag (EAS_DATA_HANDLE pEASData, EAS_BOOL searchFlag); + +/*---------------------------------------------------------------------------- + * EAS_SetPlayMode() + *---------------------------------------------------------------------------- + * Some file formats support special play modes, such as iMode partial + * play mode. This call can be used to change the play mode. The + * default play mode (usually straight playback) is always zero. + * + * Inputs: + * pEASData - instance data handle + * handle - file or stream handle + * playMode - play mode (see eas_types.h for enumerations) + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPlayMode (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 playMode); + +#ifdef DLS_SYNTHESIZER +/*---------------------------------------------------------------------------- + * EAS_LoadDLSCollection() + *---------------------------------------------------------------------------- + * Purpose: + * Downloads a DLS collection + * + * Inputs: + * pEASData - instance data handle + * streamHandle - file or stream handle + * locator - file locator + * + * Outputs: + * + * + * Side Effects: + * May overlay instruments in the GM sound set + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_LoadDLSCollection (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_FILE_LOCATOR locator); +#endif + +/*---------------------------------------------------------------------------- + * EAS_SetFrameBuffer() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the frame buffer pointer passed to the IPC communications functions + * + * Inputs: + * pEASData - instance data handle + * locator - file locator + * + * Outputs: + * + * + * Side Effects: + * May overlay instruments in the GM sound set + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetFrameBuffer (EAS_DATA_HANDLE pEASData, EAS_FRAME_BUFFER_HANDLE pFrameBuffer); + +#ifdef EXTERNAL_AUDIO +/*---------------------------------------------------------------------------- + * EAS_RegExtAudioCallback() + *---------------------------------------------------------------------------- + * Purpose: + * Registers callback functions for audio events. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * cbProgChgFunc - pointer to host callback function for program change + * cbEventFunc - pointer to host callback functio for note events + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_RegExtAudioCallback (EAS_DATA_HANDLE pEASData, + EAS_HANDLE streamHandle, + EAS_VOID_PTR pInstData, + EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, + EAS_EXT_EVENT_FUNC cbEventFunc); + +/*---------------------------------------------------------------------------- + * EAS_GetMIDIControllers() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current state of MIDI controllers on the requested channel. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - file or stream handle + * pControl - pointer to structure to receive data + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetMIDIControllers (EAS_DATA_HANDLE pEASData, EAS_HANDLE streamHandle, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl); +#endif + +/*---------------------------------------------------------------------------- + * EAS_SearchFile + *---------------------------------------------------------------------------- + * Search file for specific sequence starting at current file + * position. Returns offset to start of sequence. + * + * Inputs: + * pEASData - pointer to EAS persistent data object + * fileHandle - file handle + * searchString - pointer to search sequence + * len - length of search sequence + * pOffset - pointer to variable to store offset to sequence + * + * Returns EAS_EOF if end-of-file is reached + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_SearchFile (EAS_DATA_HANDLE pEASData, EAS_FILE_HANDLE fileHandle, const EAS_U8 *searchString, EAS_I32 len, EAS_I32 *pOffset); + +#ifdef __cplusplus +} /* end extern "C" */ +#endif + +#endif /* #ifndef _EAS_H */ diff --git a/arm-wt-22k/host_src/eas_build.h b/arm-wt-22k/host_src/eas_build.h new file mode 100644 index 0000000..4ff15cc --- /dev/null +++ b/arm-wt-22k/host_src/eas_build.h @@ -0,0 +1,36 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * host_src\eas_build.h
+ *
+ * Contents and purpose:
+ * This file contains the build configuration for this
+ * build. The buildGUIDStr is a GUID created during
+ * the build process and is guaranteed to be unique
+ * for each build.
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ * This file was autogenerated by buildid.exe
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _GUID_1feda229b9a845e996f473c0a80e7220_
+#define _GUID_1feda229b9a845e996f473c0a80e7220_
+
+#define _BUILD_VERSION_ "1feda229-b9a8-45e9-96f4-73c0a80e7220"
+#define _BUILD_TIME_ 0x4743badd
+
+#endif /* _GUID_1feda229b9a845e996f473c0a80e7220_ */
diff --git a/arm-wt-22k/host_src/eas_chorus.h b/arm-wt-22k/host_src/eas_chorus.h new file mode 100644 index 0000000..0e9057f --- /dev/null +++ b/arm-wt-22k/host_src/eas_chorus.h @@ -0,0 +1,53 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorus.h
+ *
+ * Contents and purpose:
+ * Contains parameter enumerations for the Chorus effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 309 $
+ * $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_CHORUS_H
+#define EAS_CHORUS_H
+
+/* enumerated parameter settings for Chorus effect */
+typedef enum
+{
+ EAS_PARAM_CHORUS_BYPASS,
+ EAS_PARAM_CHORUS_PRESET,
+ EAS_PARAM_CHORUS_RATE,
+ EAS_PARAM_CHORUS_DEPTH,
+ EAS_PARAM_CHORUS_LEVEL
+} E_CHORUS_PARAMS;
+
+typedef enum
+{
+ EAS_PARAM_CHORUS_PRESET1,
+ EAS_PARAM_CHORUS_PRESET2,
+ EAS_PARAM_CHORUS_PRESET3,
+ EAS_PARAM_CHORUS_PRESET4
+} E_CHORUS_PRESETS;
+
+
+#endif
\ No newline at end of file diff --git a/arm-wt-22k/host_src/eas_config.c b/arm-wt-22k/host_src/eas_config.c new file mode 100644 index 0000000..c45fbb7 --- /dev/null +++ b/arm-wt-22k/host_src/eas_config.c @@ -0,0 +1,619 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_config.c
+ *
+ * Contents and purpose:
+ * This file contains the Configuration Module interface (CM). The CM
+ * is a module compiled external to the library that sets the configuration
+ * for this build. It allows the library to find optional components and
+ * links to static memory allocations (when used in a static configuration).
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * NOTE: This module is not intended to be modified by the customer. It
+ * needs to be included in the build process with the correct configuration
+ * defines (see the library documentation for information on how to configure
+ * the library).
+ *
+ * Copyright Sonic Network Inc. 2004-2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 796 $
+ * $Date: 2007-08-01 00:15:25 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas.h"
+#include "eas_config.h"
+
+
+#ifdef _MFI_PARSER
+/*----------------------------------------------------------------------------
+ * Vendor/Device ID for MFi Extensions
+ *
+ * Define the preprocessor symbols to establish the vendor ID and
+ * device ID for the MFi PCM/ADPCM extensions.
+ *----------------------------------------------------------------------------
+*/
+const EAS_U8 eas_MFIVendorIDMSB = (MFI_VENDOR_ID >> 8) & 0xff;
+const EAS_U8 eas_MFIVendorIDLSB = MFI_VENDOR_ID & 0xff;
+const EAS_U8 eas_MFIDeviceID = MFI_DEVICE_ID;
+#endif
+
+/*----------------------------------------------------------------------------
+ *
+ * parserModules
+ *
+ * This structure is used by the EAS library to locate file parsing
+ * modules.
+ *----------------------------------------------------------------------------
+*/
+
+/* define the external file parsers */
+extern EAS_VOID_PTR EAS_SMF_Parser;
+
+#ifdef _XMF_PARSER
+extern EAS_VOID_PTR EAS_XMF_Parser;
+#endif
+
+#ifdef _SMAF_PARSER
+extern EAS_VOID_PTR EAS_SMAF_Parser;
+#endif
+
+#ifdef _WAVE_PARSER
+extern EAS_VOID_PTR EAS_Wave_Parser;
+#endif
+
+#ifdef _OTA_PARSER
+extern EAS_VOID_PTR EAS_OTA_Parser;
+#endif
+
+#ifdef _IMELODY_PARSER
+extern EAS_VOID_PTR EAS_iMelody_Parser;
+#endif
+
+#ifdef _RTTTL_PARSER
+extern EAS_VOID_PTR EAS_RTTTL_Parser;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+extern EAS_VOID_PTR EAS_CMF_Parser;
+#endif
+
+/* initalize pointers to parser interfaces */
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const parserModules[] =
+{
+ &EAS_SMF_Parser,
+
+#ifdef _XMF_PARSER
+ &EAS_XMF_Parser,
+#endif
+
+#ifdef _WAVE_PARSER
+ &EAS_Wave_Parser,
+#endif
+
+#ifdef _SMAF_PARSER
+ &EAS_SMAF_Parser,
+#endif
+
+#ifdef _OTA_PARSER
+ &EAS_OTA_Parser,
+#endif
+
+#ifdef _IMELODY_PARSER
+ &EAS_iMelody_Parser,
+#endif
+
+#ifdef _RTTTL_PARSER
+ &EAS_RTTTL_Parser,
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+ &EAS_CMF_Parser
+#endif
+};
+#define NUM_PARSER_MODULES (sizeof(parserModules) / sizeof(EAS_VOID_PTR))
+
+/*----------------------------------------------------------------------------
+ * Data Modules
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_SMFData;
+extern EAS_VOID_PTR eas_Data;
+extern EAS_VOID_PTR eas_MixBuffer;
+extern EAS_VOID_PTR eas_Synth;
+extern EAS_VOID_PTR eas_MIDI;
+extern EAS_VOID_PTR eas_PCMData;
+extern EAS_VOID_PTR eas_MIDIData;
+
+#ifdef _XMF_PARSER
+extern EAS_VOID_PTR eas_XMFData;
+#endif
+
+#ifdef _SMAF_PARSER
+extern EAS_VOID_PTR eas_SMAFData;
+#endif
+
+#ifdef _OTA_PARSER
+extern EAS_VOID_PTR eas_OTAData;
+#endif
+
+#ifdef _IMELODY_PARSER
+extern EAS_VOID_PTR eas_iMelodyData;
+#endif
+
+#ifdef _RTTTL_PARSER
+extern EAS_VOID_PTR eas_RTTTLData;
+#endif
+
+#ifdef _WAVE_PARSER
+extern EAS_VOID_PTR eas_WaveData;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+extern EAS_VOID_PTR eas_CMFData;
+#endif
+#endif
+
+/*----------------------------------------------------------------------------
+ *
+ * Effects Modules
+ *
+ * These declarations are used by the EAS library to locate
+ * effects modules.
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _ENHANCER_ENABLED
+extern EAS_VOID_PTR EAS_Enhancer;
+#define EAS_ENHANCER_INTERFACE &EAS_Enhancer
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_EnhancerData;
+#define EAS_ENHANCER_DATA &eas_EnhancerData
+#else
+#define EAS_ENHANCER_DATA NULL
+#endif
+#else
+#define EAS_ENHANCER_INTERFACE NULL
+#define EAS_ENHANCER_DATA NULL
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+extern EAS_VOID_PTR EAS_Compressor;
+#define EAS_COMPRESSOR_INTERFACE &EAS_Compressor
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_CompressorData;
+#define EAS_COMPRESSOR_DATA &eas_CompressorData
+#else
+#define EAS_COMPRESSOR_DATA NULL
+#endif
+#else
+#define EAS_COMPRESSOR_INTERFACE NULL
+#define EAS_COMPRESSOR_DATA NULL
+#endif
+
+#ifdef _MAXIMIZER_ENABLED
+extern EAS_VOID_PTR EAS_Maximizer;
+#define EAS_MAXIMIZER_INTERFACE &EAS_Maximizer
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_MaximizerData;
+#define EAS_MAXIMIZER_DATA &eas_MaximizerData
+#else
+#define EAS_MAXIMIZER_DATA NULL
+#endif
+#else
+#define EAS_MAXIMIZER_INTERFACE NULL
+#define EAS_MAXIMIZER_DATA NULL
+#endif
+
+
+#ifdef _REVERB_ENABLED
+extern EAS_VOID_PTR EAS_Reverb;
+#define EAS_REVERB_INTERFACE &EAS_Reverb
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_ReverbData;
+#define EAS_REVERB_DATA &eas_ReverbData
+#else
+#define EAS_REVERB_DATA NULL
+#endif
+#else
+#define EAS_REVERB_INTERFACE NULL
+#define EAS_REVERB_DATA NULL
+#endif
+
+#ifdef _CHORUS_ENABLED
+extern EAS_VOID_PTR EAS_Chorus;
+#define EAS_CHORUS_INTERFACE &EAS_Chorus
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_ChorusData;
+#define EAS_CHORUS_DATA &eas_ChorusData
+#else
+#define EAS_CHORUS_DATA NULL
+#endif
+#else
+#define EAS_CHORUS_INTERFACE NULL
+#define EAS_CHORUS_DATA NULL
+#endif
+
+#ifdef _WIDENER_ENABLED
+extern EAS_VOID_PTR EAS_Widener;
+#define EAS_WIDENER_INTERFACE &EAS_Widener
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_WidenerData;
+#define EAS_WIDENER_DATA &eas_WidenerData
+#else
+#define EAS_WIDENER_DATA NULL
+#endif
+#else
+#define EAS_WIDENER_INTERFACE NULL
+#define EAS_WIDENER_DATA NULL
+#endif
+
+#ifdef _GRAPHIC_EQ_ENABLED
+extern EAS_VOID_PTR EAS_GraphicEQ;
+#define EAS_GRAPHIC_EQ_INTERFACE &EAS_GraphicEQ
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_GraphicEQData;
+#define EAS_GRAPHIC_EQ_DATA &eas_GraphicEQData
+#else
+#define EAS_GRAPHIC_EQ_DATA NULL
+#endif
+#else
+#define EAS_GRAPHIC_EQ_INTERFACE NULL
+#define EAS_GRAPHIC_EQ_DATA NULL
+#endif
+
+#ifdef _WOW_ENABLED
+extern EAS_VOID_PTR EAS_Wow;
+#define EAS_WOW_INTERFACE &EAS_Wow
+#ifdef _STATIC_MEMORY
+#error "WOW module requires dynamic memory model"
+#else
+#define EAS_WOW_DATA NULL
+#endif
+#else
+#define EAS_WOW_INTERFACE NULL
+#define EAS_WOW_DATA NULL
+#endif
+
+#ifdef _TONECONTROLEQ_ENABLED
+extern EAS_VOID_PTR EAS_ToneControlEQ;
+#define EAS_TONECONTROLEQ_INTERFACE &EAS_ToneControlEQ
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_ToneControlEQData;
+#define EAS_TONECONTROLEQ_DATA &eas_ToneControlEQData
+#else
+#define EAS_TONECONTROLEQ_DATA NULL
+#endif
+#else
+#define EAS_TONECONTROLEQ_INTERFACE NULL
+#define EAS_TONECONTROLEQ_DATA NULL
+#endif
+
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const effectsModules[] =
+{
+ EAS_ENHANCER_INTERFACE,
+ EAS_COMPRESSOR_INTERFACE,
+ EAS_REVERB_INTERFACE,
+ EAS_CHORUS_INTERFACE,
+ EAS_WIDENER_INTERFACE,
+ EAS_GRAPHIC_EQ_INTERFACE,
+ EAS_WOW_INTERFACE,
+ EAS_MAXIMIZER_INTERFACE,
+ EAS_TONECONTROLEQ_INTERFACE
+};
+
+EAS_VOID_PTR const effectsData[] =
+{
+ EAS_ENHANCER_DATA,
+ EAS_COMPRESSOR_DATA,
+ EAS_REVERB_DATA,
+ EAS_CHORUS_DATA,
+ EAS_WIDENER_DATA,
+ EAS_GRAPHIC_EQ_DATA,
+ EAS_WOW_DATA,
+ EAS_MAXIMIZER_DATA,
+ EAS_TONECONTROLEQ_DATA
+};
+
+/*----------------------------------------------------------------------------
+ *
+ * Optional Modules
+ *
+ * These declarations are used by the EAS library to locate
+ * effects modules.
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _METRICS_ENABLED
+extern EAS_VOID_PTR EAS_Metrics;
+#define EAS_METRICS_INTERFACE &EAS_Metrics
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_MetricsData;
+#define EAS_METRICS_DATA &eas_MetricsData
+#else
+#define EAS_METRICS_DATA NULL
+#endif
+#else
+#define EAS_METRICS_INTERFACE NULL
+#define EAS_METRICS_DATA NULL
+#endif
+
+#ifdef MMAPI_SUPPORT
+extern EAS_VOID_PTR EAS_TC_Parser;
+#define EAS_TONE_CONTROL_PARSER &EAS_TC_Parser
+#ifdef _STATIC_MEMORY
+extern EAS_VOID_PTR eas_TCData;
+#define EAS_TONE_CONTROL_DATA &eas_TCData
+#else
+#define EAS_TONE_CONTROL_DATA NULL
+#endif
+#else
+#define EAS_TONE_CONTROL_PARSER NULL
+#define EAS_TONE_CONTROL_DATA NULL
+#endif
+
+/*lint -e{605} not pretty, but it works */
+EAS_VOID_PTR const optionalModules[] =
+{
+ EAS_TONE_CONTROL_PARSER,
+ EAS_METRICS_INTERFACE
+};
+
+EAS_VOID_PTR const optionalData[] =
+{
+ EAS_TONE_CONTROL_DATA,
+ EAS_METRICS_DATA
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_CMStaticMemoryModel()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function returns true if EAS has been configured for
+ * a static memory model. There are some limitations in the
+ * static memory model, see the documentation for more
+ * information.
+ *
+ * Outputs:
+ * returns EAS_TRUE if a module is found
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL EAS_CMStaticMemoryModel (void)
+{
+#ifdef _STATIC_MEMORY
+ return EAS_TRUE;
+#else
+ return EAS_FALSE;
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ * module - module number
+ *
+ * Outputs:
+ * returns a pointer to the module function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module)
+{
+
+ if (module >= (EAS_INT) NUM_PARSER_MODULES)
+ return NULL;
+ return parserModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, dataModule) used only when _STATIC_MEMORY is defined */
+EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule)
+{
+
+#ifdef _STATIC_MEMORY
+ switch (dataModule)
+ {
+
+ /* main instance data for synthesizer */
+ case EAS_CM_EAS_DATA:
+ return &eas_Data;
+
+ /* mix buffer for mix engine */
+ case EAS_CM_MIX_BUFFER:
+ /*lint -e{545} lint doesn't like this because it sees the underlying type */
+ return &eas_MixBuffer;
+
+ /* instance data for synth */
+ case EAS_CM_SYNTH_DATA:
+ return &eas_Synth;
+
+ /* instance data for MIDI parser */
+ case EAS_CM_MIDI_DATA:
+ return &eas_MIDI;
+
+ /* instance data for SMF parser */
+ case EAS_CM_SMF_DATA:
+ return &eas_SMFData;
+
+#ifdef _XMF_PARSER
+ /* instance data for XMF parser */
+ case EAS_CM_XMF_DATA:
+ return &eas_XMFData;
+#endif
+
+#ifdef _SMAF_PARSER
+ /* instance data for SMAF parser */
+ case EAS_CM_SMAF_DATA:
+ return &eas_SMAFData;
+#endif
+
+ /* instance data for the PCM engine */
+ case EAS_CM_PCM_DATA:
+ /*lint -e{545} lint doesn't like this because it sees the underlying type */
+ return &eas_PCMData;
+
+ case EAS_CM_MIDI_STREAM_DATA:
+ return &eas_MIDIData;
+
+#ifdef _OTA_PARSER
+ /* instance data for OTA parser */
+ case EAS_CM_OTA_DATA:
+ return &eas_OTAData;
+#endif
+
+#ifdef _IMELODY_PARSER
+ /* instance data for iMelody parser */
+ case EAS_CM_IMELODY_DATA:
+ return &eas_iMelodyData;
+#endif
+
+#ifdef _RTTTL_PARSER
+ /* instance data for RTTTL parser */
+ case EAS_CM_RTTTL_DATA:
+ return &eas_RTTTLData;
+#endif
+
+#ifdef _WAVE_PARSER
+ /* instance data for WAVE parser */
+ case EAS_CM_WAVE_DATA:
+ return &eas_WaveData;
+#endif
+
+#if defined (_CMX_PARSER) || defined(_MFI_PARSER)
+ /* instance data for CMF parser */
+ case EAS_CM_CMF_DATA:
+ return &eas_CMFData;
+#endif
+
+ default:
+ return NULL;
+ }
+
+#else
+ return NULL;
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional effects modules.
+ *
+ * Inputs:
+ * module - enumerated module number
+ * pModule - pointer to module interface
+ *
+ * Outputs:
+ * Returns pointer to function table or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module)
+{
+
+ if (module >= NUM_EFFECTS_MODULES)
+ return NULL;
+ return effectsModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ * pData - pointer to handle variable
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule)
+{
+
+ if (dataModule >= NUM_EFFECTS_MODULES)
+ return NULL;
+ return effectsData[dataModule];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ * module - enumerated module number
+ *
+ * Outputs:
+ * returns pointer to function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module)
+{
+
+ /* sanity check */
+ if (module >= NUM_OPTIONAL_MODULES)
+ return EAS_FALSE;
+ return optionalModules[module];
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule)
+{
+
+ if (dataModule >= NUM_OPTIONAL_MODULES)
+ return NULL;
+ return optionalData[dataModule];
+}
+
+
diff --git a/arm-wt-22k/host_src/eas_config.h b/arm-wt-22k/host_src/eas_config.h new file mode 100644 index 0000000..d16be4a --- /dev/null +++ b/arm-wt-22k/host_src/eas_config.h @@ -0,0 +1,191 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_config.h
+ *
+ * Contents and purpose:
+ * This header declares the Configuration Module interface (CM). The CM
+ * is a module compiled external to the library that sets the configuration
+ * for this build. It allows the library to find optional components and
+ * links to static memory allocations (when used in a static configuration).
+ *
+ * NOTE: This module is not intended to be modified by the customer. It
+ * needs to be included in the build process with the correct configuration
+ * defines (see the library documentation for information on how to configure
+ * the library).
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// sentinel
+#ifndef _EAS_CONFIG_H
+#define _EAS_CONFIG_H
+
+#include "eas_types.h"
+
+/* list of enumerators for optional modules */
+typedef enum {
+ EAS_CM_FILE_PARSERS = 1
+} E_CM_ENUM_MODULES;
+
+/* list of enumerators for module and memory pointers */
+typedef enum {
+ EAS_CM_EAS_DATA = 1,
+ EAS_CM_MIX_BUFFER,
+ EAS_CM_SYNTH_DATA,
+ EAS_CM_MIDI_DATA,
+ EAS_CM_SMF_DATA,
+ EAS_CM_XMF_DATA,
+ EAS_CM_SMAF_DATA,
+ EAS_CM_PCM_DATA,
+ EAS_CM_MIDI_STREAM_DATA,
+ EAS_CM_METRICS_DATA,
+ EAS_CM_OTA_DATA,
+ EAS_CM_IMELODY_DATA,
+ EAS_CM_RTTTL_DATA,
+ EAS_CM_WAVE_DATA,
+ EAS_CM_CMF_DATA
+} E_CM_DATA_MODULES;
+
+typedef struct
+{
+ int maxSMFStreams;
+ void *pSMFData;
+ void *pSMFStream;
+} S_EAS_SMF_PTRS;
+
+typedef struct
+{
+ int maxSMAFStreams;
+ void *pSMAFData;
+ void *pSMAFStream;
+} S_EAS_SMAF_PTRS;
+
+/*----------------------------------------------------------------------------
+ * EAS_CMStaticMemoryModel()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function returns true if EAS has been configured for
+ * a static memory model. There are some limitations in the
+ * static memory model, see the documentation for more
+ * information.
+ *
+ * Outputs:
+ * returns EAS_TRUE if a module is found
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL EAS_CMStaticMemoryModel (void);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ * module - module number
+ *
+ * Outputs:
+ * returns a pointer to the module function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumData (EAS_INT dataModule);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional effects modules.
+ *
+ * Inputs:
+ * module - enumerated module number
+ * pModule - pointer to module interface
+ *
+ * Outputs:
+ * Returns pointer to function table or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumFXData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ * pData - pointer to handle variable
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumFXData (EAS_INT dataModule);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptModules()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to optional modules.
+ *
+ * Inputs:
+ * module - enumerated module number
+ *
+ * Outputs:
+ * returns pointer to function table or NULL if no module
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptModules (EAS_INT module);
+
+/*----------------------------------------------------------------------------
+ * EAS_CMEnumOptData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function is used to find pointers to static memory allocations.
+ *
+ * Inputs:
+ * dataModule - enumerated module number
+ *
+ * Outputs:
+ * Returns handle to data or NULL if not found
+ *----------------------------------------------------------------------------
+*/
+EAS_VOID_PTR EAS_CMEnumOptData (EAS_INT dataModule);
+
+#endif /* end _EAS_CONFIG_H */
diff --git a/arm-wt-22k/host_src/eas_debugmsgs.h b/arm-wt-22k/host_src/eas_debugmsgs.h new file mode 100644 index 0000000..25913bd --- /dev/null +++ b/arm-wt-22k/host_src/eas_debugmsgs.h @@ -0,0 +1,87 @@ +/* Auto-generated from source file: eas_chorusdata.c */ +/* Auto-generated from source file: eas_xmfdata.c */ +/* Auto-generated from source file: eas_mdls.c */ +{ 0x19299ed4, 0x00000022, "eas_mdls.c[2561]: #Instruments: %u\n" }, +{ 0x19299ed4, 0x00000023, "eas_mdls.c[2562]: #Regions: %u\n" }, +{ 0x19299ed4, 0x00000024, "eas_mdls.c[2563]: #Articulations: %u\n" }, +{ 0x19299ed4, 0x00000025, "eas_mdls.c[2564]: #Waves: %u\n" }, +{ 0x19299ed4, 0x00000026, "eas_mdls.c[2569]: Locale: %lu:%lu:%lu\n" }, +{ 0x19299ed4, 0x00000027, "eas_mdls.c[2576]: \tRegion: %u\n" }, +{ 0x19299ed4, 0x00000028, "eas_mdls.c[2578]: \t\tm_nGain: %d\n" }, +{ 0x19299ed4, 0x00000029, "eas_mdls.c[2579]: \t\trangeLow:rangeHigh %u:%u\n" }, +{ 0x19299ed4, 0x0000002a, "eas_mdls.c[2580]: \t\tm_nKeyGroupAndFlags: %04x\n" }, +{ 0x19299ed4, 0x0000002b, "eas_mdls.c[2581]: \t\tm_nLoopStart: %u\n" }, +{ 0x19299ed4, 0x0000002c, "eas_mdls.c[2582]: \t\tm_nLoopEnd: %u\n" }, +{ 0x19299ed4, 0x0000002d, "eas_mdls.c[2583]: \t\tm_nNetTuningInCents: %d\n" }, +{ 0x19299ed4, 0x0000002e, "eas_mdls.c[2584]: \t\tArt Index: %u\n" }, +{ 0x19299ed4, 0x0000002f, "eas_mdls.c[2585]: \t\tWave Index: %u\n" }, +{ 0x19299ed4, 0x00000030, "eas_mdls.c[2597]: Articulation: %u\n" }, +{ 0x19299ed4, 0x00000031, "eas_mdls.c[2599]: \t\tm_nEG2toFilterDepth: %d\n" }, +{ 0x19299ed4, 0x00000032, "eas_mdls.c[2600]: \t\tm_nEG2toPitchDepth: %d\n" }, +{ 0x19299ed4, 0x00000033, "eas_mdls.c[2601]: \t\tm_nFilterCutoffFrequency: %d\n" }, +{ 0x19299ed4, 0x00000034, "eas_mdls.c[2602]: \t\tm_nFilterResonance: %u\n" }, +{ 0x19299ed4, 0x00000035, "eas_mdls.c[2603]: \t\tm_nLFOAmplitudeDepth: %d\n" }, +{ 0x19299ed4, 0x00000036, "eas_mdls.c[2604]: \t\tm_nLFODelayTime: %d\n" }, +{ 0x19299ed4, 0x00000037, "eas_mdls.c[2605]: \t\tm_nLFOFrequency: %d\n" }, +{ 0x19299ed4, 0x00000038, "eas_mdls.c[2606]: \t\tm_nLFOPitchDepth: %d\n" }, +{ 0x19299ed4, 0x00000039, "eas_mdls.c[2607]: \t\tm_nPan: %d\n" }, +{ 0x19299ed4, 0x0000003a, "eas_mdls.c[2610]: \t\tm_nAttack: %d\n" }, +{ 0x19299ed4, 0x0000003b, "eas_mdls.c[2611]: \t\tm_nDecay: %d\n" }, +{ 0x19299ed4, 0x0000003c, "eas_mdls.c[2612]: \t\tm_nSustain: %d\n" }, +{ 0x19299ed4, 0x0000003d, "eas_mdls.c[2613]: \t\tm_nRelease: %d\n" }, +{ 0x19299ed4, 0x0000003e, "eas_mdls.c[2616]: \t\tm_nAttack: %d\n" }, +{ 0x19299ed4, 0x0000003f, "eas_mdls.c[2617]: \t\tm_nDecay: %d\n" }, +{ 0x19299ed4, 0x00000040, "eas_mdls.c[2618]: \t\tm_nSustain: %d\n" }, +{ 0x19299ed4, 0x00000041, "eas_mdls.c[2619]: \t\tm_nRelease: %d\n" }, +{ 0x19299ed4, 0x00000042, "eas_mdls.c[2626]: Wave Index: %d\n" }, +{ 0x19299ed4, 0x00000043, "eas_mdls.c[2627]: \tSize %u\n" }, +{ 0x19299ed4, 0x00000044, "eas_mdls.c[2628]: \tPointer %08lx\n" }, +/* Auto-generated from source file: eas_mididata.c */ +/* Auto-generated from source file: eas_pan.c */ +/* Auto-generated from source file: eas_wavefiledata.c */ +/* Auto-generated from source file: eas_reverbdata.c */ +/* Auto-generated from source file: eas_imelodydata.c */ +/* Auto-generated from source file: eas_ota.c */ +/* Auto-generated from source file: eas_mixbuf.c */ +/* Auto-generated from source file: eas_tcdata.c */ +/* Auto-generated from source file: jet.c */ +/* Auto-generated from source file: eas_rtttl.c */ +/* Auto-generated from source file: eas_reverb.c */ +/* Auto-generated from source file: eas_pcmdata.c */ +/* Auto-generated from source file: eas_chorus.c */ +/* Auto-generated from source file: eas_math.c */ +/* Auto-generated from source file: eas_xmf.c */ +/* Auto-generated from source file: eas_smfdata.c */ +/* Auto-generated from source file: eas_imelody.c */ +/* Auto-generated from source file: eas_dlssynth.c */ +/* Auto-generated from source file: eas_public.c */ +/* Auto-generated from source file: eas_rtttldata.c */ +/* Auto-generated from source file: eas_voicemgt.c */ +/* Auto-generated from source file: eas_tonecontrol.c */ +/* Auto-generated from source file: eas_wtengine.c */ +/* Auto-generated from source file: eas_imaadpcm.c */ +{ 0x2380b977, 0x00000006, "eas_imaadpcm.c[305]: IMADecoderLocate: Time=%d, samples=%d\n" }, +{ 0x2380b977, 0x00000007, "eas_imaadpcm.c[328]: IMADecoderLocate: Looped sample, numBlocks=%d, samplesPerLoop=%d, samplesInLastBlock=%d, samples=%d\n" }, +{ 0x2380b977, 0x00000008, "eas_imaadpcm.c[335]: IMADecoderLocate: Byte location in audio = %d\n" }, +{ 0x2380b977, 0x00000009, "eas_imaadpcm.c[345]: IMADecoderLocate: bytesLeft = %d\n" }, +/* Auto-generated from source file: eas_flog.c */ +/* Auto-generated from source file: eas_midi.c */ +/* Auto-generated from source file: eas_otadata.c */ +/* Auto-generated from source file: eas_ima_tables.c */ +/* Auto-generated from source file: eas_data.c */ +/* Auto-generated from source file: eas_pcm.c */ +/* Auto-generated from source file: eas_mixer.c */ +/* Auto-generated from source file: eas_wavefile.c */ +/* Auto-generated from source file: eas_wtsynth.c */ +/* Auto-generated from source file: eas_smf.c */ +/* Auto-generated from source file: eas_wave.c */ +/* Auto-generated from source file: eas_hostmm.c */ +{ 0x1a54b6e8, 0x00000001, "eas_hostmm.c[586]: Vibrate state: %d\n" }, +{ 0x1a54b6e8, 0x00000002, "eas_hostmm.c[601]: LED state: %d\n" }, +{ 0x1a54b6e8, 0x00000003, "eas_hostmm.c[616]: Backlight state: %d\n" }, +{ 0x1a54b6e8, 0x00000004, "eas_hostmm.c[162]: HWMemCpy: bad amount: %d\n" }, +{ 0x1a54b6e8, 0x00000005, "eas_hostmm.c[179]: HWMemSet: bad amount: %d\n" }, +{ 0x1a54b6e8, 0x00000006, "eas_hostmm.c[196]: HWMemCmp: bad amount: %d\n" }, +/* Auto-generated from source file: eas_config.c */ +/* Auto-generated from source file: eas_main.c */ +{ 0xe624f4d9, 0x00000005, "eas_main.c[106]: Play length: %d.%03d (secs)\n" }, diff --git a/arm-wt-22k/host_src/eas_host.h b/arm-wt-22k/host_src/eas_host.h new file mode 100644 index 0000000..270e68c --- /dev/null +++ b/arm-wt-22k/host_src/eas_host.h @@ -0,0 +1,87 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_host.h
+ *
+ * Contents and purpose:
+ * This header defines the host wrapper functions for stdio, stdlib, etc.
+ * The host application must provide an abstraction layer for these functions
+ * to support certain features, such as SMAF and SMF-1 conversion.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// sentinel
+#ifndef _EAS_HOST_H
+#define _EAS_HOST_H
+
+#include "eas_types.h"
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* initialization and shutdown routines */
+extern EAS_RESULT EAS_HWInit(EAS_HW_DATA_HANDLE *hwInstData);
+extern EAS_RESULT EAS_HWShutdown(EAS_HW_DATA_HANDLE hwInstData);
+
+/* threading */
+extern void* EAS_HWRegisterSignalHandler();
+extern EAS_RESULT EAS_HWUnRegisterSignalHandler(void *cookie);
+
+/* memory functions */
+extern void *EAS_HWMemSet(void *s, int c, EAS_I32 n);
+extern void *EAS_HWMemCpy(void *s1, const void *s2, EAS_I32 n);
+extern EAS_I32 EAS_HWMemCmp(const void *s1, const void *s2, EAS_I32 n);
+
+/* memory allocation */
+extern void *EAS_HWMalloc(EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size);
+extern void EAS_HWFree(EAS_HW_DATA_HANDLE hwInstData, void *p);
+
+/* file I/O */
+extern EAS_RESULT EAS_HWOpenFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode);
+extern EAS_RESULT EAS_HWReadFile(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead);
+extern EAS_RESULT EAS_HWGetByte(EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p);
+extern EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst);
+extern EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst);
+extern EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition);
+extern EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position);
+extern EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position);
+extern EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength);
+extern EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE* pFile);
+extern EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file);
+
+/* vibrate, LED, and backlight functions */
+extern EAS_RESULT EAS_HWVibrate(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+extern EAS_RESULT EAS_HWLED(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+extern EAS_RESULT EAS_HWBackLight(EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+
+/* host yield function */
+extern EAS_BOOL EAS_HWYield(EAS_HW_DATA_HANDLE hwInstData);
+#endif /* end _EAS_HOST_H */
diff --git a/arm-wt-22k/host_src/eas_hostmm.c b/arm-wt-22k/host_src/eas_hostmm.c new file mode 100644 index 0000000..132239b --- /dev/null +++ b/arm-wt-22k/host_src/eas_hostmm.c @@ -0,0 +1,687 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_hostmm.c + * + * Contents and purpose: + * This file contains the host wrapper functions for stdio, stdlib, etc. + * This is a sample version that reads from a filedescriptor. + * The file locator (EAS_FILE_LOCATOR) handle passed to + * HWOpenFile is the same one that is passed to EAS_OpenFile. + * + * Modify this file to suit the needs of your particular system. + * + * EAS_MAX_FILE_HANDLES sets the maximum number of MIDI streams within + * a MIDI type 1 file that can be played. + * + * EAS_HW_FILE is a structure to support the file I/O functions. It + * comprises the file descriptor, the file read pointer, and + * the dup flag, which when set, indicates that the file handle has + * been duplicated, and offset and length within the file. + * + * Copyright 2005 Sonic Network Inc. + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 795 $ + * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +#ifdef _lint +#include "lint_stdlib.h" +#else +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <unistd.h> +#include <sys/types.h> +#include <sys/stat.h> +#include <fcntl.h> +#include <limits.h> +#include <sys/mman.h> +#include <errno.h> +#include <signal.h> +#define LOG_TAG "Sonivox" +#include <utils/Log.h> +#include <media/MediaPlayerInterface.h> +#endif + +#include "eas_host.h" + +/* Only for debugging LED, vibrate, and backlight functions */ +#include "eas_report.h" + +/* this module requires dynamic memory support */ +#ifdef _STATIC_MEMORY +#error "eas_hostmm.c requires the dynamic memory model!\n" +#endif + +#ifndef EAS_MAX_FILE_HANDLES +#define EAS_MAX_FILE_HANDLES 32 +#endif + +/* + * this structure and the related function are here + * to support the ability to create duplicate handles + * and buffering it in memory. If your system uses + * in-memory resources, you can eliminate the calls + * to malloc and free, the dup flag, and simply track + * the file size and read position. + */ +typedef struct eas_hw_file_tag +{ + EAS_I32 fileSize; + EAS_I32 filePos; + EAS_BOOL dup; + int fd; + EAS_I32 offset; +} EAS_HW_FILE; + +typedef struct eas_hw_inst_data_tag +{ + EAS_HW_FILE files[EAS_MAX_FILE_HANDLES]; +} EAS_HW_INST_DATA; + +pthread_key_t EAS_sigbuskey; + +/*---------------------------------------------------------------------------- + * EAS_HWInit + * + * Initialize host wrapper interface + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_HWInit (EAS_HW_DATA_HANDLE *pHWInstData) +{ + EAS_HW_FILE *file; + int i; + + /* need to track file opens for duplicate handles */ + *pHWInstData = malloc(sizeof(EAS_HW_INST_DATA)); + if (!(*pHWInstData)) + return EAS_ERROR_MALLOC_FAILED; + + EAS_HWMemSet(*pHWInstData, 0, sizeof(EAS_HW_INST_DATA)); + + file = (*pHWInstData)->files; + for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) + { + file->fd = -1; + file++; + } + + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_HWShutdown + * + * Shut down host wrapper interface + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_HWShutdown (EAS_HW_DATA_HANDLE hwInstData) +{ + + free(hwInstData); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWMalloc + * + * Allocates dynamic memory + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +void *EAS_HWMalloc (EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size) +{ + /* Since this whole library loves signed sizes, let's not let + * negative or 0 values through */ + if (size <= 0) + return NULL; + return malloc((size_t) size); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWFree + * + * Frees dynamic memory + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +void EAS_HWFree (EAS_HW_DATA_HANDLE hwInstData, void *p) +{ + free(p); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWMemCpy + * + * Copy memory wrapper + * + *---------------------------------------------------------------------------- +*/ +void *EAS_HWMemCpy (void *dest, const void *src, EAS_I32 amount) +{ + if (amount < 0) { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000004 , amount); + exit(255); + } + return memcpy(dest, src, (size_t) amount); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWMemSet + * + * Set memory wrapper + * + *---------------------------------------------------------------------------- +*/ +void *EAS_HWMemSet (void *dest, int val, EAS_I32 amount) +{ + if (amount < 0) { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000005 , amount); + exit(255); + } + return memset(dest, val, (size_t) amount); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWMemCmp + * + * Compare memory wrapper + * + *---------------------------------------------------------------------------- +*/ +EAS_I32 EAS_HWMemCmp (const void *s1, const void *s2, EAS_I32 amount) +{ + if (amount < 0) { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000006 , amount); + exit(255); + } + return (EAS_I32) memcmp(s1, s2, (size_t) amount); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWOpenFile + * + * Open a file for read or write + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_HWOpenFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode) +{ + EAS_HW_FILE *file; + int fd; + int i, temp; + + /* set return value to NULL */ + *pFile = NULL; + + /* only support read mode at this time */ + if (mode != EAS_FILE_READ) + return EAS_ERROR_INVALID_FILE_MODE; + + /* find an empty entry in the file table */ + file = hwInstData->files; + for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) + { + /* is this slot being used? */ + if (file->fd < 0) + { + if (locator->path) { + /* open the file */ + if ((fd = open(locator->path, O_RDONLY)) < 0) { + return EAS_ERROR_FILE_OPEN_FAILED; + } + } else { + /* else file is already open */ + fd = dup(locator->fd); + } + + /* determine the file size */ + if (locator->length == 0) { + if (lseek(fd, 0, SEEK_END) < 0) { + close(fd); + return EAS_ERROR_FILE_LENGTH; + } + if ((file->fileSize = lseek(fd, 0, SEEK_CUR)) == -1L) { + close(fd); + return EAS_ERROR_FILE_LENGTH; + } + } + + // file size was passed in + else { + file->fileSize = (EAS_I32) locator->length; + } + + file->fd = fd; + file->offset = locator->offset; + + /* initialize some values */ + file->filePos = 0; + file->dup = EAS_FALSE; + + *pFile = file; + return EAS_SUCCESS; + } + file++; + } + + /* too many open files */ + return EAS_ERROR_MAX_FILES_OPEN; +} + + +/*---------------------------------------------------------------------------- + * + * EAS_HWReadFile + * + * Read data from a file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWReadFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead) +{ + EAS_I32 count; + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + if (n < 0) + return EAS_EOF; + + /* calculate the bytes to read */ + count = file->fileSize - file->filePos; + if (n < count) + count = n; + if (count < 0) + return EAS_EOF; + + /* copy the data to the requested location, and advance the pointer */ + if (count) { + lseek(file->fd, file->filePos + file->offset, SEEK_SET); + count = read(file->fd, pBuffer, count); + } + file->filePos += count; + *pBytesRead = count; + + /* were n bytes read? */ + if (count!= n) + return EAS_EOF; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWGetByte + * + * Read a byte from a file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWGetByte (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p) +{ + EAS_I32 numread; + return EAS_HWReadFile(hwInstData, file, p, 1, &numread); +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWGetWord + * + * Read a 16 bit word from a file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst) +{ + EAS_RESULT result; + EAS_U8 c1, c2; + + /* read 2 bytes from the file */ + if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS) + return result; + + /* order them as requested */ + if (msbFirst) + *((EAS_U16*) p) = ((EAS_U16) c1 << 8) | c2; + else + *((EAS_U16*) p) = ((EAS_U16) c2 << 8) | c1; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWGetDWord + * + * Returns the current location in the file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst) +{ + EAS_RESULT result; + EAS_U8 c1, c2,c3,c4; + + /* read 4 bytes from the file */ + if ((result = EAS_HWGetByte(hwInstData, file, &c1)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetByte(hwInstData, file, &c2)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetByte(hwInstData, file, &c3)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetByte(hwInstData, file, &c4)) != EAS_SUCCESS) + return result; + + /* order them as requested */ + if (msbFirst) + *((EAS_U32*) p) = ((EAS_U32) c1 << 24) | ((EAS_U32) c2 << 16) | ((EAS_U32) c3 << 8) | c4; + else + *((EAS_U32*) p)= ((EAS_U32) c4 << 24) | ((EAS_U32) c3 << 16) | ((EAS_U32) c2 << 8) | c1; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWFilePos + * + * Returns the current location in the file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition) +{ + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + *pPosition = file->filePos; + return EAS_SUCCESS; +} /* end EAS_HWFilePos */ + +/*---------------------------------------------------------------------------- + * + * EAS_HWFileSeek + * + * Seek to a specific location in the file + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position) +{ + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + /* validate new position */ + if ((position < 0) || (position > file->fileSize)) + return EAS_ERROR_FILE_SEEK; + + /* save new position */ + file->filePos = position; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWFileSeekOfs + * + * Seek forward or back relative to the current position + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position) +{ + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + /* determine the file position */ + position += file->filePos; + if ((position < 0) || (position > file->fileSize)) + return EAS_ERROR_FILE_SEEK; + + /* save new position */ + file->filePos = position; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWFileLength + * + * Return the file length + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength) +{ + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + *pLength = file->fileSize; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWDupHandle + * + * Duplicate a file handle + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE *pDupFile) +{ + EAS_HW_FILE *dupFile; + int i; + + /* make sure we have a valid handle */ + if (file->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + /* find an empty entry in the file table */ + dupFile = hwInstData->files; + for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) + { + /* is this slot being used? */ + if (dupFile->fd < 0) + { + /* copy info from the handle to be duplicated */ + dupFile->filePos = file->filePos; + dupFile->fileSize = file->fileSize; + dupFile->fd = file->fd; + dupFile->offset = file->offset; + + /* set the duplicate handle flag */ + dupFile->dup = file->dup = EAS_TRUE; + + *pDupFile = dupFile; + return EAS_SUCCESS; + } + dupFile++; + } + + /* too many open files */ + return EAS_ERROR_MAX_FILES_OPEN; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWClose + * + * Wrapper for fclose function + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file1) +{ + EAS_HW_FILE *file2,*dupFile; + int i; + + + /* make sure we have a valid handle */ + if (file1->fd < 0) + return EAS_ERROR_INVALID_HANDLE; + + /* check for duplicate handle */ + if (file1->dup) + { + dupFile = NULL; + file2 = hwInstData->files; + for (i = 0; i < EAS_MAX_FILE_HANDLES; i++) + { + /* check for duplicate */ + if ((file1 != file2) && (file2->fd == file1->fd)) + { + /* is there more than one duplicate? */ + if (dupFile != NULL) + { + /* clear this entry and return */ + file1->fd = -1; + return EAS_SUCCESS; + } + + /* this is the first duplicate found */ + else + dupFile = file2; + } + file2++; + } + + /* there is only one duplicate, clear the dup flag */ + if (dupFile) + dupFile->dup = EAS_FALSE; + else + /* if we get here, there's a serious problem */ + return EAS_ERROR_HANDLE_INTEGRITY; + + /* clear this entry and return */ + file1->fd = -1; + return EAS_SUCCESS; + } + + /* no duplicates - close the file */ + close(file1->fd); + /* clear this entry and return */ + file1->fd = -1; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWVibrate + * + * Turn on/off vibrate function + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWVibrate (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) +{ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000001 , state); + return EAS_SUCCESS; +} /* end EAS_HWVibrate */ + +/*---------------------------------------------------------------------------- + * + * EAS_HWLED + * + * Turn on/off LED + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWLED (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) +{ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000002 , state); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWBackLight + * + * Turn on/off backlight + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_RESULT EAS_HWBackLight (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state) +{ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x1a54b6e8, 0x00000003 , state); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * + * EAS_HWYield + * + * This function is called periodically by the EAS library to give the + * host an opportunity to allow other tasks to run. There are two ways to + * use this call: + * + * If you have a multi-tasking OS, you can call the yield function in the + * OS to allow other tasks to run. In this case, return EAS_FALSE to tell + * the EAS library to continue processing when control returns from this + * function. + * + * If tasks run in a single thread by sequential function calls (sometimes + * call a "commutator loop"), return EAS_TRUE to cause the EAS Library to + * return to the caller. Be sure to check the number of bytes rendered + * before passing the audio buffer to the codec - it may not be filled. + * The next call to EAS_Render will continue processing until the buffer + * has been filled. + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, hwInstData) hwInstData available for customer use */ +EAS_BOOL EAS_HWYield (EAS_HW_DATA_HANDLE hwInstData) +{ + /* put your code here */ + return EAS_FALSE; +} + diff --git a/arm-wt-22k/host_src/eas_main.c b/arm-wt-22k/host_src/eas_main.c new file mode 100644 index 0000000..fed600e --- /dev/null +++ b/arm-wt-22k/host_src/eas_main.c @@ -0,0 +1,464 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_main.c + * + * Contents and purpose: + * The entry point and high-level functions for the EAS Synthesizer test + * harness. + * + * Copyright Sonic Network Inc. 2004 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 775 $ + * $Date: 2007-07-20 10:11:11 -0700 (Fri, 20 Jul 2007) $ + *---------------------------------------------------------------------------- +*/ + +#ifdef _lint +#include "lint_stdlib.h" +#else +#include <stdio.h> +#include <string.h> +#include <stdlib.h> +#include <time.h> +#endif + +#include "eas.h" +#include "eas_wave.h" +#include "eas_report.h" + +/* determines how many EAS buffers to fill a host buffer */ +#define NUM_BUFFERS 8 + +/* default file to play if no filename is specified on the command line */ +static const char defaultTestFile[] = "test.mid"; + +EAS_I32 polyphony; + +/* prototypes for helper functions */ +static void StrCopy(char *dest, const char *src, EAS_I32 size); +static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size); +static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize); +static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig); + +/* main is defined after playfile to avoid the need for two passes through lint */ + +/*---------------------------------------------------------------------------- + * PlayFile() + *---------------------------------------------------------------------------- + * Purpose: + * This function plays the file requested by filename + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ + +static EAS_RESULT PlayFile (EAS_DATA_HANDLE easData, const char* filename, const char* outputFile, const S_EAS_LIB_CONFIG *pLibConfig, void *buffer, EAS_I32 bufferSize) +{ + EAS_HANDLE handle; + EAS_RESULT result, reportResult; + EAS_I32 count; + EAS_STATE state; + EAS_I32 playTime; + char waveFilename[256]; + WAVE_FILE *wFile; + EAS_INT i; + EAS_PCM *p; + EAS_FILE file; + + /* determine the name of the output file */ + wFile = NULL; + if (outputFile == NULL) + { + StrCopy(waveFilename, filename, sizeof(waveFilename)); + if (!ChangeFileExt(waveFilename, "wav", sizeof(waveFilename))) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error in output filename %s\n", waveFilename); */ } + return EAS_FAILURE; + } + outputFile = waveFilename; + } + + /* call EAS library to open file */ + file.path = filename; + file.fd = 0; + if ((reportResult = EAS_OpenFile(easData, &file, &handle)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_OpenFile returned %ld\n", reportResult); */ } + return reportResult; + } + + /* prepare to play the file */ + if ((result = EAS_Prepare(easData, handle)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Prepare returned %ld\n", result); */ } + reportResult = result; + } + + /* get play length */ + if ((result = EAS_ParseMetaData(easData, handle, &playTime)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_ParseMetaData returned %ld\n", result); */ } + return result; + } + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0xe624f4d9, 0x00000005 , playTime / 1000, playTime % 1000); + + if (reportResult == EAS_SUCCESS) + { + /* create the output file */ + wFile = WaveFileCreate(outputFile, pLibConfig->numChannels, pLibConfig->sampleRate, sizeof(EAS_PCM) * 8); + if (!wFile) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unable to create output file %s\n", waveFilename); */ } + reportResult = EAS_FAILURE; + } + } + + /* rendering loop */ + while (reportResult == EAS_SUCCESS) + { + + /* we may render several buffers here to fill one host buffer */ + for (i = 0, p = buffer; i < NUM_BUFFERS; i++, p+= pLibConfig->mixBufferSize * pLibConfig->numChannels) + { + + /* get the current time */ + if ((result = EAS_GetLocation(easData, handle, &playTime)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_GetLocation returned %d\n",result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + break; + } + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Parser time: %d.%03d\n", playTime / 1000, playTime % 1000); */ } + + /* render a buffer of audio */ + if ((result = EAS_Render(easData, p, pLibConfig->mixBufferSize, &count)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Render returned %d\n",result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + } + + if (result == EAS_SUCCESS) + { + /* write it to the wave file */ + if (WaveFileWrite(wFile, buffer, bufferSize) != bufferSize) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "WaveFileWrite failed\n"); */ } + reportResult = EAS_FAILURE; + } + } + + if (reportResult == EAS_SUCCESS) + { + /* check stream state */ + if ((result = EAS_State(easData, handle, &state)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_State returned %d\n", result); */ } + reportResult = result; + } + + /* is playback complete */ + if ((state == EAS_STATE_STOPPED) || (state == EAS_STATE_ERROR)) + break; + } + } + + /* close the output file */ + if (wFile) + { + if (!WaveFileClose(wFile)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error closing wave file %s\n", waveFilename); */ } + if (reportResult == EAS_SUCCESS) + result = EAS_FAILURE; + } + } + + /* close the input file */ + if ((result = EAS_CloseFile(easData,handle)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_Close returned %ld\n", result); */ } + if (reportResult == EAS_SUCCESS) + result = EAS_FAILURE; + } + + return reportResult; +} /* end PlayFile */ + +/*---------------------------------------------------------------------------- + * main() + *---------------------------------------------------------------------------- + * Purpose: The entry point for the EAS sample application + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +int main( int argc, char **argv ) +{ + EAS_DATA_HANDLE easData; + const S_EAS_LIB_CONFIG *pLibConfig; + void *buffer; + EAS_RESULT result, playResult; + EAS_I32 bufferSize; + int i; + int temp; + FILE *debugFile; + char *outputFile = NULL; + + /* set the error reporting level */ + EAS_SetDebugLevel(_EAS_SEVERITY_INFO); + debugFile = NULL; + + /* process command-line arguments */ + for (i = 1; i < argc; i++) + { + /* check for switch */ + if (argv[i][0] == '-') + { + switch (argv[i][1]) + { + case 'd': + temp = argv[i][2]; + if ((temp >= '0') || (temp <= '9')) + EAS_SetDebugLevel(temp); + else + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid debug level %d\n", temp); */ } + break; + case 'f': + if ((debugFile = fopen(&argv[i][2],"w")) == NULL) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unable to create debug file %s\n", &argv[i][2]); */ } + else + EAS_SetDebugFile(debugFile, EAS_TRUE); + break; + case 'o': + outputFile = &argv[i][2]; + break; + case 'p': + polyphony = atoi(&argv[i][2]); + if (polyphony < 1) + polyphony = 1; + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Polyphony set to %d\n", polyphony); */ } + break; + default: + break; + } + continue; + } + } + + /* assume success */ + playResult = EAS_SUCCESS; + + /* get the library configuration */ + pLibConfig = EAS_Config(); + if (!EASLibraryCheck(pLibConfig)) + return -1; + if (polyphony > pLibConfig->maxVoices) + polyphony = pLibConfig->maxVoices; + + /* calculate buffer size */ + bufferSize = pLibConfig->mixBufferSize * pLibConfig->numChannels * (EAS_I32)sizeof(EAS_PCM) * NUM_BUFFERS; + + /* allocate output buffer memory */ + buffer = malloc((EAS_U32)bufferSize); + if (!buffer) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Error allocating memory for audio buffer\n"); */ } + return EAS_FAILURE; + } + + /* initialize the EAS library */ + polyphony = pLibConfig->maxVoices; + if ((result = EAS_Init(&easData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Init returned %ld - aborting!\n", result); */ } + free(buffer); + return result; + } + + /* + * Some debugging environments don't allow for passed parameters. + * In this case, just play the default MIDI file "test.mid" + */ + if (argc < 2) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", defaultTestFile); */ } + if ((playResult = PlayFile(easData, defaultTestFile, NULL, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, defaultTestFile); */ } + } + } + /* iterate through the list of files to be played */ + else + { + for (i = 1; i < argc; i++) + { + /* check for switch */ + if (argv[i][0] != '-') + { + + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "Playing '%s'\n", argv[i]); */ } + if ((playResult = PlayFile(easData, argv[i], outputFile, pLibConfig, buffer, bufferSize)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d playing file %s\n", playResult, argv[i]); */ } + break; + } + } + } + } + + /* shutdown the EAS library */ + if ((result = EAS_Shutdown(easData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_Shutdown returned %ld\n", result); */ } + } + + /* free the output buffer */ + free(buffer); + + /* close the debug file */ + if (debugFile) + fclose(debugFile); + + /* play errors take precedence over shutdown errors */ + if (playResult != EAS_SUCCESS) + return playResult; + return result; +} /* end main */ + +/*---------------------------------------------------------------------------- + * StrCopy() + *---------------------------------------------------------------------------- + * Purpose: + * Safe string copy + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +static void StrCopy(char *dest, const char *src, EAS_I32 size) +{ + int len; + + strncpy(dest, src, (size_t) size-1); + len = (int) strlen(src); + if (len < size) + dest[len] = 0; +} /* end StrCopy */ + +/*---------------------------------------------------------------------------- + * ChangeFileExt() + *---------------------------------------------------------------------------- + * Purpose: + * Changes the file extension of a filename + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +static EAS_BOOL ChangeFileExt(char *str, const char *ext, EAS_I32 size) +{ + char *p; + + /* find the extension, if any */ + p = strrchr(str,'.'); + if (!p) + { + if ((EAS_I32)(strlen(str) + 5) > size) + return EAS_FALSE; + strcat(str,"."); + strcat(str,ext); + return EAS_TRUE; + } + + /* make sure there's room for the extension */ + p++; + *p = 0; + if ((EAS_I32)(strlen(str) + 4) > size) + return EAS_FALSE; + strcat(str,ext); + return EAS_TRUE; +} /* end ChangeFileExt */ + +/*---------------------------------------------------------------------------- + * EASLibraryCheck() + *---------------------------------------------------------------------------- + * Purpose: + * Displays the library version and checks it against the header + * file used to build this code. + * + * Inputs: + * pLibConfig - library configuration retrieved from the library + * + * Outputs: + * returns EAS_TRUE if matched + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_BOOL EASLibraryCheck (const S_EAS_LIB_CONFIG *pLibConfig) +{ + + /* display the library version */ + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "EAS Library Version %d.%d.%d.%d\n", + pLibConfig->libVersion >> 24, + (pLibConfig->libVersion >> 16) & 0x0f, + (pLibConfig->libVersion >> 8) & 0x0f, + pLibConfig->libVersion & 0x0f); */ } + + /* display some info about the library build */ + if (pLibConfig->checkedVersion) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tChecked library\n"); */ } + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMaximum polyphony: %d\n", pLibConfig->maxVoices); */ } + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tNumber of channels: %d\n", pLibConfig->numChannels); */ } + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tSample rate: %d\n", pLibConfig->sampleRate); */ } + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tMix buffer size: %d\n", pLibConfig->mixBufferSize); */ } + if (pLibConfig->filterEnabled) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tFilter enabled\n"); */ } +#ifndef _WIN32_WCE + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build Timestamp: %s", ctime((time_t*)&pLibConfig->buildTimeStamp)); */ } +#endif + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tLibrary Build ID: %s\n", pLibConfig->buildGUID); */ } + + /* check it against the header file used to build this code */ + /*lint -e{778} constant expression used for display purposes may evaluate to zero */ + if (LIB_VERSION != pLibConfig->libVersion) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Library version does not match header files. EAS Header Version %d.%d.%d.%d\n", + LIB_VERSION >> 24, + (LIB_VERSION >> 16) & 0x0f, + (LIB_VERSION >> 8) & 0x0f, + LIB_VERSION & 0x0f); */ } + return EAS_FALSE; + } + return EAS_TRUE; +} /* end EASLibraryCheck */ + diff --git a/arm-wt-22k/host_src/eas_report.c b/arm-wt-22k/host_src/eas_report.c new file mode 100644 index 0000000..d4dd22c --- /dev/null +++ b/arm-wt-22k/host_src/eas_report.c @@ -0,0 +1,264 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_report.c
+ *
+ * Contents and purpose:
+ * This file contains the debug message handling routines for the EAS library.
+ * These routines should be modified as needed for your system.
+ *
+ * Copyright 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 659 $
+ * $Date: 2007-04-24 13:36:35 -0700 (Tue, 24 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdlib.h>
+#include <stdio.h>
+#include <stdarg.h>
+#endif
+
+#include "eas_report.h"
+
+static int severityLevel = 9999;
+
+/* debug file */
+static FILE *debugFile = NULL;
+int flush = 0;
+
+#ifndef _NO_DEBUG_PREPROCESSOR
+
+/* structure should have an #include for each error message header file */
+S_DEBUG_MESSAGES debugMessages[] =
+{
+#ifndef UNIFIED_DEBUG_MESSAGES
+#include "eas_config_msgs.h"
+
+
+#include "eas_host_msgs.h"
+#include "eas_hostmm_msgs.h"
+#include "eas_math_msgs.h"
+#include "eas_midi_msgs.h"
+#include "eas_mixer_msgs.h"
+#include "eas_pcm_msgs.h"
+#include "eas_public_msgs.h"
+#include "eas_smf_msgs.h"
+#include "eas_wave_msgs.h"
+#include "eas_voicemgt_msgs.h"
+
+#ifdef _FM_SYNTH
+#include "eas_fmsynth_msgs.h"
+#include "eas_fmengine_msgs.h"
+#endif
+
+#ifdef _WT_SYNTH
+#include "eas_wtsynth_msgs.h"
+#include "eas_wtengine_msgs.h"
+#endif
+
+#ifdef _ARM_TEST_MAIN
+#include "arm_main_msgs.h"
+#endif
+
+#ifdef _EAS_MAIN
+#include "eas_main_msgs.h"
+#endif
+
+#ifdef _EAS_MAIN_IPC
+#include "eas_main_ipc_msgs.h"
+#endif
+
+#ifdef _METRICS_ENABLED
+#include "eas_perf_msgs.h"
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+#include "eas_compressor_msgs.h"
+#endif
+
+#ifdef _ENHANCER_ENABLED
+#include "eas_enhancer_msgs.h"
+#endif
+
+#ifdef _WOW_ENABLED
+#include "eas_wow_msgs.h"
+#endif
+
+#ifdef _SMAF_PARSER
+#include "eas_smaf_msgs.h"
+#endif
+
+#ifdef _OTA_PARSER
+#include "eas_ota_msgs.h"
+#endif
+
+#ifdef _IMELODY_PARSER
+#include "eas_imelody_msgs.h"
+#endif
+
+#ifdef _WAVE_PARSER
+#include "eas_wavefile_msgs.h"
+#endif
+
+#if defined(_CMX_PARSER) || defined(_MFI_PARSER)
+#include "eas_cmf_msgs.h"
+#endif
+
+#if defined(_CMX_PARSER) || defined(_MFI_PARSER) || defined(_WAVE_PARSER)
+#include "eas_imaadpcm_msgs.h"
+#endif
+
+#else
+#include "eas_debugmsgs.h"
+#endif
+
+/* denotes end of error messages */
+{ 0,0,0 }
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_ReportEx()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...)
+{
+ va_list vargs;
+ int i;
+
+ /* check severity level */
+ if (severity > severityLevel)
+ return;
+
+ /* find the error message and output to stdout */
+ /*lint -e{661} we check for NULL pointer - no fence post error here */
+ for (i = 0; debugMessages[i].m_pDebugMsg; i++)
+ {
+ if ((debugMessages[i].m_nHashCode == hashCode) &&
+ (debugMessages[i].m_nSerialNum == serialNum))
+ {
+ /*lint -e{826} <allow variable args> */
+ va_start(vargs, serialNum);
+ if (debugFile)
+ {
+ vfprintf(debugFile, debugMessages[i].m_pDebugMsg, vargs);
+ if (flush)
+ fflush(debugFile);
+ }
+ else
+ {
+ vprintf(debugMessages[i].m_pDebugMsg, vargs);
+ }
+ va_end(vargs);
+ return;
+ }
+ }
+ printf("Unrecognized error: Severity=%d; HashCode=%lu; SerialNum=%d\n", severity, hashCode, serialNum);
+} /* end EAS_ReportEx */
+
+#else
+/*----------------------------------------------------------------------------
+ * EAS_Report()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_Report (int severity, const char *fmt, ...)
+{
+ va_list vargs;
+
+ /* check severity level */
+ if (severity > severityLevel)
+ return;
+
+ /*lint -e{826} <allow variable args> */
+ va_start(vargs, fmt);
+ if (debugFile)
+ {
+ vfprintf(debugFile, fmt, vargs);
+ if (flush)
+ fflush(debugFile);
+ }
+ else
+ {
+ vprintf(fmt, vargs);
+ }
+ va_end(vargs);
+} /* end EAS_Report */
+
+/*----------------------------------------------------------------------------
+ * EAS_ReportX()
+ *
+ * This is the error message handler. The default handler outputs error
+ * messages to stdout. Modify this as needed for your system.
+ *----------------------------------------------------------------------------
+*/
+void EAS_ReportX (int severity, const char *fmt, ...)
+{
+ va_list vargs;
+
+ /* check severity level */
+ if (severity > severityLevel)
+ return;
+
+ /*lint -e{826} <allow variable args> */
+ va_start(vargs, fmt);
+ if (debugFile)
+ {
+ vfprintf(debugFile, fmt, vargs);
+ if (flush)
+ fflush(debugFile);
+ }
+ else
+ {
+ vprintf(fmt, vargs);
+ }
+ va_end(vargs);
+} /* end EAS_ReportX */
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS_SetDebugLevel()
+ *
+ * Sets the level for debug message output
+ *----------------------------------------------------------------------------
+*/
+
+void EAS_SetDebugLevel (int severity)
+{
+ severityLevel = severity;
+} /* end EAS_SetDebugLevel */
+
+/*----------------------------------------------------------------------------
+ * EAS_SetDebugFile()
+ *
+ * Redirect debugger output to the specified file.
+ *----------------------------------------------------------------------------
+*/
+void EAS_SetDebugFile (void *file, int flushAfterWrite)
+{
+ debugFile = (FILE*) file;
+ flush = flushAfterWrite;
+} /* end EAS_SetDebugFile */
+
diff --git a/arm-wt-22k/host_src/eas_report.h b/arm-wt-22k/host_src/eas_report.h new file mode 100644 index 0000000..9d7c8e8 --- /dev/null +++ b/arm-wt-22k/host_src/eas_report.h @@ -0,0 +1,77 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_report.h
+ *
+ * Contents and purpose:
+ * This file contains the debug message handling routines for the EAS library.
+ * These routines should be modified as needed for your system.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+/* sentinel */
+#ifndef _EAS_REPORT_H
+#define _EAS_REPORT_H
+
+#define _EAS_SEVERITY_NOFILTER 0
+#define _EAS_SEVERITY_FATAL 1
+#define _EAS_SEVERITY_ERROR 2
+#define _EAS_SEVERITY_WARNING 3
+#define _EAS_SEVERITY_INFO 4
+#define _EAS_SEVERITY_DETAIL 5
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifndef _NO_DEBUG_PREPROCESSOR
+
+/* structure for included debug message header files */
+typedef struct
+{
+ unsigned long m_nHashCode;
+ int m_nSerialNum;
+ char *m_pDebugMsg;
+} S_DEBUG_MESSAGES;
+
+/* debug message handling prototypes */
+extern void EAS_ReportEx (int severity, unsigned long hashCode, int serialNum, ...);
+
+#else
+
+/* these prototypes are used if the debug preprocessor is not used */
+extern void EAS_Report (int severity, const char* fmt, ...);
+extern void EAS_ReportX (int severity, const char* fmt, ...);
+
+#endif
+
+extern void EAS_SetDebugLevel (int severity);
+extern void EAS_SetDebugFile (void *file, int flushAfterWrite);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+#endif
diff --git a/arm-wt-22k/host_src/eas_reverb.h b/arm-wt-22k/host_src/eas_reverb.h new file mode 100644 index 0000000..a2535fb --- /dev/null +++ b/arm-wt-22k/host_src/eas_reverb.h @@ -0,0 +1,54 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverb.h
+ *
+ * Contents and purpose:
+ * Contains parameter enumerations for the Reverb effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 300 $
+ * $Date: 2006-09-11 17:37:20 -0700 (Mon, 11 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_REVERB_H
+#define _EAS_REVERB_H
+
+
+/* enumerated parameter settings for Reverb effect */
+typedef enum
+{
+ EAS_PARAM_REVERB_BYPASS,
+ EAS_PARAM_REVERB_PRESET,
+ EAS_PARAM_REVERB_WET,
+ EAS_PARAM_REVERB_DRY
+} E_REVERB_PARAMS;
+
+
+typedef enum
+{
+ EAS_PARAM_REVERB_LARGE_HALL,
+ EAS_PARAM_REVERB_HALL,
+ EAS_PARAM_REVERB_CHAMBER,
+ EAS_PARAM_REVERB_ROOM,
+} E_REVERB_PRESETS;
+
+
+#endif /* _REVERB_H */
diff --git a/arm-wt-22k/host_src/eas_types.h b/arm-wt-22k/host_src/eas_types.h new file mode 100644 index 0000000..5ba1e4e --- /dev/null +++ b/arm-wt-22k/host_src/eas_types.h @@ -0,0 +1,273 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_types.h
+ *
+ * Contents and purpose:
+ * The public interface header for the EAS synthesizer.
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 726 $
+ * $Date: 2007-06-14 23:10:46 -0700 (Thu, 14 Jun 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_TYPES_H
+#define _EAS_TYPES_H
+
+/* EAS_RESULT return codes */
+typedef long EAS_RESULT;
+#define EAS_SUCCESS 0
+#define EAS_FAILURE -1
+#define EAS_ERROR_INVALID_MODULE -2
+#define EAS_ERROR_MALLOC_FAILED -3
+#define EAS_ERROR_FILE_POS -4
+#define EAS_ERROR_INVALID_FILE_MODE -5
+#define EAS_ERROR_FILE_SEEK -6
+#define EAS_ERROR_FILE_LENGTH -7
+#define EAS_ERROR_NOT_IMPLEMENTED -8
+#define EAS_ERROR_CLOSE_FAILED -9
+#define EAS_ERROR_FILE_OPEN_FAILED -10
+#define EAS_ERROR_INVALID_HANDLE -11
+#define EAS_ERROR_NO_MIX_BUFFER -12
+#define EAS_ERROR_PARAMETER_RANGE -13
+#define EAS_ERROR_MAX_FILES_OPEN -14
+#define EAS_ERROR_UNRECOGNIZED_FORMAT -15
+#define EAS_BUFFER_SIZE_MISMATCH -16
+#define EAS_ERROR_FILE_FORMAT -17
+#define EAS_ERROR_SMF_NOT_INITIALIZED -18
+#define EAS_ERROR_LOCATE_BEYOND_END -19
+#define EAS_ERROR_INVALID_PCM_TYPE -20
+#define EAS_ERROR_MAX_PCM_STREAMS -21
+#define EAS_ERROR_NO_VOICE_ALLOCATED -22
+#define EAS_ERROR_INVALID_CHANNEL -23
+#define EAS_ERROR_ALREADY_STOPPED -24
+#define EAS_ERROR_FILE_READ_FAILED -25
+#define EAS_ERROR_HANDLE_INTEGRITY -26
+#define EAS_ERROR_MAX_STREAMS_OPEN -27
+#define EAS_ERROR_INVALID_PARAMETER -28
+#define EAS_ERROR_FEATURE_NOT_AVAILABLE -29
+#define EAS_ERROR_SOUND_LIBRARY -30
+#define EAS_ERROR_NOT_VALID_IN_THIS_STATE -31
+#define EAS_ERROR_NO_VIRTUAL_SYNTHESIZER -32
+#define EAS_ERROR_FILE_ALREADY_OPEN -33
+#define EAS_ERROR_FILE_ALREADY_CLOSED -34
+#define EAS_ERROR_INCOMPATIBLE_VERSION -35
+#define EAS_ERROR_QUEUE_IS_FULL -36
+#define EAS_ERROR_QUEUE_IS_EMPTY -37
+#define EAS_ERROR_FEATURE_ALREADY_ACTIVE -38
+
+/* special return codes */
+#define EAS_EOF 3
+#define EAS_STREAM_BUFFERING 4
+#define EAS_BUFFER_FULL 5
+
+/* EAS_STATE return codes */
+typedef long EAS_STATE;
+typedef enum
+{
+ EAS_STATE_READY = 0,
+ EAS_STATE_PLAY,
+ EAS_STATE_STOPPING,
+ EAS_STATE_PAUSING,
+ EAS_STATE_STOPPED,
+ EAS_STATE_PAUSED,
+ EAS_STATE_OPEN,
+ EAS_STATE_ERROR,
+ EAS_STATE_EMPTY
+} E_EAS_STATE;
+
+/* constants */
+#ifndef EAS_CONST
+#define EAS_CONST const
+#endif
+
+/* definition for public interface functions */
+#ifndef EAS_PUBLIC
+#define EAS_PUBLIC
+#endif
+
+/* boolean values */
+typedef unsigned EAS_BOOL;
+typedef unsigned char EAS_BOOL8;
+
+#define EAS_FALSE 0
+#define EAS_TRUE 1
+
+/* scalar variable definitions */
+typedef unsigned char EAS_U8;
+typedef signed char EAS_I8;
+typedef char EAS_CHAR;
+
+typedef unsigned short EAS_U16;
+typedef short EAS_I16;
+
+typedef unsigned long EAS_U32;
+typedef long EAS_I32;
+
+typedef unsigned EAS_UINT;
+typedef int EAS_INT;
+typedef long EAS_LONG;
+
+/* audio output type */
+typedef short EAS_PCM;
+
+/* file open modes */
+typedef EAS_I32 EAS_FILE_MODE;
+#define EAS_FILE_READ 1
+#define EAS_FILE_WRITE 2
+
+/* file locator e.g. filename or memory pointer */
+typedef struct s_eas_file_tag {
+ const char* path;
+ int fd;
+ long long offset;
+ long long length;
+} EAS_FILE, *EAS_FILE_LOCATOR;
+
+/* handle to stream */
+typedef struct s_eas_stream_tag *EAS_HANDLE;
+
+/* handle to file */
+typedef struct eas_hw_file_tag *EAS_FILE_HANDLE;
+
+/* handle for synthesizer data */
+typedef struct s_eas_data_tag *EAS_DATA_HANDLE;
+
+/* handle to persistent data for host wrapper interface */
+typedef struct eas_hw_inst_data_tag *EAS_HW_DATA_HANDLE;
+
+/* handle to sound library */
+typedef struct s_eas_sndlib_tag *EAS_SNDLIB_HANDLE;
+typedef struct s_eas_dls_tag *EAS_DLSLIB_HANDLE;
+
+/* pointer to frame buffer - used in split architecture only */
+typedef struct s_eas_frame_buffer_tag *EAS_FRAME_BUFFER_HANDLE;
+
+/* untyped pointer for instance data */
+typedef void *EAS_VOID_PTR;
+
+/* inline functions */
+#ifndef EAS_INLINE
+#if defined (__XCC__)
+#define EAS_INLINE __inline__
+#elif defined (__GNUC__)
+#define EAS_INLINE inline static
+#else
+#define EAS_INLINE __inline
+#endif
+#endif
+
+/* define NULL value */
+#ifndef NULL
+#define NULL 0
+#endif
+
+/* metadata types for metadata return codes */
+typedef enum
+{
+ EAS_METADATA_UNKNOWN = 0,
+ EAS_METADATA_TITLE,
+ EAS_METADATA_AUTHOR,
+ EAS_METADATA_COPYRIGHT,
+ EAS_METADATA_LYRIC,
+ EAS_METADATA_TEXT
+} E_EAS_METADATA_TYPE;
+
+/* metadata callback function */
+typedef void (*EAS_METADATA_CBFUNC) (E_EAS_METADATA_TYPE metaDataType, char *metaDataBuf, EAS_VOID_PTR pUserData);
+
+/* file types for metadata return codes */
+typedef enum
+{
+ EAS_FILE_UNKNOWN = 0,
+ EAS_FILE_SMF0,
+ EAS_FILE_SMF1,
+ EAS_FILE_SMAF_UNKNOWN,
+ EAS_FILE_SMAF_MA2,
+ EAS_FILE_SMAF_MA3,
+ EAS_FILE_SMAF_MA5,
+ EAS_FILE_CMX,
+ EAS_FILE_MFI,
+ EAS_FILE_OTA,
+ EAS_FILE_IMELODY,
+ EAS_FILE_RTTTL,
+ EAS_FILE_XMF0,
+ EAS_FILE_XMF1,
+ EAS_FILE_WAVE_PCM,
+ EAS_FILE_WAVE_IMA_ADPCM,
+ EAS_FILE_MMAPI_TONE_CONTROL
+} E_EAS_FILE_TYPE;
+
+/* enumeration for synthesizers */
+typedef enum
+{
+ EAS_MCU_SYNTH = 0,
+ EAS_DSP_SYNTH
+} E_SYNTHESIZER;
+
+/* external audio callback program change */
+typedef struct s_ext_audio_prg_chg_tag
+{
+ EAS_U16 bank;
+ EAS_U8 program;
+ EAS_U8 channel;
+} S_EXT_AUDIO_PRG_CHG;
+
+/* external audio callback event */
+typedef struct s_ext_audio_event_tag
+{
+ EAS_U8 channel;
+ EAS_U8 note;
+ EAS_U8 velocity;
+ EAS_BOOL8 noteOn;
+} S_EXT_AUDIO_EVENT;
+
+typedef struct s_midi_controllers_tag
+{
+ EAS_U8 modWheel; /* CC1 */
+ EAS_U8 volume; /* CC7 */
+ EAS_U8 pan; /* CC10 */
+ EAS_U8 expression; /* CC11 */
+ EAS_U8 channelPressure; /* MIDI channel pressure */
+
+#ifdef _REVERB
+ EAS_U8 reverbSend; /* CC91 */
+#endif
+
+#ifdef _CHORUS
+ EAS_U8 chorusSend; /* CC93 */
+#endif
+} S_MIDI_CONTROLLERS;
+
+/* iMode play modes enumeration for EAS_SetPlayMode */
+typedef enum
+{
+ IMODE_PLAY_ALL = 0,
+ IMODE_PLAY_PARTIAL
+} E_I_MODE_PLAY_MODE;
+
+typedef EAS_BOOL (*EAS_EXT_PRG_CHG_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_PRG_CHG *pPrgChg);
+typedef EAS_BOOL (*EAS_EXT_EVENT_FUNC) (EAS_VOID_PTR pInstData, S_EXT_AUDIO_EVENT *pEvent);
+
+#endif /* #ifndef _EAS_TYPES_H */
diff --git a/arm-wt-22k/host_src/eas_wave.c b/arm-wt-22k/host_src/eas_wave.c new file mode 100644 index 0000000..02fed6e --- /dev/null +++ b/arm-wt-22k/host_src/eas_wave.c @@ -0,0 +1,423 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wave.c
+ *
+ * Contents and purpose:
+ * This module contains .WAV file functions for the EAS synthesizer
+ * test harness.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 658 $
+ * $Date: 2007-04-24 13:35:49 -0700 (Tue, 24 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* lint complaints about most C library headers, so we use our own during lint step */
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdio.h>
+#include <stdlib.h>
+#endif
+
+#include "eas_wave.h"
+
+/* .WAV file format tags */
+const EAS_U32 riffTag = 0x46464952;
+const EAS_U32 waveTag = 0x45564157;
+const EAS_U32 fmtTag = 0x20746d66;
+const EAS_U32 dataTag = 0x61746164;
+
+#ifdef _BIG_ENDIAN
+/*----------------------------------------------------------------------------
+ * FlipDWord()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip a DWORD for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipDWord (EAS_U32 *pValue)
+{
+ EAS_U8 *p;
+ EAS_U32 temp;
+
+ p = (EAS_U8*) pValue;
+ temp = (((((p[3] << 8) | p[2]) << 8) | p[1]) << 8) | p[0];
+ *pValue = temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FlipWord()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip a WORD for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipWord (EAS_U16 *pValue)
+{
+ EAS_U8 *p;
+ EAS_U16 temp;
+
+ p = (EAS_U8*) pValue;
+ temp = (p[1] << 8) | p[0];
+ *pValue = temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FlipWaveHeader()
+ *----------------------------------------------------------------------------
+ * Purpose: Endian flip the wave header for big-endian processors
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void FlipWaveHeader (WAVE_HEADER *p)
+{
+
+ FlipDWord(&p->nRiffTag);
+ FlipDWord(&p->nRiffSize);
+ FlipDWord(&p->nWaveTag);
+ FlipDWord(&p->nFmtTag);
+ FlipDWord(&p->nFmtSize);
+ FlipDWord(&p->nDataTag);
+ FlipDWord(&p->nDataSize);
+ FlipWord(&p->fc.wFormatTag);
+ FlipWord(&p->fc.nChannels);
+ FlipDWord(&p->fc.nSamplesPerSec);
+ FlipDWord(&p->fc.nAvgBytesPerSec);
+ FlipWord(&p->fc.nBlockAlign);
+ FlipWord(&p->fc.wBitsPerSample);
+
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * WaveFileCreate()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for writing and writes the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample)
+{
+ WAVE_FILE *wFile;
+
+ /* allocate memory */
+ wFile = malloc(sizeof(WAVE_FILE));
+ if (!wFile)
+ return NULL;
+ wFile->write = EAS_TRUE;
+
+ /* create the file */
+ wFile->file = fopen(filename,"wb");
+ if (!wFile->file)
+ {
+ free(wFile);
+ return NULL;
+ }
+
+ /* initialize PCM format .WAV file header */
+ wFile->wh.nRiffTag = riffTag;
+ wFile->wh.nRiffSize = sizeof(WAVE_HEADER) - 8;
+ wFile->wh.nWaveTag = waveTag;
+ wFile->wh.nFmtTag = fmtTag;
+ wFile->wh.nFmtSize = sizeof(FMT_CHUNK);
+
+ /* initalize 'fmt' chunk */
+ wFile->wh.fc.wFormatTag = 1;
+ wFile->wh.fc.nChannels = (EAS_U16) nChannels;
+ wFile->wh.fc.nSamplesPerSec = (EAS_U32) nSamplesPerSec;
+ wFile->wh.fc.wBitsPerSample = (EAS_U16) wBitsPerSample;
+ wFile->wh.fc.nBlockAlign = (EAS_U16) (nChannels * (EAS_U16) (wBitsPerSample / 8));
+ wFile->wh.fc.nAvgBytesPerSec = wFile->wh.fc.nBlockAlign * (EAS_U32) nSamplesPerSec;
+
+ /* initialize 'data' chunk */
+ wFile->wh.nDataTag = dataTag;
+ wFile->wh.nDataSize = 0;
+
+#ifdef _BIG_ENDIAN
+ FlipWaveHeader(&wFile->wh);
+#endif
+
+ /* write the header */
+ if (fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file) != 1)
+ {
+ fclose(wFile->file);
+ free(wFile);
+ return NULL;
+ }
+
+#ifdef _BIG_ENDIAN
+ FlipWaveHeader(&wFile->wh);
+#endif
+
+ /* return the file handle */
+ return wFile;
+} /* end WaveFileCreate */
+
+/*----------------------------------------------------------------------------
+ * WaveFileWrite()
+ *----------------------------------------------------------------------------
+ * Purpose: Writes data to the wave file
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n)
+{
+ EAS_I32 count;
+
+ /* make sure we have an open file */
+ if (wFile == NULL)
+ {
+ return 0;
+ }
+
+#ifdef _BIG_ENDIAN
+ {
+ EAS_I32 i;
+ EAS_U16 *p;
+ p = buffer;
+ i = n >> 1;
+ while (i--)
+ FlipWord(p++);
+ }
+#endif
+
+ /* write the data */
+ count = (EAS_I32) fwrite(buffer, 1, (size_t) n, wFile->file);
+
+ /* add the number of bytes written */
+ wFile->wh.nRiffSize += (EAS_U32) count;
+ wFile->wh.nDataSize += (EAS_U32) count;
+
+ /* return the count of bytes written */
+ return count;
+} /* end WriteWaveHeader */
+
+/*----------------------------------------------------------------------------
+ * WaveFileClose()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for writing and writes the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+EAS_BOOL WaveFileClose (WAVE_FILE *wFile)
+{
+ EAS_I32 count = 1;
+
+ /* return to beginning of file and write the header */
+ if (wFile->write)
+ {
+ if (fseek(wFile->file, 0L, SEEK_SET) == 0)
+ {
+
+#ifdef _BIG_ENDIAN
+ FlipWaveHeader(&wFile->wh);
+#endif
+ count = (EAS_I32) fwrite(&wFile->wh, sizeof(WAVE_HEADER), 1, wFile->file);
+#ifdef _BIG_ENDIAN
+ FlipWaveHeader(&wFile->wh);
+#endif
+ }
+ }
+
+ /* close the file */
+ if (fclose(wFile->file) != 0)
+ count = 0;
+
+ /* free the memory */
+ free(wFile);
+
+ /* return the file handle */
+ return (count == 1 ? EAS_TRUE : EAS_FALSE);
+} /* end WaveFileClose */
+
+#ifdef _WAVE_FILE_READ
+#ifdef _BIG_ENDIAN
+#error "WaveFileOpen not currently supported on big-endian processors"
+#endif
+/*----------------------------------------------------------------------------
+ * WaveFileOpen()
+ *----------------------------------------------------------------------------
+ * Purpose: Opens a wave file for reading and reads the header
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+WAVE_FILE *WaveFileOpen (const char *filename)
+{
+ WAVE_FILE *wFile;
+ struct
+ {
+ EAS_U32 tag;
+ EAS_U32 size;
+ } chunk;
+ EAS_U32 tag;
+ EAS_I32 startChunkPos;
+ EAS_INT state;
+ EAS_BOOL done;
+
+ /* allocate memory */
+ wFile = malloc(sizeof(WAVE_FILE));
+ if (!wFile)
+ return NULL;
+
+ /* open the file */
+ wFile->write = EAS_FALSE;
+ wFile->file = fopen(filename,"rb");
+ if (!wFile->file)
+ {
+ free(wFile);
+ return NULL;
+ }
+
+ /* make lint happy */
+ chunk.tag = chunk.size = 0;
+ startChunkPos = 0;
+
+ /* read the RIFF tag and file size */
+ state = 0;
+ done = EAS_FALSE;
+ while (!done)
+ {
+
+ switch(state)
+ {
+ /* read the RIFF tag */
+ case 0:
+ if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+ done = EAS_TRUE;
+ else
+ {
+ if (chunk.tag != riffTag)
+ done = EAS_TRUE;
+ else
+ state++;
+ }
+ break;
+
+ /* read the WAVE tag */
+ case 1:
+ if (fread(&tag, sizeof(tag), 1, wFile->file) != 1)
+ done = EAS_TRUE;
+ else
+ {
+ if (tag != waveTag)
+ done = EAS_TRUE;
+ else
+ state++;
+ }
+ break;
+
+ /* looking for fmt chunk */
+ case 2:
+ if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+ done = EAS_TRUE;
+ else
+ {
+ startChunkPos = ftell(wFile->file);
+
+ /* not fmt tag, skip it */
+ if (chunk.tag != fmtTag)
+ fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+ else
+ state++;
+ }
+ break;
+
+ /* read fmt chunk */
+ case 3:
+ if (fread(&wFile->wh.fc, sizeof(FMT_CHUNK), 1, wFile->file) != 1)
+ done = EAS_TRUE;
+ else
+ {
+ fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+ state++;
+ }
+ break;
+
+ /* looking for data chunk */
+ case 4:
+ if (fread(&chunk, sizeof(chunk), 1, wFile->file) != 1)
+ done = EAS_TRUE;
+ else
+ {
+ startChunkPos = ftell(wFile->file);
+
+ /* not data tag, skip it */
+ if (chunk.tag != dataTag)
+ fseek(wFile->file, startChunkPos + (EAS_I32) chunk.size, SEEK_SET);
+ else
+ {
+ wFile->dataSize = chunk.size;
+ state++;
+ done = EAS_TRUE;
+ }
+ }
+ break;
+
+ default:
+ done = EAS_TRUE;
+ break;
+ }
+ }
+
+ /* if not final state, an error occurred */
+ if (state != 5)
+ {
+ fclose(wFile->file);
+ free(wFile);
+ return NULL;
+ }
+
+ /* return the file handle */
+ return wFile;
+} /* end WaveFileOpen */
+#endif
+
+
+
diff --git a/arm-wt-22k/host_src/eas_wave.h b/arm-wt-22k/host_src/eas_wave.h new file mode 100644 index 0000000..ca388f5 --- /dev/null +++ b/arm-wt-22k/host_src/eas_wave.h @@ -0,0 +1,74 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wave.h
+ *
+ * Contents and purpose:
+ * Writes output to a .WAV file
+ *
+ * DO NOT MODIFY THIS FILE!
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+
+/* sentinel */
+#ifndef _EAS_WAVE_H
+#define _EAS_WAVE_H
+
+/* .WAV file format chunk */
+typedef struct {
+ EAS_U16 wFormatTag;
+ EAS_U16 nChannels;
+ EAS_U32 nSamplesPerSec;
+ EAS_U32 nAvgBytesPerSec;
+ EAS_U16 nBlockAlign;
+ EAS_U16 wBitsPerSample;
+} FMT_CHUNK;
+
+/* .WAV file header */
+typedef struct {
+ EAS_U32 nRiffTag;
+ EAS_U32 nRiffSize;
+ EAS_U32 nWaveTag;
+ EAS_U32 nFmtTag;
+ EAS_U32 nFmtSize;
+ FMT_CHUNK fc;
+ EAS_U32 nDataTag;
+ EAS_U32 nDataSize;
+} WAVE_HEADER;
+
+typedef struct {
+ WAVE_HEADER wh;
+ FILE *file;
+ EAS_BOOL write;
+ EAS_U32 dataSize;
+} WAVE_FILE;
+
+WAVE_FILE *WaveFileCreate (const char *filename, EAS_I32 nChannels, EAS_I32 nSamplesPerSec, EAS_I32 wBitsPerSample);
+EAS_I32 WaveFileWrite (WAVE_FILE *wFile, void *buffer, EAS_I32 n);
+EAS_BOOL WaveFileClose (WAVE_FILE *wFile);
+WAVE_FILE *WaveFileOpen (const char *filename);
+
+#endif /* end #ifndef _EAS_WAVE_H */
+
+
+
diff --git a/arm-wt-22k/host_src/jet.h b/arm-wt-22k/host_src/jet.h new file mode 100644 index 0000000..b5bce00 --- /dev/null +++ b/arm-wt-22k/host_src/jet.h @@ -0,0 +1,199 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * jet.h
+ *
+ * Contents and purpose:
+ * Public interface for JET sound engine
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 554 $
+ * $Date: 2007-02-02 11:06:10 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _JET_H
+#define _JET_H
+
+#include "eas_types.h"
+#include "eas.h"
+
+/* for C++ linkage */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* opaque handle types for JET interface */
+typedef struct s_jet_data_tag *JET_DATA_HANDLE;
+
+typedef struct s_jet_config_tag
+{
+ EAS_U8 appEventRangeLow;
+ EAS_U8 appEventRangeHigh;
+} S_JET_CONFIG;
+
+typedef struct s_jet_status_tag
+{
+ EAS_INT currentUserID;
+ EAS_INT segmentRepeatCount;
+ EAS_INT numQueuedSegments;
+ EAS_BOOL paused;
+ EAS_I32 location;
+ EAS_U8 currentPlayingSegment;
+ EAS_U8 currentQueuedSegment;
+} S_JET_STATUS;
+
+typedef struct s_jet_event_tag
+{
+ EAS_U8 segment;
+ EAS_U8 channel;
+ EAS_U8 track;
+ EAS_U8 controller;
+ EAS_U8 value;
+} S_JET_EVENT;
+
+/*----------------------------------------------------------------------------
+ * JET_Init()
+ *----------------------------------------------------------------------------
+ * Initializes the JET library, allocates memory, etc. Call
+ * JET_Shutdown to de-allocate memory. Pass NULL for pConfig
+ * to use defaults. If passing config data, configSize should be
+ * sizeof(S_JET_CONFIG). This allows for future expansion of the
+ * config structure while maintaining compatibility.
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Init (EAS_DATA_HANDLE easHandle, const S_JET_CONFIG *pConfig, EAS_INT configSize);
+
+/*----------------------------------------------------------------------------
+ * JET_Shutdown()
+ *----------------------------------------------------------------------------
+ * Frees any memory used by the JET library
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Shutdown (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_OpenFile()
+ *----------------------------------------------------------------------------
+ * Opens a JET content file for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_OpenFile (EAS_DATA_HANDLE easHandle, EAS_FILE_LOCATOR locator);
+
+/*----------------------------------------------------------------------------
+ * JET_GetAppData()
+ *----------------------------------------------------------------------------
+ * Returns location and size of application data in the JET file
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT JET_GetAppData (EAS_DATA_HANDLE easHandle, EAS_I32 *pAppDataOffset, EAS_I32 *pAppDataSize);
+
+/*----------------------------------------------------------------------------
+ * JET_CloseFile()
+ *----------------------------------------------------------------------------
+ * Closes a JET content file and releases associated resources
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_CloseFile (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_Status()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Status (EAS_DATA_HANDLE easHandle, S_JET_STATUS *pStatus);
+
+/*----------------------------------------------------------------------------
+ * JET_GetEvent()
+ *----------------------------------------------------------------------------
+ * Checks for application events
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL JET_GetEvent (EAS_DATA_HANDLE easHandle, EAS_U32 *pEventRaw, S_JET_EVENT *pEvent);
+
+/*----------------------------------------------------------------------------
+ * JET_ParseEvent()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC void JET_ParseEvent (EAS_U32 event, S_JET_EVENT *pEvent);
+
+/*----------------------------------------------------------------------------
+ * JET_QueueSegment()
+ *----------------------------------------------------------------------------
+ * Queue a segment for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_QueueSegment (EAS_DATA_HANDLE easHandle, EAS_INT segmentNum, EAS_INT libNum, EAS_INT repeatCount, EAS_INT transpose, EAS_U32 muteFlags, EAS_U8 userID);
+
+/*----------------------------------------------------------------------------
+ * JET_Play()
+ *----------------------------------------------------------------------------
+ * Starts playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Play (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_Pause()
+ *----------------------------------------------------------------------------
+ * Pauses playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Pause (EAS_DATA_HANDLE easHandle);
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlags()
+ *----------------------------------------------------------------------------
+ * Change the state of the mute flags
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlags (EAS_DATA_HANDLE easHandle, EAS_U32 muteFlags, EAS_BOOL sync);
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlag()
+ *----------------------------------------------------------------------------
+ * Change the state of a single mute flag
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlag (EAS_DATA_HANDLE easHandle, EAS_INT trackNum, EAS_BOOL muteFlag, EAS_BOOL sync);
+
+/*----------------------------------------------------------------------------
+ * JET_TriggerClip()
+ *----------------------------------------------------------------------------
+ * Unmute a track and then mute it when it is complete
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_TriggerClip (EAS_DATA_HANDLE easHandle, EAS_INT clipID);
+
+/*----------------------------------------------------------------------------
+ * JET_Clear_Queue()
+ *----------------------------------------------------------------------------
+ * Clears all segments in the queue
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Clear_Queue (EAS_DATA_HANDLE easHandle);
+
+#ifdef __cplusplus
+} /* end extern "C" */
+#endif
+
+
+#endif
+
diff --git a/arm-wt-22k/lib/libarm-wt-22k.a b/arm-wt-22k/lib/libarm-wt-22k.a Binary files differnew file mode 100644 index 0000000..b130714 --- /dev/null +++ b/arm-wt-22k/lib/libarm-wt-22k.a diff --git a/arm-wt-22k/lib_src/ARM-E_filter_gnu.s b/arm-wt-22k/lib_src/ARM-E_filter_gnu.s new file mode 100644 index 0000000..871cd7d --- /dev/null +++ b/arm-wt-22k/lib_src/ARM-E_filter_gnu.s @@ -0,0 +1,134 @@ +@***********************************************************
+@ Function: WT_VoiceFilter
+@ Processor: ARM
+@ Description:
+@ Implements a 2-pole low-pass filter with resonanance
+@
+@ Usage:
+@ void WT_VoiceFilter(
+@ S_FILTER CONTROL *pFilter,
+@ S_WT_FRAME *pWTFrame);
+@
+@ Copyright 2005 Sonic Network, Inc.
+@****************************************************************
+@ Revision Control:
+@ $Revision: 496 $
+@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@ where:
+@ S_FILTER_CONTROL *pFilter
+@ PASSED IN: r0
+@
+@ S_WT_FRAME *pWTFrame
+@ PASSED IN: r1
+@****************************************************************
+
+ .include "ARM_synth_constants_gnu.inc"
+
+ .arm
+ .text
+
+
+ .global WT_VoiceFilter
+
+
+@ Register usage
+@ --------------
+pFilter .req r0
+pWTFrame .req r1
+pBuffer .req r2
+numSamples .req r3
+
+z1 .req r4
+z2 .req r5
+b1 .req r6
+b2 .req r7
+K .req r8
+
+tmp0 .req r1 @ reuse register
+tmp1 .req r9
+tmp2 .req r10
+
+
+@SaveRegs RLIST {r4-r10, lr}
+@RestoreRegs RLIST {r4-r10, pc}
+
+
+ .func WT_VoiceFilter
+WT_VoiceFilter:
+
+ STMFD sp!, {r4-r10, lr}
+
+@
+@ Setup passed parameters in their destination registers
+@----------------------------------------------------------------
+
+ LDR pBuffer, [pWTFrame, #m_pAudioBuffer]
+ LDR numSamples, [pWTFrame, #m_numSamples]
+
+ @load state variables from pFilter structure
+ LDRSH z1, [pFilter, #m_z1]
+ LDRSH z2, [pFilter, #m_z2]
+
+ @load coefficients from pWTFrame structure
+ LDR K, [pWTFrame, #m_k]
+ LDR b1, [pWTFrame, #m_b1]
+ LDR b2, [pWTFrame, #m_b2]
+
+ RSB b1, b1, #0 @ b1 = -b1
+ RSB b2, b2, #0 @ b2 = -b2
+ MOV b2, b2, ASR #1 @ b2 = b2 >> 1
+ MOV K, K, ASR #1 @ K = K >> 1
+
+@
+@ Start processing
+@----------------------------------------------------------------
+
+ LDRSH tmp0, [pBuffer] @ fetch sample
+
+FilterLoop:
+ SMULBB tmp2, z1, b1 @ tmp2 = z1 * -b1
+ SMLABB tmp2, z2, b2, tmp2 @ tmp2 = (-b1 * z1) + (-b2 * z2)
+
+ MOV z2, z1 @ delay line
+
+ SMLABB tmp0, tmp0, K, tmp2 @ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2)
+
+ LDRSH tmp1, [pBuffer, #NEXT_OUTPUT_PCM] @ fetch next sample
+
+ MOV z1, tmp0, ASR #14 @ shift result to low word
+ STRH z1, [pBuffer], #NEXT_OUTPUT_PCM @ write back to buffer
+
+ SMULBB tmp2, z1, b1 @ tmp2 = z1 * -b1
+
+ SUBS numSamples, numSamples, #2 @ unroll loop once
+
+ SMLABB tmp2, z2, b2, tmp2 @ tmp2 = (-b1 * z1) + (-b2 * z2)
+
+ SMLABB tmp1, tmp1, K, tmp2 @ tmp1 = (K * x[n]) + (-b1 * z1) + (-b2 * z2)
+
+ MOV z2, z1 @ delay line
+
+ MOV z1, tmp1, ASR #14 @ shift result to low word
+
+ LDRGTSH tmp0, [pBuffer, #NEXT_OUTPUT_PCM] @ fetch next sample
+
+ STRH z1, [pBuffer], #NEXT_OUTPUT_PCM @ write back to buffer
+
+ BGT FilterLoop
+@ save z terms
+@----------------------------------------------------------------
+
+ STRH z1, [pFilter, #m_z1]
+ STRH z2, [pFilter, #m_z2]
+
+@ Return to calling function
+@----------------------------------------------------------------
+
+ LDMFD sp!,{r4-r10, lr}
+ BX lr
+
+ .endfunc
+ .end
+
diff --git a/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s b/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s new file mode 100644 index 0000000..847a7f0 --- /dev/null +++ b/arm-wt-22k/lib_src/ARM-E_interpolate_loop_gnu.s @@ -0,0 +1,131 @@ +@***********************************************************
+@ Function: WT_Interpolate
+@ Processor: ARM-E
+@ Description: the main synthesis function when fetching
+@ wavetable samples.
+@ C-callable.
+@
+@ Usage:
+@ void WT_Interpolate(
+@ S_WT_VOICE *pWTVoice,
+@ S_WT_FRAME *pWTFrame);
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@ $Revision: 496 $
+@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@ where:
+@ S_WT_VOICE *pWTVoice
+@ PASSED IN: r0
+@
+@ S_WT_FRAME *pWTFrame;
+@ PASSED IN: r1
+@****************************************************************
+
+ .include "ARM_synth_constants_gnu.inc"
+
+ .arm
+ .text
+
+ .global WT_Interpolate
+
+
+@ Register usage
+@ --------------
+pWTVoice .req r0
+pWTFrame .req r1
+
+numSamples .req r2
+phaseIncrement .req r3
+pOutputBuffer .req r4
+
+tmp0 .req r1 @reuse register
+tmp1 .req r5
+tmp2 .req r6
+
+pLoopEnd .req r7
+pLoopStart .req r8
+
+pPhaseAccum .req r9
+phaseFrac .req r10
+phaseFracMask .req r11
+
+@SaveRegs RLIST {r4-r11,lr}
+@RestoreRegs RLIST {r4-r11,pc}
+
+ .func WT_Interpolate
+WT_Interpolate:
+
+ STMFD sp!,{r4-r11,lr}
+
+@
+@ Fetch parameters from structures
+@----------------------------------------------------------------
+
+ LDR pOutputBuffer, [pWTFrame, #m_pAudioBuffer]
+ LDR numSamples, [pWTFrame, #m_numSamples]
+
+ LDR phaseIncrement, [pWTFrame, #m_phaseIncrement]
+ LDR pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+ LDR phaseFrac, [pWTVoice, #m_phaseFrac]
+ LDR phaseFracMask,=PHASE_FRAC_MASK
+
+ LDR pLoopStart, [pWTVoice, #m_pLoopStart]
+ LDR pLoopEnd, [pWTVoice, #m_pLoopEnd]
+ ADD pLoopEnd, pLoopEnd, #1 @ need loop end to equal last sample + 1
+
+InterpolationLoop:
+ SUBS tmp0, pPhaseAccum, pLoopEnd @ check for loop end
+ ADDGE pPhaseAccum, pLoopStart, tmp0 @ loop back to start
+
+ .ifdef SAMPLES_8_BIT
+ LDRSB tmp0, [pPhaseAccum] @ tmp0 = x0
+ LDRSB tmp1, [pPhaseAccum, #1] @ tmp1 = x1
+ .else
+ LDRSH tmp0, [pPhaseAccum] @ tmp0 = x0
+ LDRSH tmp1, [pPhaseAccum, #2] @ tmp1 = x1
+ .endif
+
+ ADD tmp2, phaseIncrement, phaseFrac @ increment pointer here to avoid pipeline stall
+
+ SUB tmp1, tmp1, tmp0 @ tmp1 = x1 - x0
+ SMULBB tmp1, phaseFrac, tmp1 @ tmp1 = phaseFrac * tmp2
+
+@ This section performs a gain adjustment of -12dB for 16-bit samples
+@ or +36dB for 8-bit samples. For a high quality synthesizer, the output
+@ can be set to full scale, however if the filter is used, it can overflow
+@ with certain coefficients and signal sources. In this case, either a
+@ saturation operation should take in the filter before scaling back to
+@ 16 bits or the signal path should be increased to 18 bits or more.
+
+ .ifdef SAMPLES_8_BIT
+ MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB
+ .else
+ MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB
+ .endif
+
+ ADD tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6) @ tmp1 = tmp0 + (tmp1 >> (15-6))
+ @ = x0 + f * (x1 - x0) == interpolated result
+
+ STRH tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM @ *pOutputBuffer++ = interpolated result
+
+@ carry overflow from fraction to integer portion
+ ADD pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT)
+ AND phaseFrac, tmp2, phaseFracMask @ nphaseFrac = frac part
+
+ SUBS numSamples, numSamples, #1
+ BGT InterpolationLoop
+
+@ update and store phase
+ STR pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+ STR phaseFrac, [pWTVoice, #m_phaseFrac]
+
+ LDMFD sp!,{r4-r11,lr}
+ BX lr
+
+ .endfunc
+ .end
+
diff --git a/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s b/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s new file mode 100644 index 0000000..6343762 --- /dev/null +++ b/arm-wt-22k/lib_src/ARM-E_interpolate_noloop_gnu.s @@ -0,0 +1,130 @@ +@***********************************************************
+@ Function: WT_InterpolateNoLoop
+@ Processor: ARM-E
+@ Description: the main synthesis function when fetching
+@ wavetable samples.
+@ C-callable.
+@
+@ Usage:
+@ void WT_InterpolateNoLoop(
+@ S_WT_VOICE *pWTVoice,
+@ S_WT_FRAME *pWTFrame);
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@ $Revision: 496 $
+@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@ where:
+@ S_WT_VOICE *pWTVoice
+@ PASSED IN: r0
+@
+@ S_WT_FRAME *pWTFrame;
+@ PASSED IN: r1
+@****************************************************************
+
+ .include "ARM_synth_constants_gnu.inc"
+
+ .arm
+ .text
+
+
+ .global WT_InterpolateNoLoop
+
+
+@ Register usage
+@ --------------
+pWTVoice .req r0
+pWTFrame .req r1
+pOutputBuffer .req r2
+numSamples .req r3
+
+phaseIncrement .req r4
+pPhaseAccum .req r5
+phaseFrac .req r6
+phaseFracMask .req r7
+
+tmp0 .req r1 @ reuse register
+tmp1 .req r8
+tmp2 .req r9
+
+
+@SaveRegs RLIST {r4-r9,lr}
+@RestoreRegs RLIST {r4-r9,pc}
+
+ .func WT_InterpolateNoLoop
+WT_InterpolateNoLoop:
+
+ STMFD sp!, {r4-r9,lr}
+
+@
+@ Fetch parameters from structures
+@----------------------------------------------------------------
+
+ LDR pOutputBuffer, [pWTFrame, #m_pAudioBuffer]
+ LDR numSamples, [pWTFrame, #m_numSamples]
+
+ LDR phaseIncrement, [pWTFrame, #m_phaseIncrement]
+ LDR pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+ LDR phaseFrac, [pWTVoice, #m_phaseFrac]
+ LDR phaseFracMask,=PHASE_FRAC_MASK
+
+InterpolationLoop:
+
+ .ifdef SAMPLES_8_BIT
+ LDRSB tmp0, [pPhaseAccum] @ tmp0 = x0
+ LDRSB tmp1, [pPhaseAccum, #1] @ tmp1 = x1
+ .else
+ LDRSH tmp0, [pPhaseAccum] @ tmp0 = x0
+ LDRSH tmp1, [pPhaseAccum, #2] @ tmp1 = x1
+ .endif
+
+ ADD tmp2, phaseIncrement, phaseFrac @ increment pointer here to avoid pipeline stall
+
+ SUB tmp1, tmp1, tmp0 @ tmp1 = x1 - x0
+ SMULBB tmp1, phaseFrac, tmp1 @ tmp1 = phaseFrac * tmp2
+
+@ This section performs a gain adjustment of -12dB for 16-bit samples
+@ or +36dB for 8-bit samples. For a high quality synthesizer, the output
+@ can be set to full scale, however if the filter is used, it can overflow
+@ with certain coefficients and signal sources. In this case, either a
+@ saturation operation should take in the filter before scaling back to
+@ 16 bits or the signal path should be increased to 18 bits or more.
+
+ .ifdef SAMPLES_8_BIT
+ MOV tmp0, tmp0, LSL #6 @ boost 8-bit signal by 36dB
+ .else
+ MOV tmp0, tmp0, ASR #2 @ reduce 16-bit signal by 12dB
+ .endif
+
+ ADD tmp1, tmp0, tmp1, ASR #(NUM_EG1_FRAC_BITS-6) @ tmp1 = tmp0 + (tmp1 >> (15-6))
+ @ = x0 + f * (x1 - x0) == interpolated result
+
+ STRH tmp1, [pOutputBuffer], #NEXT_OUTPUT_PCM @ *pOutputBuffer++ = interpolated result
+
+@ carry overflow from fraction to integer portion
+ ADD pPhaseAccum, pPhaseAccum, tmp2, LSR #(NUM_PHASE_FRAC_BITS - NEXT_INPUT_PCM_SHIFT)
+ AND phaseFrac, tmp2, phaseFracMask @ nphaseFrac = frac part
+
+ SUBS numSamples, numSamples, #1
+ BGT InterpolationLoop
+
+@ Clean up and store any changes that were caused during the loop
+@----------------------------------------------------------------
+
+ @ update and store phase
+ STR pPhaseAccum, [pWTVoice, #m_pPhaseAccum]
+ STR phaseFrac, [pWTVoice, #m_phaseFrac]
+
+@
+@ Return to calling function
+@----------------------------------------------------------------
+
+ LDMFD sp!,{r4-r9,lr}
+ BX lr
+
+ .endfunc
+ .end
+
diff --git a/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s b/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s new file mode 100644 index 0000000..b4e905b --- /dev/null +++ b/arm-wt-22k/lib_src/ARM-E_mastergain_gnu.s @@ -0,0 +1,109 @@ +@***********************************************************
+@ Function: SynthMasterGain
+@ Processor: ARM-E
+@ Description: Copies 32-bit synth output to 16-bit buffer
+@ with saturated gain control
+@ C-callable.
+@
+@ Usage:
+@ SynthMasterGain(
+@ pInputBuffer
+@ pOutputBuffer,
+@ nGain,
+@ nNumLoopSamples
+@ );
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@ $Revision: 496 $
+@ $Date: 2006-12-11 14:33:26 -0800 (Mon, 11 Dec 2006) $
+@****************************************************************
+@
+@ where:
+@ long *pInputBuffer
+@ PASSED IN: r0
+@
+@ EAS_PCM *pOutputBuffer
+@ PASSED IN: r1
+@
+@ short nGain
+@ PASSED IN: r2
+@
+@ EAS_U16 nNumLoopSamples
+@ PASSED IN: r3
+@
+@****************************************************************
+
+ .include "ARM_synth_constants_gnu.inc"
+
+ .arm
+ .text
+
+ .func SynthMasterGain
+SynthMasterGain:
+
+ .global SynthMasterGain @ allow other files to use this function
+
+
+
+
+
+@ Stack frame
+@ -----------
+ .equ RET_ADDR_SZ, 0 @return address
+ .equ REG_SAVE_SZ, 0 @save-on-entry registers saved
+ .equ FRAME_SZ, (8) @local variables
+ .equ ARG_BLK_SZ, 0 @argument block
+
+ .equ PARAM_OFFSET, (ARG_BLK_SZ + FRAME_SZ + REG_SAVE_SZ + RET_ADDR_SZ)
+
+@ Register usage
+@ --------------
+pnInputBuffer .req r0
+pnOutputBuffer .req r1
+nGain .req r2
+nNumLoopSamples .req r3
+
+ STMFD sp!,{r4-r6,r14} @Save any save-on-entry registers that are used
+
+ LDR r6, =0x7fff @constant for saturation tests
+
+loop:
+ LDR r4, [pnInputBuffer], #4 @fetch 1st output sample
+
+ LDR r5, [pnInputBuffer], #4 @fetch 2nd output sample
+
+ SMULWB r4, r4, nGain @output = gain * input
+
+ CMP r4, r6 @check for positive saturation
+ MOVGT r4, r6 @saturate
+ CMN r4, r6 @check for negative saturation
+ MVNLT r4, r6 @saturate
+
+ SMULWB r5, r5, nGain @output = gain * input
+
+ STRH r4, [pnOutputBuffer], #NEXT_OUTPUT_PCM @save 1st output sample
+
+ CMP r5, r6 @check for positive saturation
+ MOVGT r5, r6 @saturate
+ CMN r5, r6 @check for negative saturation
+ MVNLT r5, r6 @saturate
+ STRH r5, [pnOutputBuffer], #NEXT_OUTPUT_PCM @save 2nd output sample
+
+ SUBS nNumLoopSamples, nNumLoopSamples, #2
+ BGT loop
+
+@
+@ Return to calling function
+@----------------------------------------------------------------
+
+ LDMFD sp!,{r4-r6, lr} @ return to calling function
+ BX lr
+
+@*****************************************************************************
+
+ .endfunc @ end of function/procedure
+
+ .end @ end of assembly code
+
diff --git a/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s b/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s new file mode 100644 index 0000000..4517a3d --- /dev/null +++ b/arm-wt-22k/lib_src/ARM-E_voice_gain_gnu.s @@ -0,0 +1,166 @@ +@***********************************************************
+@ Function: WT_VoiceGain
+@ Processor: ARM-E
+@ Description: the main synthesis function when fetching
+@ wavetable samples.
+@ C-callable.
+@
+@ Usage:
+@ Usage:
+@ WT_VoiceGain(
+@ S_WT_VOICE *pWTVoice,
+@ S_WT_FRAME *pWTFrame);
+@
+@ Copyright 2004, 2005 Sonic Network, Inc.
+@****************************************************************
+@ Revision Control:
+@ $Revision: 814 $
+@ $Date: 2007-08-02 10:34:53 -0700 (Thu, 02 Aug 2007) $
+@****************************************************************
+@
+@ where:
+@ S_WT_VOICE *psVoice
+@ PASSED IN: r0
+@
+@ S_WT_FRAME *pWTFrame
+@ PASSED IN: r1
+@****************************************************************
+
+
+
+ .include "ARM_synth_constants_gnu.inc"
+
+ .arm
+ .text
+
+ .global WT_VoiceGain
+
+@ Register usage
+@ --------------
+pWTVoice .req r0
+pWTFrame .req r1
+pInputBuffer .req r2
+pMixBuffer .req r3
+
+tmp0 .req r4
+tmp1 .req r5
+tmp2 .req r1 @ reuse register
+tmp3 .req r6
+
+numSamples .req r9
+
+ .if STEREO_OUTPUT
+gainIncLeft .req r7
+gainIncRight .req r8
+gainLeft .req r10
+gainRight .req r11
+ .else
+gainIncrement .req r7
+gain .req r8
+ .endif
+
+
+@ register context for local variables
+@SaveRegs RLIST {r4-r11,lr}
+@RestoreRegs RLIST {r4-r11,pc}
+
+ .func WT_VoiceGain
+WT_VoiceGain:
+
+ STMFD sp!, {r4-r11,lr}
+
+ LDR pInputBuffer, [pWTFrame, #m_pAudioBuffer]
+ LDR pMixBuffer, [pWTFrame, #m_pMixBuffer]
+ LDR numSamples, [pWTFrame, #m_numSamples]
+
+@----------------------------------------------------------------
+@ Stereo version
+@----------------------------------------------------------------
+@ NOTE: instructions are reordered to reduce the effect of latency
+@ due to storage and computational dependencies.
+@----------------------------------------------------------------
+
+ .if STEREO_OUTPUT
+
+ LDR tmp0, [pWTFrame, #m_prevGain]
+ LDR tmp1, [pWTFrame, #m_gainTarget]
+
+ LDRSH gainLeft, [pWTVoice, #m_gainLeft]
+ LDRSH gainRight, [pWTVoice, #m_gainRight]
+
+ MOV gainIncLeft, gainLeft
+ SMULBB gainLeft, tmp0, gainLeft
+
+ SMULBB gainIncLeft, tmp1, gainIncLeft
+ SUB gainIncLeft, gainIncLeft, gainLeft
+ MOV gainLeft, gainLeft, ASR #(NUM_MIXER_GUARD_BITS - 2)
+ MOV gainIncLeft, gainIncLeft, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2)
+
+ MOV gainIncRight, gainRight
+ SMULBB gainRight, tmp0, gainRight
+
+ SMULBB gainIncRight, tmp1, gainIncRight
+ SUB gainIncRight, gainIncRight, gainRight
+ MOV gainRight, gainRight, ASR #(NUM_MIXER_GUARD_BITS - 2)
+ MOV gainIncRight, gainIncRight, ASR #(SYNTH_UPDATE_PERIOD_IN_BITS + NUM_MIXER_GUARD_BITS - 2)
+
+ LDRSH tmp0, [pInputBuffer], #2
+
+StereoGainLoop:
+ LDR tmp1, [pMixBuffer]
+
+ ADD gainLeft, gainLeft, gainIncLeft
+
+ SMLAWB tmp1, gainLeft, tmp0, tmp1
+
+ LDR tmp2, [pMixBuffer, #4]
+
+ ADD gainRight, gainRight, gainIncRight
+
+ STR tmp1, [pMixBuffer], #4
+
+ SMLAWB tmp2, gainRight, tmp0, tmp2
+
+ SUBS numSamples, numSamples, #1
+
+ LDRGTSH tmp0, [pInputBuffer], #2
+
+ STR tmp2, [pMixBuffer], #4
+
+ BGT StereoGainLoop
+
+@----------------------------------------------------------------
+@ Mono version
+@----------------------------------------------------------------
+ .else
+
+ LDR gain, [pWTFrame, #m_prevGain]
+ MOV gain, gain, LSL #(NUM_MIXER_GUARD_BITS + 4)
+ LDR gainIncrement, [pWTFrame, #m_gainTarget]
+ MOV gainIncrement, gainIncrement, LSL #(NUM_MIXER_GUARD_BITS + 4)
+ SUB gainIncrement, gainIncrement, gain
+ MOV gainIncrement, gainIncrement, ASR #SYNTH_UPDATE_PERIOD_IN_BITS
+
+MonoGainLoop:
+
+ LDRSH tmp0, [pInputBuffer], #NEXT_OUTPUT_PCM @ fetch voice output
+
+ LDR tmp1, [pMixBuffer] @ get left channel output sample
+ ADD gain, gain, gainIncrement @ gain step to eliminate zipper noise
+ SMULWB tmp0, gain, tmp0 @ sample * local gain
+
+ MOV tmp0, tmp0, ASR #1 @ add 6dB headroom
+ ADD tmp1, tmp0, tmp1
+ STR tmp1, [pMixBuffer], #4 @ save and bump pointer
+
+ SUBS numSamples, numSamples, #1
+ BGT MonoGainLoop
+
+ .endif @end Mono version
+
+ LDMFD sp!,{r4-r11,lr}
+ BX lr
+
+ .endfunc
+ .end
+
diff --git a/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc b/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc new file mode 100644 index 0000000..b4cd29a --- /dev/null +++ b/arm-wt-22k/lib_src/ARM_synth_constants_gnu.inc @@ -0,0 +1,153 @@ +@***********************************************************
+@ File: ARM_synth_constants.inc
+@ Processor: ARM
+@ Description: Contains constants and defines, most of which
+@ are mirrored in synth.h
+@
+@ Copyright Sonic Network Inc. 2004
+@****************************************************************
+@ Revision Control:
+@ $Revision: 741 $
+@ $Date: 2007-06-22 16:39:21 -0700 (Fri, 22 Jun 2007) $
+@****************************************************************
+
+
+ .ifdef SAMPLE_RATE_8000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 5
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 32
+ .endif
+
+ .ifdef SAMPLE_RATE_16000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 6
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 64
+ .endif
+
+ .ifdef SAMPLE_RATE_20000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128
+ .endif
+
+ .ifdef SAMPLE_RATE_22050
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128
+ .endif
+
+ .ifdef SAMPLE_RATE_24000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128
+ .endif
+
+ .ifdef SAMPLE_RATE_32000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 7
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 128
+ .endif
+
+ .ifdef SAMPLE_RATE_44100
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 8
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 256
+ .endif
+
+ .ifdef SAMPLE_RATE_48000
+ .equ SYNTH_UPDATE_PERIOD_IN_BITS, 8
+ .equ BUFFER_SIZE_IN_MONO_SAMPLES, 256
+ .endif
+
+
+@ if the OUTPUT PCM sample is 16-bits, then when using indexed addressing,
+@ the next sample is this many bytes away
+ .equ NEXT_OUTPUT_PCM, 2
+
+@****************************************************************************
+@/* macros for fractional phase accumulator */
+ .equ NUM_PHASE_FRAC_BITS, 15
+
+ .equ PHASE_FRAC_MASK, 0x7FFF
+
+@ shift for phase accumulator when fraction carries over
+ .ifdef SAMPLES_8_BIT
+ .equ NEXT_INPUT_PCM_SHIFT, 0
+ .endif
+
+ .ifdef SAMPLES_16_BIT
+ .equ NEXT_INPUT_PCM_SHIFT, 1
+ .endif
+
+@****************************************************************************
+ .equ NUM_MIXER_GUARD_BITS, 4
+
+@****************************************************************************
+@/* Envelope 1 (EG1) calculation macros */
+ .equ NUM_EG1_FRAC_BITS, 15
+
+@****************************************************************************
+
+ .equ NUM_ENHANCER_FILTER_COEF_FRAC_BITS, 5
+
+@****************************************************************************
+
+@
+@ I've temporarily given up on the idea of getting ADS/RV and gcc to
+@ handle a struct in a compatible fashion. Switching to old fashion EQU
+@
+
+ .if FILTER_ENABLED
+@**************************************
+@ typedef struct s_filter_tag
+ .equ m_z1, 0
+ .equ m_z2, 2
+ .endif
+
+@**************************************
+@ typedef struct s_wt_frame_tag
+ .equ m_gainTarget, 0
+ .equ m_phaseIncrement, 4
+
+ .if FILTER_ENABLED
+ .equ m_k, 8
+ .equ m_b1, 12
+ .equ m_b2, 16
+ .equ m_pAudioBuffer, 20
+ .equ m_pMixBuffer, 24
+ .equ m_numSamples, 28
+ .equ m_prevGain, 32
+ .else
+ .equ m_pAudioBuffer, 8
+ .equ m_pMixBuffer, 12
+ .equ m_numSamples, 16
+ .equ m_prevGain, 20
+ .endif
+
+
+@**************************************
+@ typedef struct s_wt_voice_tag
+ .equ m_pLoopEnd, 0 @ /* points to last PCM sample (not 1 beyond last) */
+ .equ m_pLoopStart, 4 @ /* points to first sample at start of loop */
+ .equ m_pPhaseAccum, 8 @ /* points to first sample at start of loop */
+ .equ m_phaseFrac, 12 @ /* points to first sample at start of loop */
+
+ .if STEREO_OUTPUT
+ .equ m_gainLeft, 16 @ /* current gain, left ch */
+ .equ m_gainRight, 18 @ /* current gain, right ch */
+ .endif
+
+
+@****************************************************************************
+@ enhancer
+ .equ m_nEnhancerFeedForward1, 0
+ .equ m_nEnhancerFeedback1, 1
+ .equ m_nDriveCoef, 2
+ .equ m_nEnhancerFeedback2, 3
+ .equ m_nWet, 4
+ .equ m_nDry, 5
+
+ .equ m_zF0L, 6 @ filter 1 zero state var, left
+ .equ m_zF1L, 8 @ filter 1 pole state var, left
+ .equ m_zF2L, 10 @ filter 2 zero state var, left
+ .equ m_zF0R, 12 @ filter 1 zero state var, right
+ .equ m_zF1R, 14 @ filter 1 pole state var, right
+ .equ m_zF2R, 16 @ filter 2 zero state var, right
+
+@****************************************************************************
+
+
+
diff --git a/arm-wt-22k/lib_src/arm-wt-22k_lib.mak b/arm-wt-22k/lib_src/arm-wt-22k_lib.mak new file mode 100644 index 0000000..8e9824f --- /dev/null +++ b/arm-wt-22k/lib_src/arm-wt-22k_lib.mak @@ -0,0 +1,25 @@ +#
+# Auto-generated sample makefile
+#
+# This makefile is intended for use with GNU make.
+# Set the paths to the tools (CC, AR, LD, etc.)
+#
+
+vpath %.c lib_src
+
+CC = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AS = C:\Program Files\GNUARM\bin\arm-elf-as.exe
+LD = C:\Program Files\GNUARM\bin\arm-elf-gcc.exe
+AR = C:\Program Files\GNUARM\bin\arm-elf-ar.exe
+
+%.o: %.c
+ $(CC) -c -O2 -o $@ -I lib_src -I host_src -D NUM_OUTPUT_CHANNELS=2 -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=64 -D EAS_WT_SYNTH -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _XMF_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D DLS_SYNTHESIZER -D _IMA_DECODER -D MMAPI_SUPPORT -D NATIVE_EAS_KERNEL -D JET_INTERFACE $<
+
+%.o: %.s
+ $(AS) -o $@ -EL -mcpu=arm946e-s -mfpu=softfpa -I lib_src --defsym CHECK_STACK=0 --defsym REVERB=0 --defsym CHORUS=0 --defsym STEREO_OUTPUT=1 --defsym SAMPLE_RATE_22050=1 --defsym SAMPLES_8_BIT=1 --defsym FILTER_ENABLED=1 $<
+
+OBJS = eas_mididata.o eas_pan.o eas_wavefiledata.o eas_imelody.o eas_xmfdata.o ARM-E_interpolate_noloop_gnu.o eas_chorusdata.o ARM-E_voice_gain_gnu.o eas_ota.o eas_reverbdata.o eas_rtttl.o eas_reverb.o jet.o eas_mdls.o eas_mixbuf.o eas_smf.o eas_tcdata.o eas_chorus.o eas_pcmdata.o eas_xmf.o eas_smfdata.o eas_math.o eas_tonecontrol.o eas_rtttldata.o eas_voicemgt.o eas_public.o eas_dlssynth.o ARM-E_interpolate_loop_gnu.o ARM-E_filter_gnu.o eas_midi.o eas_otadata.o eas_flog.o eas_wtengine.o eas_imaadpcm.o eas_wtsynth.o wt_22khz.o eas_pcm.o eas_mixer.o eas_wavefile.o eas_ima_tables.o eas_data.o ARM-E_mastergain_gnu.o eas_imelodydata.o
+
+arm-wt-22k.a: $(OBJS)
+ $(AR) rc lib$@ $(OBJS)
+
diff --git a/arm-wt-22k/lib_src/dls.h b/arm-wt-22k/lib_src/dls.h new file mode 100644 index 0000000..ca1d4c7 --- /dev/null +++ b/arm-wt-22k/lib_src/dls.h @@ -0,0 +1,268 @@ + /*
+
+ dls.h
+
+ Description:
+
+ Interface defines and structures for the Instrument Collection Form
+ RIFF DLS.
+
+ Written by Sonic Foundry 1996. Released for public use.
+
+ */
+
+ #ifndef _INC_DLS
+ #define _INC_DLS
+
+ /*
+
+ Layout of an instrument collection:
+
+
+ RIFF [] 'DLS ' [colh,INSTLIST,WAVEPOOL,INFOLIST]
+
+ INSTLIST
+ LIST [] 'lins'
+ LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+ LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+ LIST [] 'ins ' [insh,RGNLIST,ARTLIST,INFOLIST]
+
+ RGNLIST
+ LIST [] 'lrgn'
+ LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
+ LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
+ LIST [] 'rgn ' [rgnh,wsmp,wlnk,ARTLIST]
+
+ ARTLIST
+ LIST [] 'lart'
+ 'art1' level 1 Articulation connection graph
+ 'art2' level 2 Articulation connection graph
+ '3rd1' Possible 3rd party articulation structure 1
+ '3rd2' Possible 3rd party articulation structure 2 .... and so on
+
+ WAVEPOOL
+ ptbl [] [pool table]
+ LIST [] 'wvpl'
+ [path],
+ [path],
+ LIST [] 'wave',RIFFWAVE
+ LIST [] 'wave',RIFFWAVE
+ LIST [] 'wave',RIFFWAVE
+ LIST [] 'wave',RIFFWAVE
+ LIST [] 'wave',RIFFWAVE
+
+ INFOLIST
+ LIST [] 'INFO'
+ 'icmt' 'One of those crazy comments.'
+ 'icop' 'Copyright (C) 1996 Sonic Foundry'
+
+ */
+
+
+ /*
+ FOURCC's used in the DLS file
+ */
+/* shree */
+//#define FAR
+
+/* shree
+
+ #define FOURCC_DLS mmioFOURCC('D','L','S',' ')
+ #define FOURCC_COLH mmioFOURCC('c','o','l','h')
+ #define FOURCC_WVPL mmioFOURCC('w','v','p','l')
+ #define FOURCC_PTBL mmioFOURCC('p','t','b','l')
+ #define FOURCC_PATH mmioFOURCC('p','a','t','h')
+ #define FOURCC_wave mmioFOURCC('w','a','v','e')
+ #define FOURCC_LINS mmioFOURCC('l','i','n','s')
+ #define FOURCC_INS mmioFOURCC('i','n','s',' ')
+ #define FOURCC_INSH mmioFOURCC('i','n','s','h')
+ #define FOURCC_LRGN mmioFOURCC('l','r','g','n')
+ #define FOURCC_RGN mmioFOURCC('r','g','n',' ')
+ #define FOURCC_RGNH mmioFOURCC('r','g','n','h')
+ #define FOURCC_LART mmioFOURCC('l','a','r','t')
+ #define FOURCC_ART1 mmioFOURCC('a','r','t','1')
+ #define FOURCC_WLNK mmioFOURCC('w','l','n','k')
+ #define FOURCC_WSMP mmioFOURCC('w','s','m','p')
+ #define FOURCC_VERS mmioFOURCC('v','e','r','s')
+*/
+ /*
+ Articulation connection graph definitions
+ */
+
+ /* Generic Sources */
+ #define CONN_SRC_NONE 0x0000
+ #define CONN_SRC_LFO 0x0001
+ #define CONN_SRC_KEYONVELOCITY 0x0002
+ #define CONN_SRC_KEYNUMBER 0x0003
+ #define CONN_SRC_EG1 0x0004
+ #define CONN_SRC_EG2 0x0005
+ #define CONN_SRC_PITCHWHEEL 0x0006
+
+ /* Midi Controllers 0-127 */
+ #define CONN_SRC_CC1 0x0081
+ #define CONN_SRC_CC7 0x0087
+ #define CONN_SRC_CC10 0x008a
+ #define CONN_SRC_CC11 0x008b
+
+ /* Registered Parameter Numbers */
+ #define CONN_SRC_RPN0 0x0100
+ #define CONN_SRC_RPN1 0x0101
+ #define CONN_SRC_RPN2 0x0102
+
+ /* Generic Destinations */
+ #define CONN_DST_NONE 0x0000
+ #define CONN_DST_ATTENUATION 0x0001
+ #define CONN_DST_RESERVED 0x0002
+ #define CONN_DST_PITCH 0x0003
+ #define CONN_DST_PAN 0x0004
+
+ /* LFO Destinations */
+ #define CONN_DST_LFO_FREQUENCY 0x0104
+ #define CONN_DST_LFO_STARTDELAY 0x0105
+
+ /* EG1 Destinations */
+ #define CONN_DST_EG1_ATTACKTIME 0x0206
+ #define CONN_DST_EG1_DECAYTIME 0x0207
+ #define CONN_DST_EG1_RESERVED 0x0208
+ #define CONN_DST_EG1_RELEASETIME 0x0209
+ #define CONN_DST_EG1_SUSTAINLEVEL 0x020a
+
+ /* EG2 Destinations */
+ #define CONN_DST_EG2_ATTACKTIME 0x030a
+ #define CONN_DST_EG2_DECAYTIME 0x030b
+ #define CONN_DST_EG2_RESERVED 0x030c
+ #define CONN_DST_EG2_RELEASETIME 0x030d
+ #define CONN_DST_EG2_SUSTAINLEVEL 0x030e
+
+ #define CONN_TRN_NONE 0x0000
+ #define CONN_TRN_CONCAVE 0x0001
+
+ typedef struct _DLSVERSION {
+ DWORD dwVersionMS;
+ DWORD dwVersionLS;
+ }DLSVERSION, FAR *LPDLSVERSION;
+
+
+ typedef struct _CONNECTION {
+ USHORT usSource;
+ USHORT usControl;
+ USHORT usDestination;
+ USHORT usTransform;
+ LONG lScale;
+ }CONNECTION, FAR *LPCONNECTION;
+
+
+ /* Level 1 Articulation Data */
+
+ typedef struct _CONNECTIONLIST {
+ ULONG cbSize; /* size of the connection list structure */
+ ULONG cConnections; /* count of connections in the list */
+ } CONNECTIONLIST, FAR *LPCONNECTIONLIST;
+
+
+
+ /*
+ Generic type defines for regions and instruments
+ */
+
+ typedef struct _RGNRANGE {
+ USHORT usLow;
+ USHORT usHigh;
+ }RGNRANGE, FAR * LPRGNRANGE;
+
+ #define F_INSTRUMENT_DRUMS 0x80000000
+
+ typedef struct _MIDILOCALE {
+ ULONG ulBank;
+ ULONG ulInstrument;
+ }MIDILOCALE, FAR *LPMIDILOCALE;
+
+ /*
+ Header structures found in an DLS file for collection, instruments, and
+ regions.
+ */
+
+ #define F_RGN_OPTION_SELFNONEXCLUSIVE 0x0001
+
+ typedef struct _RGNHEADER {
+ RGNRANGE RangeKey; /* Key range */
+ RGNRANGE RangeVelocity; /* Velocity Range */
+ USHORT fusOptions; /* Synthesis options for this range */
+ USHORT usKeyGroup; /* Key grouping for non simultaneous play
+ 0 = no group, 1 up is group
+ for Level 1 only groups 1-15 are allowed */
+ }RGNHEADER, FAR *LPRGNHEADER;
+
+ typedef struct _INSTHEADER {
+ ULONG cRegions; /* Count of regions in this instrument */
+ MIDILOCALE Locale; /* Intended MIDI locale of this instrument */
+ }INSTHEADER, FAR *LPINSTHEADER;
+
+ typedef struct _DLSHEADER {
+ ULONG cInstruments; /* Count of instruments in the collection */
+ }DLSHEADER, FAR *LPDLSHEADER;
+
+ /*
+ definitions for the Wave link structure
+ */
+
+ /***** For level 1 only WAVELINK_CHANNEL_MONO is valid ****
+ ulChannel allows for up to 32 channels of audio with each bit position
+ specifiying a channel of playback */
+
+ #define WAVELINK_CHANNEL_LEFT 0x0001
+ #define WAVELINK_CHANNEL_RIGHT 0x0002
+
+ #define F_WAVELINK_PHASE_MASTER 0x0001
+
+ typedef struct _WAVELINK { /* any paths or links are stored right after struct */
+ USHORT fusOptions; /* options flags for this wave */
+ USHORT usPhaseGroup; /* Phase grouping for locking channels */
+ ULONG ulChannel; /* channel placement */
+ ULONG ulTableIndex; /* index into the wave pool table, 0 based */
+ }WAVELINK, FAR *LPWAVELINK;
+
+ #define POOL_CUE_NULL 0xffffffff
+
+ typedef struct _POOLCUE {
+ // ULONG ulEntryIndex; /* Index entry in the list */
+ ULONG ulOffset; /* Offset to the entry in the list */
+ }POOLCUE, FAR *LPPOOLCUE;
+
+ typedef struct _POOLTABLE {
+ ULONG cbSize; /* size of the pool table structure */
+ ULONG cCues; /* count of cues in the list */
+ } POOLTABLE, FAR *LPPOOLTABLE;
+
+ /*
+ Structures for the "wsmp" chunk
+ */
+
+ #define F_WSMP_NO_TRUNCATION 0x0001
+ #define F_WSMP_NO_COMPRESSION 0x0002
+
+
+ typedef struct _rwsmp {
+ ULONG cbSize;
+ USHORT usUnityNote; /* MIDI Unity Playback Note */
+ SHORT sFineTune; /* Fine Tune in log tuning */
+ LONG lAttenuation; /* Overall Attenuation to be applied to data */
+ ULONG fulOptions; /* Flag options */
+ ULONG cSampleLoops; /* Count of Sample loops, 0 loops is one shot */
+ } WSMPL, FAR *LPWSMPL;
+
+
+ /* This loop type is a normal forward playing loop which is continually
+ played until the envelope reaches an off threshold in the release
+ portion of the volume envelope */
+
+ #define WLOOP_TYPE_FORWARD 0
+
+ typedef struct _rloop {
+ ULONG cbSize;
+ ULONG ulType; /* Loop Type */
+ ULONG ulStart; /* Start of loop in samples */
+ ULONG ulLength; /* Length of loop in samples */
+ } WLOOP, FAR *LPWLOOP;
+
+ #endif /* _INC_DLS */
diff --git a/arm-wt-22k/lib_src/dls2.h b/arm-wt-22k/lib_src/dls2.h new file mode 100644 index 0000000..17d6efd --- /dev/null +++ b/arm-wt-22k/lib_src/dls2.h @@ -0,0 +1,122 @@ +/*
+
+ dls2.h
+
+ Description:
+
+ Interface defines and structures for the DLS2 extensions of DLS.
+
+
+ Written by Microsoft 1998. Released for public use.
+
+*/
+
+#ifndef _INC_DLS2
+#define _INC_DLS2
+
+/*
+ FOURCC's used in the DLS2 file, in addition to DLS1 chunks
+*/
+
+#define FOURCC_RGN2 mmioFOURCC('r','g','n','2')
+#define FOURCC_LAR2 mmioFOURCC('l','a','r','2')
+#define FOURCC_ART2 mmioFOURCC('a','r','t','2')
+#define FOURCC_CDL mmioFOURCC('c','d','l',' ')
+#define FOURCC_DLID mmioFOURCC('d','l','i','d')
+
+/*
+ Articulation connection graph definitions. These are in addition to
+ the definitions in the DLS1 header.
+*/
+
+/* Generic Sources (in addition to DLS1 sources. */
+#define CONN_SRC_POLYPRESSURE 0x0007 /* Polyphonic Pressure */
+#define CONN_SRC_CHANNELPRESSURE 0x0008 /* Channel Pressure */
+#define CONN_SRC_VIBRATO 0x0009 /* Vibrato LFO */
+#define CONN_SRC_MONOPRESSURE 0x000a /* MIDI Mono pressure */
+
+
+/* Midi Controllers */
+#define CONN_SRC_CC91 0x00db /* Reverb Send */
+#define CONN_SRC_CC93 0x00dd /* Chorus Send */
+
+
+/* Generic Destinations */
+#define CONN_DST_GAIN 0x0001 /* Same as CONN_DST_ ATTENUATION, but more appropriate terminology. */
+#define CONN_DST_KEYNUMBER 0x0005 /* Key Number Generator */
+
+/* Audio Channel Output Destinations */
+#define CONN_DST_LEFT 0x0010 /* Left Channel Send */
+#define CONN_DST_RIGHT 0x0011 /* Right Channel Send */
+#define CONN_DST_CENTER 0x0012 /* Center Channel Send */
+#define CONN_DST_LEFTREAR 0x0013 /* Left Rear Channel Send */
+#define CONN_DST_RIGHTREAR 0x0014 /* Right Rear Channel Send */
+#define CONN_DST_LFE_CHANNEL 0x0015 /* LFE Channel Send */
+#define CONN_DST_CHORUS 0x0080 /* Chorus Send */
+#define CONN_DST_REVERB 0x0081 /* Reverb Send */
+
+/* Vibrato LFO Destinations */
+#define CONN_DST_VIB_FREQUENCY 0x0114 /* Vibrato Frequency */
+#define CONN_DST_VIB_STARTDELAY 0x0115 /* Vibrato Start Delay */
+
+/* EG1 Destinations */
+#define CONN_DST_EG1_DELAYTIME 0x020B /* EG1 Delay Time */
+#define CONN_DST_EG1_HOLDTIME 0x020C /* EG1 Hold Time */
+#define CONN_DST_EG1_SHUTDOWNTIME 0x020D /* EG1 Shutdown Time */
+
+
+/* EG2 Destinations */
+#define CONN_DST_EG2_DELAYTIME 0x030F /* EG2 Delay Time */
+#define CONN_DST_EG2_HOLDTIME 0x0310 /* EG2 Hold Time */
+
+
+/* Filter Destinations */
+#define CONN_DST_FILTER_CUTOFF 0x0500 /* Filter Cutoff Frequency */
+#define CONN_DST_FILTER_Q 0x0501 /* Filter Resonance */
+
+
+/* Transforms */
+#define CONN_TRN_CONVEX 0x0002 /* Convex Transform */
+#define CONN_TRN_SWITCH 0x0003 /* Switch Transform */
+
+
+/* Conditional chunk operators */
+ #define DLS_CDL_AND 0x0001 /* X = X & Y */
+ #define DLS_CDL_OR 0x0002 /* X = X | Y */
+ #define DLS_CDL_XOR 0x0003 /* X = X ^ Y */
+ #define DLS_CDL_ADD 0x0004 /* X = X + Y */
+ #define DLS_CDL_SUBTRACT 0x0005 /* X = X - Y */
+ #define DLS_CDL_MULTIPLY 0x0006 /* X = X * Y */
+ #define DLS_CDL_DIVIDE 0x0007 /* X = X / Y */
+ #define DLS_CDL_LOGICAL_AND 0x0008 /* X = X && Y */
+ #define DLS_CDL_LOGICAL_OR 0x0009 /* X = X || Y */
+ #define DLS_CDL_LT 0x000A /* X = (X < Y) */
+ #define DLS_CDL_LE 0x000B /* X = (X <= Y) */
+ #define DLS_CDL_GT 0x000C /* X = (X > Y) */
+ #define DLS_CDL_GE 0x000D /* X = (X >= Y) */
+ #define DLS_CDL_EQ 0x000E /* X = (X == Y) */
+ #define DLS_CDL_NOT 0x000F /* X = !X */
+ #define DLS_CDL_CONST 0x0010 /* 32-bit constant */
+ #define DLS_CDL_QUERY 0x0011 /* 32-bit value returned from query */
+ #define DLS_CDL_QUERYSUPPORTED 0x0012 /* 32-bit value returned from query */
+
+/*
+ Loop and release
+*/
+
+#define WLOOP_TYPE_RELEASE 1
+
+/*
+ DLSID queries for <cdl-ck>
+*/
+DEFINE_DLSID(DLSID_GMInHardware, 0x178f2f24, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_GSInHardware, 0x178f2f25, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_XGInHardware, 0x178f2f26, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_SupportsDLS1, 0x178f2f27, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_SupportsDLS2, 0xf14599e5, 0x4689, 0x11d2, 0xaf, 0xa6, 0x0, 0xaa, 0x0, 0x24, 0xd8, 0xb6);
+DEFINE_DLSID(DLSID_SampleMemorySize, 0x178f2f28, 0xc364, 0x11d1, 0xa7, 0x60, 0x00, 0x00, 0xf8, 0x75, 0xac, 0x12);
+DEFINE_DLSID(DLSID_ManufacturersID, 0xb03e1181, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+DEFINE_DLSID(DLSID_ProductID, 0xb03e1182, 0x8095, 0x11d2, 0xa1, 0xef, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+DEFINE_DLSID(DLSID_SamplePlaybackRate, 0x2a91f713, 0xa4bf, 0x11d2, 0xbb, 0xdf, 0x0, 0x60, 0x8, 0x33, 0xdb, 0xd8);
+#endif /* _INC_DLS2 */
+
diff --git a/arm-wt-22k/lib_src/eas_audioconst.h b/arm-wt-22k/lib_src/eas_audioconst.h new file mode 100644 index 0000000..1cfa404 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_audioconst.h @@ -0,0 +1,97 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_audioconst.h
+ *
+ * Contents and purpose:
+ * Defines audio constants related to the sample rate, bit size, etc.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_AUDIOCONST_H
+#define _EAS_AUDIOCONST_H
+
+/*----------------------------------------------------------------------------
+ * These macros define the various characteristics of the defined sample rates
+ *----------------------------------------------------------------------------
+ * BUFFER_SIZE_IN_MONO_SAMPLES size of buffer in samples
+ * _OUTPUT_SAMPLE_RATE compiled output sample rate
+ * AUDIO_FRAME_LENGTH length of an audio frame in 256ths of a millisecond
+ * SYNTH_UPDATE_PERIOD_IN_BITS length of an audio frame (2^x samples)
+ *----------------------------------------------------------------------------
+*/
+
+#if defined (_SAMPLE_RATE_8000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 32
+#define _OUTPUT_SAMPLE_RATE 8000
+#define AUDIO_FRAME_LENGTH 1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS 5
+
+#elif defined (_SAMPLE_RATE_16000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 64
+#define _OUTPUT_SAMPLE_RATE 16000
+#define AUDIO_FRAME_LENGTH 1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS 6
+
+#elif defined (_SAMPLE_RATE_20000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 128
+#define _OUTPUT_SAMPLE_RATE 20000
+#define AUDIO_FRAME_LENGTH 1638
+#define SYNTH_UPDATE_PERIOD_IN_BITS 7
+
+#elif defined (_SAMPLE_RATE_22050)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 128
+#define _OUTPUT_SAMPLE_RATE 22050
+#define AUDIO_FRAME_LENGTH 1486
+#define SYNTH_UPDATE_PERIOD_IN_BITS 7
+
+#elif defined (_SAMPLE_RATE_24000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 128
+#define _OUTPUT_SAMPLE_RATE 24000
+#define AUDIO_FRAME_LENGTH 1365
+#define SYNTH_UPDATE_PERIOD_IN_BITS 7
+
+#elif defined (_SAMPLE_RATE_32000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 128
+#define _OUTPUT_SAMPLE_RATE 32000
+#define AUDIO_FRAME_LENGTH 1024
+#define SYNTH_UPDATE_PERIOD_IN_BITS 7
+
+#elif defined (_SAMPLE_RATE_44100)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 256
+#define _OUTPUT_SAMPLE_RATE 44100
+#define AUDIO_FRAME_LENGTH 1486
+#define SYNTH_UPDATE_PERIOD_IN_BITS 8
+
+#elif defined (_SAMPLE_RATE_48000)
+#define BUFFER_SIZE_IN_MONO_SAMPLES 256
+#define _OUTPUT_SAMPLE_RATE 48000
+#define AUDIO_FRAME_LENGTH 1365
+#define SYNTH_UPDATE_PERIOD_IN_BITS 8
+
+#else
+#error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
+#endif
+
+#endif /* #ifndef _EAS_AUDIOCONST_H */
+
diff --git a/arm-wt-22k/lib_src/eas_chorus.c b/arm-wt-22k/lib_src/eas_chorus.c new file mode 100644 index 0000000..bc42237 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_chorus.c @@ -0,0 +1,604 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorus.c
+ *
+ * Contents and purpose:
+ * Contains the implementation of the Chorus effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 499 $
+ * $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_effects.h"
+#include "eas_math.h"
+#include "eas_chorusdata.h"
+#include "eas_chorus.h"
+#include "eas_config.h"
+#include "eas_host.h"
+#include "eas_report.h"
+
+/* prototypes for effects interface */
+static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples);
+static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+
+/* common effects interface for configuration module */
+const S_EFFECTS_INTERFACE EAS_Chorus =
+{
+ ChorusInit,
+ ChorusProcess,
+ ChorusShutdown,
+ ChorusGetParam,
+ ChorusSetParam
+};
+
+
+
+//LFO shape table used by the chorus, larger table would sound better
+//this is a sine wave, where 32767 = 1.0
+static const EAS_I16 EAS_chorusShape[CHORUS_SHAPE_SIZE] = {
+ 0, 1608, 3212, 4808, 6393, 7962, 9512, 11309, 12539, 14010, 15446, 16846, 18204, 19519, 20787, 22005, 23170,
+ 24279, 25329, 26319, 27245, 28105, 28898, 29621, 30273, 30852, 31356, 31785, 32137, 32412, 32609, 32728,
+ 32767, 32728, 32609, 32412, 32137, 31785, 31356, 30852, 30273, 29621, 28898, 28105, 27245, 26319, 25329,
+ 24279, 23170, 22005, 20787, 19519, 18204, 16846, 15446, 14010, 12539, 11039, 9512, 7962, 6393, 4808, 3212,
+ 1608, 0, -1608, -3212, -4808, -6393, -7962, -9512, -11309, -12539, -14010, -15446, -16846, -18204, -19519,
+ -20787, -22005, -23170, -24279, -25329, -26319, -27245, -28105, -28898, -29621, -30273, -30852, -31356, -31785,
+ -32137, -32412, -32609, -32728, -32767, -32728, -32609, -32412, -32137, -31785, -31356, -30852, -30273, -29621,
+ -28898, -28105, -27245, -26319, -25329, -24279, -23170, -22005, -20787, -19519, -18204, -16846, -15446, -14010,
+ -12539, -11039, -9512, -7962, -6393, -4808, -3212, -1608
+};
+
+/*----------------------------------------------------------------------------
+ * InitializeChorus()
+ *----------------------------------------------------------------------------
+ * Purpose: Initializes chorus parameters
+ *
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
+{
+ S_CHORUS_OBJECT *pChorusData;
+ S_CHORUS_PRESET *pPreset;
+ EAS_I32 index;
+
+ /* check Configuration Module for data allocation */
+ if (pEASData->staticMemoryModel)
+ pChorusData = EAS_CMEnumFXData(EAS_MODULE_CHORUS);
+
+ /* allocate dynamic memory */
+ else
+ pChorusData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_CHORUS_OBJECT));
+
+ if (pChorusData == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate Chorus memory\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+
+ /* clear the structure */
+ EAS_HWMemSet(pChorusData, 0, sizeof(S_CHORUS_OBJECT));
+
+ ChorusReadInPresets(pChorusData);
+
+ /* set some default values */
+ pChorusData->bypass = EAS_CHORUS_BYPASS_DEFAULT;
+ pChorusData->preset = EAS_CHORUS_PRESET_DEFAULT;
+ pChorusData->m_nLevel = EAS_CHORUS_LEVEL_DEFAULT;
+ pChorusData->m_nRate = EAS_CHORUS_RATE_DEFAULT;
+ pChorusData->m_nDepth = EAS_CHORUS_DEPTH_DEFAULT;
+
+ //chorus rate and depth need some massaging from preset value (which is sample rate independent)
+
+ //convert rate from steps of .05 Hz to value which can be used as phase increment,
+ //with current CHORUS_SHAPE_SIZE and rate limits, this fits into 16 bits
+ //want to compute ((shapeSize * 65536) * (storedRate/20))/sampleRate;
+ //computing it as below allows rate steps to be evenly spaced
+ //uses 32 bit divide, but only once when new value is selected
+ pChorusData->m_nRate = (EAS_I16)
+ ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+ //convert depth from steps of .05 ms, to samples, with 16 bit whole part, discard fraction
+ //want to compute ((depth * sampleRate)/20000)
+ //use the following approximation since 105/32 is roughly 65536/20000
+ /*lint -e{704} use shift for performance */
+ pChorusData->m_nDepth = (EAS_I16)
+ (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+ pChorusData->m_nLevel = pChorusData->m_nLevel;
+
+ //zero delay memory for chorus
+ for (index = CHORUS_L_SIZE - 1; index >= 0; index--)
+ {
+ pChorusData->chorusDelayL[index] = 0;
+ }
+ for (index = CHORUS_R_SIZE - 1; index >= 0; index--)
+ {
+ pChorusData->chorusDelayR[index] = 0;
+ }
+
+ //init delay line index, these are used to implement circular delay buffer
+ pChorusData->chorusIndexL = 0;
+ pChorusData->chorusIndexR = 0;
+
+ //init LFO phase
+ //16 bit whole part, 16 bit fraction
+ pChorusData->lfoLPhase = 0;
+ pChorusData->lfoRPhase = (CHORUS_SHAPE_SIZE << 16) >> 2; // 1/4 of total, i.e. 90 degrees out of phase;
+
+ //init chorus delay position
+ //right now chorus delay is a compile-time value, as is sample rate
+ pChorusData->chorusTapPosition = (EAS_I16)((CHORUS_DELAY_MS * _OUTPUT_SAMPLE_RATE)/1000);
+
+ //now copy from the new preset into Chorus
+ pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus];
+
+ pChorusData->m_nLevel = pPreset->m_nLevel;
+ pChorusData->m_nRate = pPreset->m_nRate;
+ pChorusData->m_nDepth = pPreset->m_nDepth;
+
+ pChorusData->m_nRate = (EAS_I16)
+ ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+ /*lint -e{704} use shift for performance */
+ pChorusData->m_nDepth = (EAS_I16)
+ (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+ *pInstData = pChorusData;
+
+ return EAS_SUCCESS;
+} /* end ChorusInit */
+
+/*----------------------------------------------------------------------------
+ * WeightedTap()
+ *----------------------------------------------------------------------------
+ * Purpose: Does fractional array look-up using linear interpolation
+ *
+ * first convert indexDesired to actual desired index by taking into account indexReference
+ * then do linear interpolation between two actual samples using fractional part
+ *
+ * Inputs:
+ * array: pointer to array of signed 16 bit values, typically either PCM data or control data
+ * indexReference: the circular buffer relative offset
+ * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction)
+ * indexLimit: the total size of the array, used to compute buffer wrap
+ *
+ * Outputs:
+ * Value from the input array, linearly interpolated between two actual data values
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit)
+{
+ EAS_I16 index;
+ EAS_I16 fraction;
+ EAS_I16 val1;
+ EAS_I16 val2;
+
+ //separate indexDesired into whole and fractional parts
+ /*lint -e{704} use shift for performance */
+ index = (EAS_I16)(indexDesired >> 16);
+ /*lint -e{704} use shift for performance */
+ fraction = (EAS_I16)((indexDesired>>1) & 0x07FFF); //just use 15 bits of fractional part
+
+ //adjust whole part by indexReference
+ index = indexReference - index;
+ //make sure we stay within array bounds, this implements circular buffer
+ while (index < 0)
+ {
+ index += indexLimit;
+ }
+
+ //get two adjacent values from the array
+ val1 = array[index];
+
+ //handle special case when index == 0, else typical case
+ if (index == 0)
+ {
+ val2 = array[indexLimit-1]; //get last value from array
+ }
+ else
+ {
+ val2 = array[index-1]; //get previous value from array
+ }
+
+ //compute linear interpolation as (val1 + ((val2-val1)*fraction))
+ return(val1 + (EAS_I16)MULT_EG1_EG1(val2-val1,fraction));
+}
+
+/*----------------------------------------------------------------------------
+ * ChorusProcess()
+ *----------------------------------------------------------------------------
+ * Purpose: compute the chorus on the input buffer, and mix into output buffer
+ *
+ *
+ * Inputs:
+ * src: pointer to input buffer of PCM values to be processed
+ * dst: pointer to output buffer of PCM values we are to sume the result with
+ * bufSize: the number of sample frames (i.e. stereo samples) in the buffer
+ *
+ * Outputs:
+ * None
+ *
+ *----------------------------------------------------------------------------
+*/
+//compute the chorus, and mix into output buffer
+static void ChorusProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples)
+{
+ EAS_I32 ix;
+ EAS_I32 nChannelNumber;
+ EAS_I16 lfoValueLeft;
+ EAS_I16 lfoValueRight;
+ EAS_I32 positionOffsetL;
+ EAS_I32 positionOffsetR;
+ EAS_PCM tapL;
+ EAS_PCM tapR;
+ EAS_I32 tempValue;
+ EAS_PCM nInputSample;
+ EAS_I32 nOutputSample;
+ EAS_PCM *pIn;
+ EAS_PCM *pOut;
+
+ S_CHORUS_OBJECT *pChorusData;
+
+ pChorusData = (S_CHORUS_OBJECT*) pInstData;
+
+ //if the chorus is disabled or turned all the way down
+ if (pChorusData->bypass == EAS_TRUE || pChorusData->m_nLevel == 0)
+ {
+ if (pSrc != pDst)
+ EAS_HWMemCpy(pSrc, pDst, numSamples * NUM_OUTPUT_CHANNELS * (EAS_I32) sizeof(EAS_PCM));
+ return;
+ }
+
+ if (pChorusData->m_nNextChorus != pChorusData->m_nCurrentChorus)
+ {
+ ChorusUpdate(pChorusData);
+ }
+
+ for (nChannelNumber = 0; nChannelNumber < NUM_OUTPUT_CHANNELS; nChannelNumber++)
+ {
+
+ pIn = pSrc + nChannelNumber;
+ pOut = pDst + nChannelNumber;
+
+ if(nChannelNumber==0)
+ {
+ for (ix = 0; ix < numSamples; ix++)
+ {
+ nInputSample = *pIn;
+ pIn += NUM_OUTPUT_CHANNELS;
+
+ //feed input into chorus delay line
+ pChorusData->chorusDelayL[pChorusData->chorusIndexL] = nInputSample;
+
+ //compute chorus lfo value using phase as fractional index into chorus shape table
+ //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number
+ lfoValueLeft = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoLPhase, CHORUS_SHAPE_SIZE);
+
+ //scale chorus depth by lfo value to get relative fractional sample index
+ //index is expressed as 32 bit number with 16 bit fractional part
+ /*lint -e{703} use shift for performance */
+ positionOffsetL = pChorusData->m_nDepth * (((EAS_I32)lfoValueLeft) << 1);
+
+ //add fixed chorus delay to get actual fractional sample index
+ positionOffsetL += ((EAS_I32)pChorusData->chorusTapPosition) << 16;
+
+ //get tap value from chorus delay using fractional sample index
+ tapL = WeightedTap(pChorusData->chorusDelayL, pChorusData->chorusIndexL, positionOffsetL, CHORUS_L_SIZE);
+
+ //scale by chorus level, then sum with input buffer contents and saturate
+ tempValue = MULT_EG1_EG1(tapL, pChorusData->m_nLevel);
+ nOutputSample = SATURATE(tempValue + nInputSample);
+
+ *pOut = (EAS_I16)SATURATE(nOutputSample);
+ pOut += NUM_OUTPUT_CHANNELS;
+
+
+ //increment chorus delay index and make it wrap as needed
+ //this implements circular buffer
+ if ((pChorusData->chorusIndexL+=1) >= CHORUS_L_SIZE)
+ pChorusData->chorusIndexL = 0;
+
+ //increment fractional lfo phase, and make it wrap as needed
+ pChorusData->lfoLPhase += pChorusData->m_nRate;
+ while (pChorusData->lfoLPhase >= (CHORUS_SHAPE_SIZE<<16))
+ {
+ pChorusData->lfoLPhase -= (CHORUS_SHAPE_SIZE<<16);
+ }
+ }
+ }
+ else
+ {
+ for (ix = 0; ix < numSamples; ix++)
+ {
+ nInputSample = *pIn;
+ pIn += NUM_OUTPUT_CHANNELS;
+
+ //feed input into chorus delay line
+ pChorusData->chorusDelayR[pChorusData->chorusIndexR] = nInputSample;
+
+ //compute chorus lfo value using phase as fractional index into chorus shape table
+ //resulting value is between -1.0 and 1.0, expressed as signed 16 bit number
+ lfoValueRight = WeightedTap(EAS_chorusShape, 0, pChorusData->lfoRPhase, CHORUS_SHAPE_SIZE);
+
+ //scale chorus depth by lfo value to get relative fractional sample index
+ //index is expressed as 32 bit number with 16 bit fractional part
+ /*lint -e{703} use shift for performance */
+ positionOffsetR = pChorusData->m_nDepth * (((EAS_I32)lfoValueRight) << 1);
+
+ //add fixed chorus delay to get actual fractional sample index
+ positionOffsetR += ((EAS_I32)pChorusData->chorusTapPosition) << 16;
+
+ //get tap value from chorus delay using fractional sample index
+ tapR = WeightedTap(pChorusData->chorusDelayR, pChorusData->chorusIndexR, positionOffsetR, CHORUS_R_SIZE);
+
+ //scale by chorus level, then sum with output buffer contents and saturate
+ tempValue = MULT_EG1_EG1(tapR, pChorusData->m_nLevel);
+ nOutputSample = SATURATE(tempValue + nInputSample);
+
+ *pOut = (EAS_I16)SATURATE(nOutputSample);
+ pOut += NUM_OUTPUT_CHANNELS;
+
+ //increment chorus delay index and make it wrap as needed
+ //this implements circular buffer
+ if ((pChorusData->chorusIndexR+=1) >= CHORUS_R_SIZE)
+ pChorusData->chorusIndexR = 0;
+
+ //increment fractional lfo phase, and make it wrap as needed
+ pChorusData->lfoRPhase += pChorusData->m_nRate;
+ while (pChorusData->lfoRPhase >= (CHORUS_SHAPE_SIZE<<16))
+ {
+ pChorusData->lfoRPhase -= (CHORUS_SHAPE_SIZE<<16);
+ }
+ }
+ }
+
+ }
+} /* end ChorusProcess */
+
+
+
+/*----------------------------------------------------------------------------
+ * ChorusShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the Chorus effect.
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData)
+{
+ /* check Configuration Module for static memory allocation */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pInstData);
+ return EAS_SUCCESS;
+} /* end ChorusShutdown */
+
+/*----------------------------------------------------------------------------
+ * ChorusGetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get a Chorus parameter
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ * param - parameter index
+ * *pValue - pointer to variable to hold retrieved value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_CHORUS_OBJECT *p;
+
+ p = (S_CHORUS_OBJECT*) pInstData;
+
+ switch (param)
+ {
+ case EAS_PARAM_CHORUS_BYPASS:
+ *pValue = (EAS_I32) p->bypass;
+ break;
+ case EAS_PARAM_CHORUS_PRESET:
+ *pValue = (EAS_I8) p->m_nCurrentChorus;
+ break;
+ case EAS_PARAM_CHORUS_RATE:
+ *pValue = (EAS_I32) p->m_nRate;
+ break;
+ case EAS_PARAM_CHORUS_DEPTH:
+ *pValue = (EAS_I32) p->m_nDepth;
+ break;
+ case EAS_PARAM_CHORUS_LEVEL:
+ *pValue = (EAS_I32) p->m_nLevel;
+ break;
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+} /* end ChorusGetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ChorusSetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set a Chorus parameter
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ * param - parameter index
+ * *pValue - new paramter value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_CHORUS_OBJECT *p;
+
+ p = (S_CHORUS_OBJECT*) pInstData;
+
+ switch (param)
+ {
+ case EAS_PARAM_CHORUS_BYPASS:
+ p->bypass = (EAS_BOOL) value;
+ break;
+ case EAS_PARAM_CHORUS_PRESET:
+ if(value!=EAS_PARAM_CHORUS_PRESET1 && value!=EAS_PARAM_CHORUS_PRESET2 &&
+ value!=EAS_PARAM_CHORUS_PRESET3 && value!=EAS_PARAM_CHORUS_PRESET4)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nNextChorus = (EAS_I8)value;
+ break;
+ case EAS_PARAM_CHORUS_RATE:
+ if(value<EAS_CHORUS_RATE_MIN || value>EAS_CHORUS_RATE_MAX)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nRate = (EAS_I16) value;
+ break;
+ case EAS_PARAM_CHORUS_DEPTH:
+ if(value<EAS_CHORUS_DEPTH_MIN || value>EAS_CHORUS_DEPTH_MAX)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nDepth = (EAS_I16) value;
+ break;
+ case EAS_PARAM_CHORUS_LEVEL:
+ if(value<EAS_CHORUS_LEVEL_MIN || value>EAS_CHORUS_LEVEL_MAX)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nLevel = (EAS_I16) value;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+} /* end ChorusSetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ChorusReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global Chorus preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData)
+{
+
+ int preset = 0;
+ int defaultPreset = 0;
+
+ //now init any remaining presets to defaults
+ for (defaultPreset = preset; defaultPreset < CHORUS_MAX_TYPE; defaultPreset++)
+ {
+ S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[defaultPreset];
+ if (defaultPreset == 0 || defaultPreset > CHORUS_MAX_TYPE-1)
+ {
+ pPreset->m_nDepth = 39;
+ pPreset->m_nRate = 30;
+ pPreset->m_nLevel = 32767;
+ }
+ else if (defaultPreset == 1)
+ {
+ pPreset->m_nDepth = 21;
+ pPreset->m_nRate = 45;
+ pPreset->m_nLevel = 25000;
+ }
+ else if (defaultPreset == 2)
+ {
+ pPreset->m_nDepth = 53;
+ pPreset->m_nRate = 25;
+ pPreset->m_nLevel = 32000;
+ }
+ else if (defaultPreset == 3)
+ {
+ pPreset->m_nDepth = 32;
+ pPreset->m_nRate = 37;
+ pPreset->m_nLevel = 29000;
+ }
+ }
+
+ return EAS_SUCCESS;
+}
+
+
+/*----------------------------------------------------------------------------
+ * ChorusUpdate
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the Chorus preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - chorus paramters will be changed
+ * - m_nCurrentRoom := m_nNextRoom
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT *pChorusData)
+{
+ S_CHORUS_PRESET *pPreset = &pChorusData->m_sPreset.m_sPreset[pChorusData->m_nNextChorus];
+
+ pChorusData->m_nLevel = pPreset->m_nLevel;
+ pChorusData->m_nRate = pPreset->m_nRate;
+ pChorusData->m_nDepth = pPreset->m_nDepth;
+
+ pChorusData->m_nRate = (EAS_I16)
+ ((((EAS_I32)CHORUS_SHAPE_SIZE<<16)/(20*(EAS_I32)_OUTPUT_SAMPLE_RATE)) * pChorusData->m_nRate);
+
+ /*lint -e{704} use shift for performance */
+ pChorusData->m_nDepth = (EAS_I16)
+ (((((EAS_I32)pChorusData->m_nDepth * _OUTPUT_SAMPLE_RATE)>>5) * 105) >> 16);
+
+ pChorusData->m_nCurrentChorus = pChorusData->m_nNextChorus;
+
+ return EAS_SUCCESS;
+
+} /* end ChorusUpdate */
diff --git a/arm-wt-22k/lib_src/eas_chorusdata.c b/arm-wt-22k/lib_src/eas_chorusdata.c new file mode 100644 index 0000000..caee1ed --- /dev/null +++ b/arm-wt-22k/lib_src/eas_chorusdata.c @@ -0,0 +1,34 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorusdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data allocation for the Chorus effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 550 $
+ * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_chorusdata.h"
+
+S_CHORUS_OBJECT eas_ChorusData;
+
diff --git a/arm-wt-22k/lib_src/eas_chorusdata.h b/arm-wt-22k/lib_src/eas_chorusdata.h new file mode 100644 index 0000000..4420ddd --- /dev/null +++ b/arm-wt-22k/lib_src/eas_chorusdata.h @@ -0,0 +1,160 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_chorusdata.h
+ *
+ * Contents and purpose:
+ * Contains the prototypes for the Chorus effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 309 $
+ * $Date: 2006-09-12 18:52:45 -0700 (Tue, 12 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_CHORUS_H
+#define _EAS_CHORUS_H
+
+#include "eas_types.h"
+#include "eas_audioconst.h"
+
+//defines for chorus
+
+#define EAS_CHORUS_BYPASS_DEFAULT 1
+#define EAS_CHORUS_PRESET_DEFAULT 0
+#define EAS_CHORUS_RATE_DEFAULT 30
+#define EAS_CHORUS_DEPTH_DEFAULT 39
+#define EAS_CHORUS_LEVEL_DEFAULT 32767
+
+#define EAS_CHORUS_LEVEL_MIN 0
+#define EAS_CHORUS_LEVEL_MAX 32767
+
+#define EAS_CHORUS_RATE_MIN 10
+#define EAS_CHORUS_RATE_MAX 50
+
+#define EAS_CHORUS_DEPTH_MIN 15
+#define EAS_CHORUS_DEPTH_MAX 60
+
+#define CHORUS_SIZE_MS 20
+#define CHORUS_L_SIZE ((CHORUS_SIZE_MS*_OUTPUT_SAMPLE_RATE)/1000)
+#define CHORUS_R_SIZE CHORUS_L_SIZE
+#define CHORUS_SHAPE_SIZE 128
+#define CHORUS_DELAY_MS 10
+
+#define CHORUS_MAX_TYPE 4 // any Chorus numbers larger than this are invalid
+
+typedef struct
+{
+ EAS_I16 m_nRate;
+ EAS_I16 m_nDepth;
+ EAS_I16 m_nLevel;
+
+} S_CHORUS_PRESET;
+
+typedef struct
+{
+ S_CHORUS_PRESET m_sPreset[CHORUS_MAX_TYPE]; //array of presets
+
+} S_CHORUS_PRESET_BANK;
+
+/* parameters for each Chorus */
+typedef struct
+{
+ EAS_I32 lfoLPhase;
+ EAS_I32 lfoRPhase;
+ EAS_I16 chorusIndexL;
+ EAS_I16 chorusIndexR;
+ EAS_U16 chorusTapPosition;
+
+ EAS_I16 m_nRate;
+ EAS_I16 m_nDepth;
+ EAS_I16 m_nLevel;
+
+ //delay lines used by the chorus, longer would sound better
+ EAS_PCM chorusDelayL[CHORUS_L_SIZE];
+ EAS_PCM chorusDelayR[CHORUS_R_SIZE];
+
+ EAS_BOOL bypass;
+ EAS_I8 preset;
+
+ EAS_I16 m_nCurrentChorus; // preset number for current Chorus
+ EAS_I16 m_nNextChorus; // preset number for next Chorus
+
+ S_CHORUS_PRESET pPreset;
+
+ S_CHORUS_PRESET_BANK m_sPreset;
+
+} S_CHORUS_OBJECT;
+
+
+/*----------------------------------------------------------------------------
+ * WeightedTap()
+ *----------------------------------------------------------------------------
+ * Purpose: Does fractional array look-up using linear interpolation
+ *
+ * first convert indexDesired to actual desired index by taking into account indexReference
+ * then do linear interpolation between two actual samples using fractional part
+ *
+ * Inputs:
+ * array: pointer to array of signed 16 bit values, typically either PCM data or control data
+ * indexReference: the circular buffer relative offset
+ * indexDesired: the fractional index we are looking up (16 bits index + 16 bits fraction)
+ * indexLimit: the total size of the array, used to compute buffer wrap
+ *
+ * Outputs:
+ * Value from the input array, linearly interpolated between two actual data values
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_I16 WeightedTap(const EAS_I16 *array, EAS_I16 indexReference, EAS_I32 indexDesired, EAS_I16 indexLimit);
+
+/*----------------------------------------------------------------------------
+ * ChorusReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global Chorus preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusReadInPresets(S_CHORUS_OBJECT *pChorusData);
+
+/*----------------------------------------------------------------------------
+ * ChorusUpdate
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the Chorus preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - chorus paramters will be changed
+ * - m_nCurrentChorus := m_nNextChorus
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ChorusUpdate(S_CHORUS_OBJECT* pChorusData);
+
+#endif /* #ifndef _EAS_CHORUSDATA_H */
+
+
diff --git a/arm-wt-22k/lib_src/eas_ctype.h b/arm-wt-22k/lib_src/eas_ctype.h new file mode 100644 index 0000000..8503870 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_ctype.h @@ -0,0 +1,41 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_ctype.h
+ *
+ * Contents and purpose:
+ * This is a replacement for the CRT ctype.h functions. These
+ * functions are currently ASCII only, but eventually, we will want
+ * to support wide-characters for localization.
+ *
+ * Copyright (c) 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 429 $
+ * $Date: 2006-10-19 23:50:15 -0700 (Thu, 19 Oct 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_CTYPE_H
+#define _EAS_CTYPE_H
+
+EAS_INLINE EAS_I8 IsDigit (EAS_I8 c) { return ((c >= '0') && (c <= '9')); }
+EAS_INLINE EAS_I8 IsSpace (EAS_I8 c) { return (((c >= 9) && (c <= 13)) || (c == ' ')); }
+EAS_INLINE EAS_I8 ToUpper (EAS_I8 c) { if ((c >= 'a') && (c <= 'z')) return c & ~0x20; else return c; }
+EAS_INLINE EAS_I8 ToLower (EAS_I8 c) { if ((c >= 'A') && (c <= 'Z')) return c |= 0x20; else return c; }
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_data.c b/arm-wt-22k/lib_src/eas_data.c new file mode 100644 index 0000000..bb60ef2 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_data.c @@ -0,0 +1,37 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_data.c
+ *
+ * Contents and purpose:
+ * Contains a data allocation for synthesizer
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+// includes
+#include "eas_data.h"
+
+// globals
+S_EAS_DATA eas_Data;
+S_VOICE_MGR eas_Synth;
+S_SYNTH eas_MIDI;
+
diff --git a/arm-wt-22k/lib_src/eas_data.h b/arm-wt-22k/lib_src/eas_data.h new file mode 100644 index 0000000..28957fd --- /dev/null +++ b/arm-wt-22k/lib_src/eas_data.h @@ -0,0 +1,133 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_data.h
+ *
+ * Contents and purpose:
+ * This header defines all types, to support dynamic allocation of the
+ * memory resources needed for persistent EAS data.
+ *
+ * Copyright 2004 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 842 $
+ * $Date: 2007-08-23 14:32:31 -0700 (Thu, 23 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_DATA_H
+#define _EAS_DATA_H
+
+#define JET_INTERFACE
+
+#include "eas_types.h"
+#include "eas_synthcfg.h"
+#include "eas.h"
+#include "eas_audioconst.h"
+#include "eas_sndlib.h"
+#include "eas_pcm.h"
+#include "eas_pcmdata.h"
+#include "eas_synth.h"
+#include "eas_miditypes.h"
+#include "eas_effects.h"
+
+#ifdef AUX_MIXER
+#include "eas_auxmixdata.h"
+#endif
+
+#ifdef JET_INTERFACE
+#include "jet.h"
+#endif
+
+#ifdef _METRICS_ENABLED
+#include "eas_perf.h"
+#endif
+
+#ifndef MAX_NUMBER_STREAMS
+#define MAX_NUMBER_STREAMS 4
+#endif
+
+/* flags for S_EAS_STREAM */
+#define STREAM_FLAGS_PARSED 1
+#define STREAM_FLAGS_PAUSE 2
+#define STREAM_FLAGS_LOCATE 4
+#define STREAM_FLAGS_RESUME 8
+
+/* structure for parsing a stream */
+typedef struct s_eas_stream_tag
+{
+ void *pParserModule;
+ EAS_U32 time;
+ EAS_U32 frameLength;
+ EAS_I32 repeatCount;
+ EAS_VOID_PTR handle;
+ EAS_U8 volume;
+ EAS_BOOL8 streamFlags;
+} S_EAS_STREAM;
+
+/* default master volume is -10dB */
+#define DEFAULT_VOLUME 90
+#define DEFAULT_STREAM_VOLUME 100
+#define DEFAULT_STREAM_GAIN 14622
+
+/* 10 dB of boost available for individual parsers */
+#define STREAM_VOLUME_HEADROOM 10
+
+/* amalgamated persistent data type */
+typedef struct s_eas_data_tag
+{
+#ifdef _CHECKED_BUILD
+ EAS_U32 handleCheck;
+#endif
+ EAS_HW_DATA_HANDLE hwInstData;
+
+ S_EFFECTS_MODULE effectsModules[NUM_EFFECTS_MODULES];
+
+#ifdef _METRICS_ENABLED
+ S_METRICS_INTERFACE *pMetricsModule;
+ EAS_VOID_PTR pMetricsData;
+#endif
+
+ EAS_I32 *pMixBuffer;
+ EAS_PCM *pOutputAudioBuffer;
+
+#ifdef AUX_MIXER
+ S_EAS_AUX_MIXER auxMixer;
+#endif
+
+#ifdef _MAXIMIZER_ENABLED
+ EAS_VOID_PTR pMaximizerData;
+#endif
+
+ S_EAS_STREAM streams[MAX_NUMBER_STREAMS];
+
+ S_PCM_STATE *pPCMStreams;
+
+ S_VOICE_MGR *pVoiceMgr;
+
+#ifdef JET_INTERFACE
+ JET_DATA_HANDLE jetHandle;
+#endif
+
+ EAS_U32 renderTime;
+ EAS_I16 masterGain;
+ EAS_U8 masterVolume;
+ EAS_BOOL8 staticMemoryModel;
+ EAS_BOOL8 searchHeaderFlag;
+} S_EAS_DATA;
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_dlssynth.c b/arm-wt-22k/lib_src/eas_dlssynth.c new file mode 100644 index 0000000..c1fa1ef --- /dev/null +++ b/arm-wt-22k/lib_src/eas_dlssynth.c @@ -0,0 +1,578 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_dlssynth.c + * + * Contents and purpose: + * Implements the Mobile DLS synthesizer. + * + * Copyright Sonic Network Inc. 2006 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 795 $ + * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +// includes +#include "eas_data.h" +#include "eas_report.h" +#include "eas_host.h" +#include "eas_math.h" +#include "eas_synth_protos.h" +#include "eas_wtsynth.h" +#include "eas_pan.h" +#include "eas_mdls.h" +#include "eas_dlssynth.h" + +#ifdef _METRICS_ENABLED +#include "eas_perf.h" +#endif + +static void DLS_UpdateEnvelope (S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, const S_DLS_ENVELOPE *pEnvParams, EAS_I16 *pValue, EAS_I16 *pIncrement, EAS_U8 *pState); + +/*---------------------------------------------------------------------------- + * DLS_MuteVoice() + *---------------------------------------------------------------------------- + * Mute the voice using shutdown time from the DLS articulation data + *---------------------------------------------------------------------------- +*/ +void DLS_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) +{ + S_WT_VOICE *pWTVoice; + const S_DLS_ARTICULATION *pDLSArt; + + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex]; + + /* clear deferred action flags */ + pVoice->voiceFlags &= + ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF | + VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF | + VOICE_FLAG_DEFER_MUTE); + + /* set the envelope state */ + pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateRelease; + pWTVoice->eg1Increment = pDLSArt->eg1ShutdownTime; + pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateRelease; + pWTVoice->eg2Increment = pDLSArt->eg2.releaseTime; +} + +/*---------------------------------------------------------------------------- + * DLS_ReleaseVoice() + *---------------------------------------------------------------------------- + * Release the selected voice. + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pVoice) standard API, pVoice may be used by other synthesizers */ +void DLS_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) +{ + S_WT_VOICE *pWTVoice; + const S_DLS_ARTICULATION *pDLSArt; + + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex]; + + /* if still in attack phase, convert units to log */ + /*lint -e{732} eg1Value is never negative */ + /*lint -e{703} use shift for performance */ + if (pWTVoice->eg1State == eEnvelopeStateAttack) + pWTVoice->eg1Value = (EAS_I16) ((EAS_flog2(pWTVoice->eg1Value) << 1) + 2048); + + /* release EG1 */ + pWTVoice->eg1State = eEnvelopeStateRelease; + pWTVoice->eg1Increment = pDLSArt->eg1.releaseTime; + + /* release EG2 */ + pWTVoice->eg2State = eEnvelopeStateRelease; + pWTVoice->eg2Increment = pDLSArt->eg2.releaseTime; +} + +/*---------------------------------------------------------------------------- + * DLS_SustainPedal() + *---------------------------------------------------------------------------- + * The sustain pedal was just depressed. If the voice is still + * above the sustain level, catch the voice and continue holding. + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pChannel) pChannel reserved for future use */ +void DLS_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum) +{ + S_WT_VOICE *pWTVoice; + const S_DLS_ARTICULATION *pDLSArt; + + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex]; + + /* don't catch the voice if below the sustain level */ + if (pWTVoice->eg1Value < pDLSArt->eg1.sustainLevel) + return; + + /* defer releasing this note until the damper pedal is off */ + pWTVoice->eg1State = eEnvelopeStateDecay; + pVoice->voiceState = eVoiceStatePlay; + pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; + +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "DLS_SustainPedal: defer note off because sustain pedal is on\n"); */ } +#endif +} + +/*---------------------------------------------------------------------------- + * DLS_UpdatePhaseInc() + *---------------------------------------------------------------------------- + * Calculate the oscillator phase increment for the next frame + *---------------------------------------------------------------------------- +*/ +static EAS_I32 DLS_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_DLS_ARTICULATION *pDLSArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents) +{ + EAS_I32 temp; + + /* start with base mod LFO modulation */ + temp = pDLSArt->modLFOToPitch; + + /* add mod wheel effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOCC1ToPitch * pChannel->modWheel) >> 7); + + /* add channel pressure effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOChanPressToPitch * pChannel->channelPressure) >> 7); + + /* add total mod LFO effect */ + pitchCents += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue); + + /* start with base vib LFO modulation */ + temp = pDLSArt->vibLFOToPitch; + + /* add mod wheel effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->vibLFOCC1ToPitch * pChannel->modWheel) >> 7); + + /* add channel pressure effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->vibLFOChanPressToPitch * pChannel->channelPressure) >> 7); + + /* add total vibrato LFO effect */ + pitchCents += FMUL_15x15(temp, pWTVoice->vibLFO.lfoValue); + + /* add EG2 effect */ + pitchCents += FMUL_15x15(pDLSArt->eg2ToPitch, pWTVoice->eg2Value); + + /* convert from cents to linear phase increment */ + return EAS_Calculate2toX(pitchCents); +} + +/*---------------------------------------------------------------------------- + * DLS_UpdateGain() + *---------------------------------------------------------------------------- + * Calculate the gain for the next frame + *---------------------------------------------------------------------------- +*/ +static EAS_I32 DLS_UpdateGain (S_WT_VOICE *pWTVoice, const S_DLS_ARTICULATION *pDLSArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain, EAS_U8 velocity) +{ + EAS_I32 temp; + + /* start with base mod LFO modulation */ + temp = pDLSArt->modLFOToGain; + + /* add mod wheel effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOCC1ToGain * pChannel->modWheel) >> 7); + + /* add channel pressure effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOChanPressToGain * pChannel->channelPressure) >> 7); + + /* add total mod LFO effect */ + gain += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue); + if (gain > 0) + gain = 0; + + /* convert to linear gain including EG1 */ + if (pWTVoice->eg1State != eEnvelopeStateAttack) + { + gain = (DLS_GAIN_FACTOR * gain) >> DLS_GAIN_SHIFT; + /*lint -e{702} use shift for performance */ +#if 1 + gain += (pWTVoice->eg1Value - 32767) >> 1; + gain = EAS_LogToLinear16(gain); +#else + gain = EAS_LogToLinear16(gain); + temp = EAS_LogToLinear16((pWTVoice->eg1Value - 32767) >> 1); + gain = FMUL_15x15(gain, temp); +#endif + } + else + { + gain = (DLS_GAIN_FACTOR * gain) >> DLS_GAIN_SHIFT; + gain = EAS_LogToLinear16(gain); + gain = FMUL_15x15(gain, pWTVoice->eg1Value); + } + + /* include MIDI channel gain */ + gain = FMUL_15x15(gain, pChannel->staticGain); + + /* include velocity */ + if (pDLSArt->filterQandFlags & FLAG_DLS_VELOCITY_SENSITIVE) + { + temp = velocity << 8; + temp = FMUL_15x15(temp, temp); + gain = FMUL_15x15(gain, temp); + } + + /* return gain */ + return gain; +} + +/*---------------------------------------------------------------------------- + * DLS_UpdateFilter() + *---------------------------------------------------------------------------- + * Update the Filter parameters + *---------------------------------------------------------------------------- +*/ +static void DLS_UpdateFilter (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, S_SYNTH_CHANNEL *pChannel, const S_DLS_ARTICULATION *pDLSArt) +{ + EAS_I32 cutoff; + EAS_I32 temp; + + /* no need to calculate filter coefficients if it is bypassed */ + if (pDLSArt->filterCutoff == DEFAULT_DLS_FILTER_CUTOFF_FREQUENCY) + { + pIntFrame->frame.k = 0; + return; + } + + /* start with base cutoff frequency */ + cutoff = pDLSArt->filterCutoff; + + /* get base mod LFO modulation */ + temp = pDLSArt->modLFOToFc; + + /* add mod wheel effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOCC1ToFc * pChannel->modWheel) >> 7); + + /* add channel pressure effect */ + /*lint -e{702} use shift for performance */ + temp += ((pDLSArt->modLFOChanPressToFc* pChannel->channelPressure) >> 7); + + /* add total mod LFO effect */ + cutoff += FMUL_15x15(temp, pWTVoice->modLFO.lfoValue); + + /* add EG2 effect */ + cutoff += FMUL_15x15(pWTVoice->eg2Value, pDLSArt->eg2ToFc); + + /* add velocity effect */ + /*lint -e{702} use shift for performance */ + cutoff += (pVoice->velocity * pDLSArt->velToFc) >> 7; + + /* add velocity effect */ + /*lint -e{702} use shift for performance */ + cutoff += (pVoice->note * pDLSArt->keyNumToFc) >> 7; + + /* subtract the A5 offset and the sampling frequency */ + cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS; + + /* limit the cutoff frequency */ + if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS) + cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS; + else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS) + cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS; + + WT_SetFilterCoeffs(pIntFrame, cutoff, pDLSArt->filterQandFlags & FILTER_Q_MASK); +} + +/*---------------------------------------------------------------------------- + * DLS_StartVoice() + *---------------------------------------------------------------------------- + * Start up a DLS voice + *---------------------------------------------------------------------------- +*/ +EAS_RESULT DLS_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex) +{ + S_WT_VOICE *pWTVoice; + const S_DLS_REGION *pDLSRegion; + const S_DLS_ARTICULATION *pDLSArt; + S_SYNTH_CHANNEL *pChannel; + +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "DLS_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ } +#endif + + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pChannel = &pSynth->channels[pVoice->channel & 15]; + pDLSRegion = &pSynth->pDLS->pDLSRegions[regionIndex & REGION_INDEX_MASK]; + pWTVoice->artIndex = pDLSRegion->wtRegion.artIndex; + pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex]; + + /* initialize the envelopes */ + pWTVoice->eg1State = eEnvelopeStateInit; + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State); + pWTVoice->eg2State = eEnvelopeStateInit; + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State); + + /* initialize the LFOs */ + pWTVoice->modLFO.lfoValue = 0; + pWTVoice->modLFO.lfoPhase = pDLSArt->modLFO.lfoDelay; + pWTVoice->vibLFO.lfoValue = 0; + pWTVoice->vibLFO.lfoPhase = pDLSArt->vibLFO.lfoDelay; + + /* initalize the envelopes and calculate initial gain */ + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State); + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State); + pVoice->gain = (EAS_I16) DLS_UpdateGain(pWTVoice, pDLSArt, pChannel, pDLSRegion->wtRegion.gain, pVoice->velocity); + +#if (NUM_OUTPUT_CHANNELS == 2) + EAS_CalcPanControl((EAS_INT) pChannel->pan - 64 + (EAS_INT) pDLSArt->pan, &pWTVoice->gainLeft, &pWTVoice->gainRight); +#endif + + /* initialize the filter states */ + pWTVoice->filter.z1 = 0; + pWTVoice->filter.z2 = 0; + + /* initialize the oscillator */ + pWTVoice->phaseAccum = (EAS_U32) pSynth->pDLS->pDLSSamples + pSynth->pDLS->pDLSSampleOffsets[pDLSRegion->wtRegion.waveIndex]; + if (pDLSRegion->wtRegion.region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED) + { + pWTVoice->loopStart = pWTVoice->phaseAccum + pDLSRegion->wtRegion.loopStart; + pWTVoice->loopEnd = pWTVoice->phaseAccum + pDLSRegion->wtRegion.loopEnd - 1; + } + else + pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pDLS->pDLSSampleLen[pDLSRegion->wtRegion.waveIndex] - 1; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * DLS_UpdateVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesize a block of samples for the given voice. + * Use linear interpolation. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * number of samples actually written to buffer + * + * Side Effects: + * - samples are added to the presently free buffer + * + *---------------------------------------------------------------------------- +*/ +EAS_BOOL DLS_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples) +{ + S_WT_VOICE *pWTVoice; + S_SYNTH_CHANNEL *pChannel; + const S_DLS_REGION *pDLSRegion; + const S_DLS_ARTICULATION *pDLSArt; + S_WT_INT_FRAME intFrame; + EAS_I32 temp; + EAS_BOOL done = EAS_FALSE; + + /* establish pointers to critical data */ + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pDLSRegion = &pSynth->pDLS->pDLSRegions[pVoice->regionIndex & REGION_INDEX_MASK]; + pChannel = &pSynth->channels[pVoice->channel & 15]; + pDLSArt = &pSynth->pDLS->pDLSArticulations[pWTVoice->artIndex]; + + /* update the envelopes */ + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg1, &pWTVoice->eg1Value, &pWTVoice->eg1Increment, &pWTVoice->eg1State); + DLS_UpdateEnvelope(pVoice, pChannel, &pDLSArt->eg2, &pWTVoice->eg2Value, &pWTVoice->eg2Increment, &pWTVoice->eg2State); + + /* update the LFOs using the EAS synth function */ + WT_UpdateLFO(&pWTVoice->modLFO, pDLSArt->modLFO.lfoFreq); + WT_UpdateLFO(&pWTVoice->vibLFO, pDLSArt->vibLFO.lfoFreq); + + /* calculate base frequency */ + temp = pDLSArt->tuning + pChannel->staticPitch + pDLSRegion->wtRegion.tuning + + (((EAS_I32) pVoice->note * (EAS_I32) pDLSArt->keyNumToPitch) >> 7); + + /* don't transpose rhythm channel */ + if ((pChannel ->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) == 0) + temp += pSynth->globalTranspose * 100; + + /* calculate phase increment including modulation effects */ + intFrame.frame.phaseIncrement = DLS_UpdatePhaseInc(pWTVoice, pDLSArt, pChannel, temp); + + /* calculate gain including modulation effects */ + intFrame.frame.gainTarget = DLS_UpdateGain(pWTVoice, pDLSArt, pChannel, pDLSRegion->wtRegion.gain, pVoice->velocity); + intFrame.prevGain = pVoice->gain; + + DLS_UpdateFilter(pVoice, pWTVoice, &intFrame, pChannel, pDLSArt); + + /* call into engine to generate samples */ + intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer; + intFrame.pMixBuffer = pMixBuffer; + intFrame.numSamples = numSamples; + if (numSamples < 0) + return EAS_FALSE; + + /* check for end of sample */ + if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd)) + done = WT_CheckSampleEnd(pWTVoice, &intFrame, EAS_FALSE); + + WT_ProcessVoice(pWTVoice, &intFrame); + + /* clear flag */ + pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; + + /* if the update interval has elapsed, then force the current gain to the next + * gain since we never actually reach the next gain when ramping -- we just get + * very close to the target gain. + */ + pVoice->gain = (EAS_I16) intFrame.frame.gainTarget; + + /* if voice has finished, set flag for voice manager */ + if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted)) + done = EAS_TRUE; + + return done; +} + +/*---------------------------------------------------------------------------- + * DLS_UpdateEnvelope() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesize a block of samples for the given voice. + * Use linear interpolation. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * number of samples actually written to buffer + * + * Side Effects: + * - samples are added to the presently free buffer + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pChannel) pChannel not used in this instance */ +static void DLS_UpdateEnvelope (S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, const S_DLS_ENVELOPE *pEnvParams, EAS_I16 *pValue, EAS_I16 *pIncrement, EAS_U8 *pState) +{ + EAS_I32 temp; + + switch (*pState) + { + /* initial state */ + case eEnvelopeStateInit: + *pState = eEnvelopeStateDelay; + *pValue = 0; + *pIncrement = pEnvParams->delayTime; + if (*pIncrement != 0) + return; + /*lint -e{825} falls through to next case */ + + case eEnvelopeStateDelay: + if (*pIncrement) + { + *pIncrement = *pIncrement - 1; + return; + } + + /* calculate attack rate */ + *pState = eEnvelopeStateAttack; + if (pEnvParams->attackTime != ZERO_TIME_IN_CENTS) + { + /*lint -e{702} use shift for performance */ + temp = pEnvParams->attackTime + ((pEnvParams->velToAttack * pVoice->velocity) >> 7); + *pIncrement = ConvertRate(temp); + return; + } + + *pValue = SYNTH_FULL_SCALE_EG1_GAIN; + /*lint -e{825} falls through to next case */ + + case eEnvelopeStateAttack: + if (*pValue < SYNTH_FULL_SCALE_EG1_GAIN) + { + temp = *pValue + *pIncrement; + *pValue = (EAS_I16) (temp < SYNTH_FULL_SCALE_EG1_GAIN ? temp : SYNTH_FULL_SCALE_EG1_GAIN); + return; + } + + /* calculate hold time */ + *pState = eEnvelopeStateHold; + if (pEnvParams->holdTime != ZERO_TIME_IN_CENTS) + { + /*lint -e{702} use shift for performance */ + temp = pEnvParams->holdTime + ((pEnvParams->keyNumToHold * pVoice->note) >> 7); + *pIncrement = ConvertDelay(temp); + return; + } + else + *pIncrement = 0; + /*lint -e{825} falls through to next case */ + + case eEnvelopeStateHold: + if (*pIncrement) + { + *pIncrement = *pIncrement - 1; + return; + } + + /* calculate decay rate */ + *pState = eEnvelopeStateDecay; + if (pEnvParams->decayTime != ZERO_TIME_IN_CENTS) + { + /*lint -e{702} use shift for performance */ + temp = pEnvParams->decayTime + ((pEnvParams->keyNumToDecay * pVoice->note) >> 7); + *pIncrement = ConvertRate(temp); + return; + } + +// *pValue = pEnvParams->sustainLevel; + /*lint -e{825} falls through to next case */ + + case eEnvelopeStateDecay: + if (*pValue > pEnvParams->sustainLevel) + { + temp = *pValue - *pIncrement; + *pValue = (EAS_I16) (temp > pEnvParams->sustainLevel ? temp : pEnvParams->sustainLevel); + return; + } + + *pState = eEnvelopeStateSustain; + *pValue = pEnvParams->sustainLevel; + /*lint -e{825} falls through to next case */ + + case eEnvelopeStateSustain: + return; + + case eEnvelopeStateRelease: + temp = *pValue - *pIncrement; + if (temp <= 0) + { + *pState = eEnvelopeStateMuted; + *pValue = 0; + } + else + *pValue = (EAS_I16) temp; + break; + + case eEnvelopeStateMuted: + *pValue = 0; + return; + + default: + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Envelope in invalid state %d\n", *pState); */ } + break; + } +} + diff --git a/arm-wt-22k/lib_src/eas_dlssynth.h b/arm-wt-22k/lib_src/eas_dlssynth.h new file mode 100644 index 0000000..0dab41e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_dlssynth.h @@ -0,0 +1,41 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_dlssynth.h
+ *
+ * Contents and purpose:
+ * Implements the Mobile DLS synthesizer.
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 143 $
+ * $Date: 2006-07-17 14:09:35 -0700 (Mon, 17 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_DLSSYNTH_H
+#define _EAS_DLSSYNTH_H
+
+/* prototypes */
+void DLS_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+void DLS_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+void DLS_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
+EAS_RESULT DLS_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
+EAS_BOOL DLS_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples);
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_effects.h b/arm-wt-22k/lib_src/eas_effects.h new file mode 100644 index 0000000..01e64c0 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_effects.h @@ -0,0 +1,61 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_effects.h
+ *
+ * Contents and purpose:
+ * Defines a generic effects interface.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_EFFECTS_H
+#define _EAS_EFFECTS_H
+
+#include "eas_types.h"
+
+typedef struct
+{
+ EAS_RESULT (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+ void (*pfProcess)(EAS_VOID_PTR pInstData, EAS_PCM *in, EAS_PCM *out, EAS_I32 numSamples);
+ EAS_RESULT (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+ EAS_RESULT (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+} S_EFFECTS_INTERFACE;
+
+typedef struct
+{
+ EAS_RESULT (*pfInit)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+ void (*pfProcess)(EAS_VOID_PTR pInstData, EAS_I32 *in, EAS_I32 *out, EAS_I32 numSamples);
+ EAS_RESULT (*pfShutdown)(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (*pFGetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+ EAS_RESULT (*pFSetParam)(EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+} S_EFFECTS32_INTERFACE;
+
+/* mixer instance data */
+typedef struct
+{
+ S_EFFECTS_INTERFACE *effect;
+ EAS_VOID_PTR effectData;
+} S_EFFECTS_MODULE;
+
+#endif /* end _EAS_EFFECTS_H */
+
diff --git a/arm-wt-22k/lib_src/eas_flog.c b/arm-wt-22k/lib_src/eas_flog.c new file mode 100644 index 0000000..647da2f --- /dev/null +++ b/arm-wt-22k/lib_src/eas_flog.c @@ -0,0 +1,96 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_flog2.c
+ *
+ * Contents and purpose:
+ * Fixed point square root
+ *
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision$
+ * $Date$
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_math.h"
+
+#define MANTISSA_SHIFT 27
+#define MANTISSA_MASK 0x0000000f
+#define MANTISSA_LSB_SHIFT 7
+#define MANTISSA_LSB_MASK 0x000fffff
+#define LOG_EXPONENT_SHIFT 10
+#define INTERPOLATION_SHIFT 20
+#define MAX_NEGATIVE (-2147483647-1)
+
+/* log lookup table */
+static const EAS_U16 eas_log2_table[] =
+{
+ 0, 90, 174, 254, 330, 402, 470, 536,
+ 599, 659, 717, 773, 827, 879, 929, 977,
+ 1024
+};
+
+/*----------------------------------------------------------------------------
+ * EAS_flog2()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the log2 of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the log2 of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_flog2 (EAS_U32 n)
+{
+ EAS_U32 exp;
+ EAS_U32 interp;
+
+ /* check for error condition */
+ if (n == 0)
+ return MAX_NEGATIVE;
+
+ /* find exponent */
+ for (exp = 31; exp > 0; exp--)
+ {
+ /* shift until we get a 1 bit in bit 31 */
+ if ((n & (EAS_U32) MAX_NEGATIVE) != 0)
+ break;
+ n <<= 1;
+ }
+ /*lint -e{701} use shift for performance */
+ exp <<= LOG_EXPONENT_SHIFT;
+
+ /* get the least significant bits for interpolation */
+ interp = (n >> MANTISSA_LSB_SHIFT) & MANTISSA_LSB_MASK;
+
+ /* get the most significant bits for mantissa lookup */
+ n = (n >> MANTISSA_SHIFT) & MANTISSA_MASK;
+
+ /* interpolate mantissa */
+ interp = ((eas_log2_table[n+1] - eas_log2_table[n]) * interp) >> INTERPOLATION_SHIFT;
+ exp += eas_log2_table[n] + interp;
+
+ return (EAS_I32) exp;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_ima_tables.c b/arm-wt-22k/lib_src/eas_ima_tables.c new file mode 100644 index 0000000..56bd1eb --- /dev/null +++ b/arm-wt-22k/lib_src/eas_ima_tables.c @@ -0,0 +1,54 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_ima_tables.c
+ *
+ * Contents and purpose:
+ * Contains the constant tables for IMA encode/decode
+ *
+ * Copyright (c) 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 760 $
+ * $Date: 2007-07-17 23:09:36 -0700 (Tue, 17 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+
+/*----------------------------------------------------------------------------
+ * ADPCM decode tables
+ *----------------------------------------------------------------------------
+*/
+const EAS_I16 imaIndexTable[16] =
+{
+ -1, -1, -1, -1, 2, 4, 6, 8,
+ -1, -1, -1, -1, 2, 4, 6, 8
+};
+
+const EAS_I16 imaStepSizeTable[89] =
+{
+ 7, 8, 9, 10, 11, 12, 13, 14, 16, 17,
+ 19, 21, 23, 25, 28, 31, 34, 37, 41, 45,
+ 50, 55, 60, 66, 73, 80, 88, 97, 107, 118,
+ 130, 143, 157, 173, 190, 209, 230, 253, 279, 307,
+ 337, 371, 408, 449, 494, 544, 598, 658, 724, 796,
+ 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066,
+ 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358,
+ 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899,
+ 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+};
+
diff --git a/arm-wt-22k/lib_src/eas_imaadpcm.c b/arm-wt-22k/lib_src/eas_imaadpcm.c new file mode 100644 index 0000000..68bf257 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_imaadpcm.c @@ -0,0 +1,368 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imaadpcm.c
+ *
+ * Contents and purpose:
+ * Implements the IMA ADPCM decoder
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 847 $
+ * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_host.h"
+#include "eas_pcm.h"
+#include "eas_math.h"
+#include "eas_report.h"
+
+// #define _DEBUG_IMA_ADPCM_LOCATE
+
+/*----------------------------------------------------------------------------
+ * externs
+ *----------------------------------------------------------------------------
+*/
+extern const EAS_I16 imaIndexTable[];
+extern const EAS_I16 imaStepSizeTable[];
+
+/*----------------------------------------------------------------------------
+ * prototypes
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble);
+static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time);
+
+/*----------------------------------------------------------------------------
+ * IMA ADPCM Decoder interface
+ *----------------------------------------------------------------------------
+*/
+const S_DECODER_INTERFACE IMADecoder =
+{
+ IMADecoderInit,
+ IMADecoderSample,
+ IMADecoderLocate
+};
+
+/*----------------------------------------------------------------------------
+ * IMADecoderInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the IMA ADPCM decoder
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMADecoderInit (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState)
+{
+ pState->decoderL.step = 0;
+ pState->decoderR.step = 0;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderSample()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes an IMA ADPCM sample
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderSample (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState)
+{
+ EAS_RESULT result;
+ EAS_I16 sTemp;
+
+ /* if high nibble, decode */
+ if (pState->hiNibble)
+ {
+ IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte >> 4));
+ pState->hiNibble = EAS_FALSE;
+ }
+
+ /* low nibble, need to fetch another byte */
+ else
+ {
+ /* check for loop */
+ if ((pState->bytesLeft == 0) && (pState->loopSamples != 0))
+ {
+ /* seek to start of loop */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, (EAS_I32) (pState->startPos + pState->loopLocation))) != EAS_SUCCESS)
+ return result;
+ pState->bytesLeft = pState->byteCount = (EAS_I32) pState->bytesLeftLoop;
+ pState->blockCount = 0;
+ pState->flags &= ~PCM_FLAGS_EMPTY;
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMADecoderSample: Rewind file to %d, bytesLeft = %d\n", pState->startPos, pState->bytesLeft); */ }
+ }
+
+ /* if start of block, fetch new predictor and step index */
+ if ((pState->blockSize != 0) && (pState->blockCount == 0) && (pState->bytesLeft != 0))
+ {
+
+ /* get predicted sample for left channel */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+#ifdef _DEBUG_IMA_ADPCM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Predictor: Was %d, now %d\n", pState->decoderL.acc, sTemp); */ }
+#endif
+ pState->decoderL.acc = pState->decoderL.x1 = sTemp;
+
+ /* get step index for left channel - upper 8 bits are reserved */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+#ifdef _DEBUG_IMA_ADPCM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderL.step, sTemp); */ }
+#endif
+ pState->decoderL.step = sTemp & 0xff;
+
+ if (pState->flags & PCM_FLAGS_STEREO)
+ {
+ /* get predicted sample for right channel */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+ pState->decoderR.acc = pState->decoderR.x1 = sTemp;
+
+ /* get step index for right channel - upper 8 bits are reserved */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, pState->fileHandle, &sTemp, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+#ifdef _DEBUG_IMA_ADPCM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Step: Was %d, now %d\n", pState->decoderR.step, sTemp); */ }
+#endif
+ pState->decoderR.step = sTemp & 0xff;
+
+ pState->blockCount = pState->blockSize - 8;
+ pState->bytesLeft -= 8;
+ }
+ else
+ {
+ pState->blockCount = pState->blockSize - 4;
+ pState->bytesLeft -= 4;
+ }
+ }
+ else
+ {
+
+ /* get another ADPCM data pair */
+ if (pState->bytesLeft)
+ {
+
+ if ((result = EAS_HWGetByte(pEASData->hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS)
+ return result;
+
+ /* decode the low nibble */
+ pState->bytesLeft--;
+ pState->blockCount--;
+ IMADecoderADPCM(&pState->decoderL, (EAS_U8)(pState->srcByte & 0x0f));
+
+ if (pState->flags & PCM_FLAGS_STEREO)
+ IMADecoderADPCM(&pState->decoderR, (EAS_U8)(pState->srcByte >> 4));
+ else
+ pState->hiNibble = EAS_TRUE;
+ }
+
+ /* out of ADPCM data, generate enough samples to fill buffer */
+ else
+ {
+ pState->decoderL.x1 = pState->decoderL.x0;
+ pState->decoderR.x1 = pState->decoderR.x0;
+ }
+ }
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderADPCM()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes an IMA ADPCM sample
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void IMADecoderADPCM (S_DECODER_STATE *pState, EAS_U8 nibble)
+{
+ EAS_INT delta;
+ EAS_INT stepSize;
+
+ /* get stepsize from table */
+ stepSize = imaStepSizeTable[pState->step];
+
+ /* delta = (abs(delta) + 0.5) * step / 4 */
+ delta = 0;
+ if (nibble & 4)
+ delta += stepSize;
+
+ if (nibble & 2)
+ /*lint -e{702} use shift for performance */
+ delta += stepSize >> 1;
+
+ if (nibble & 1)
+ /*lint -e{702} use shift for performance */
+ delta += stepSize >> 2;
+
+ /*lint -e{702} use shift for performance */
+ delta += stepSize >> 3;
+
+ /* integrate the delta */
+ if (nibble & 8)
+ pState->acc -= delta;
+ else
+ pState->acc += delta;
+
+ /* saturate */
+ if (pState->acc > 32767)
+ pState->acc = 32767;
+ if (pState->acc < -32768)
+ pState->acc = -32768;
+ pState->x1 = (EAS_PCM) pState->acc;
+
+ /* compute new step size */
+ pState->step += imaIndexTable[nibble];
+ if (pState->step < 0)
+ pState->step = 0;
+ if (pState->step > 88)
+ pState->step = 88;
+
+#ifdef _DEBUG_IMA_ADPCM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "In=%u, Pred=%d, Step=%d\n", nibble, pState->acc, imaStepSizeTable[pState->step]); */ }
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * IMADecoderLocate()
+ *----------------------------------------------------------------------------
+ * Locate in an IMA ADPCM stream
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMADecoderLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time)
+{
+ EAS_RESULT result;
+ EAS_I32 temp;
+ EAS_I32 samplesPerBlock;
+ EAS_I32 secs, msecs;
+
+ /* no need to calculate if time is zero */
+ if (time == 0)
+ temp = 0;
+
+ /* not zero */
+ else
+ {
+
+ /* can't seek if not a blocked file */
+ if (pState->blockSize == 0)
+ return EAS_ERROR_FEATURE_NOT_AVAILABLE;
+
+ /* calculate number of samples per block */
+ if (pState->flags & PCM_FLAGS_STEREO)
+ samplesPerBlock = pState->blockSize - 7;
+ else
+ samplesPerBlock = (pState->blockSize << 1) - 7;
+
+ /* break down into secs and msecs */
+ secs = time / 1000;
+ msecs = time - (secs * 1000);
+
+ /* calculate sample number fraction from msecs */
+ temp = (msecs * pState->sampleRate);
+ temp = (temp >> 10) + ((temp * 49) >> 21);
+
+ /* add integer sample count */
+ temp += secs * pState->sampleRate;
+
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+ EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000006 , time, temp);
+#endif
+
+ /* for looped samples, calculate position in the loop */
+ if ((temp > pState->byteCount) && (pState->loopSamples != 0))
+ {
+ EAS_I32 numBlocks;
+ EAS_I32 samplesPerLoop;
+ EAS_I32 samplesInLastBlock;
+
+ numBlocks = (EAS_I32) (pState->loopStart / pState->blockSize);
+ samplesInLastBlock = (EAS_I32) pState->loopStart - (numBlocks * pState->blockSize);
+ if (samplesInLastBlock)
+ {
+ if (pState->flags & PCM_FLAGS_STEREO)
+ samplesInLastBlock = samplesInLastBlock - 7;
+ else
+ /*lint -e{703} use shift for performance */
+ samplesInLastBlock = (samplesInLastBlock << 1) - 7;
+ }
+ samplesPerLoop = numBlocks * samplesPerBlock + samplesInLastBlock;
+ temp = temp % samplesPerLoop;
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+ EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000007 , numBlocks, samplesPerLoop, samplesInLastBlock, temp);
+#endif
+ }
+
+ /* find start of block for requested sample */
+ temp = (temp / samplesPerBlock) * pState->blockSize;
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+ EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000008 , temp);
+#endif
+
+ }
+
+ /* seek to new location */
+ if ((result = EAS_PESeek(pEASData, pState, &temp)) != EAS_SUCCESS)
+ return result;
+
+#ifdef _DEBUG_IMA_ADPCM_LOCATE
+ EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x2380b977, 0x00000009 , pState->bytesLeft);
+#endif
+
+ /* reset state */
+ pState->blockCount = 0;
+ pState->hiNibble = EAS_FALSE;
+ if ((pState->state != EAS_STATE_PAUSING) && (pState->state != EAS_STATE_PAUSED))
+ pState->state = EAS_STATE_READY;
+
+ return EAS_SUCCESS;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_imelody.c b/arm-wt-22k/lib_src/eas_imelody.c new file mode 100644 index 0000000..5b7e7b8 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_imelody.c @@ -0,0 +1,1742 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imelody.c
+ *
+ * Contents and purpose:
+ * iMelody parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 797 $
+ * $Date: 2007-08-01 00:15:56 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* lint doesn't like the way some string.h files look */
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <string.h>
+#endif
+
+#include "eas_data.h"
+#include "eas_miditypes.h"
+#include "eas_parser.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_midi.h"
+#include "eas_config.h"
+#include "eas_vm_protos.h"
+#include "eas_imelodydata.h"
+#include "eas_ctype.h"
+
+// #define _DEBUG_IMELODY
+
+/* increase gain for mono ringtones */
+#define IMELODY_GAIN_OFFSET 8
+
+/* length of 32nd note in 1/256ths of a msec for 120 BPM tempo */
+#define DEFAULT_TICK_CONV 16000
+#define TICK_CONVERT 1920000
+
+/* default channel and program for iMelody playback */
+#define IMELODY_CHANNEL 0
+#define IMELODY_PROGRAM 80
+#define IMELODY_VEL_MUL 4
+#define IMELODY_VEL_OFS 67
+
+/* multiplier for fixed point triplet conversion */
+#define TRIPLET_MULTIPLIER 683
+#define TRIPLET_SHIFT 10
+
+static const char* const tokens[] =
+{
+ "BEGIN:IMELODY",
+ "VERSION:",
+ "FORMAT:CLASS",
+ "NAME:",
+ "COMPOSER:",
+ "BEAT:",
+ "STYLE:",
+ "VOLUME:",
+ "MELODY:",
+ "END:IMELODY"
+};
+
+/* ledon or ledoff */
+static const char ledStr[] = "edo";
+
+/* vibeon or vibeoff */
+static const char vibeStr[] = "ibeo";
+
+/* backon or backoff */
+static const char backStr[] = "cko";
+
+typedef enum
+{
+ TOKEN_BEGIN,
+ TOKEN_VERSION,
+ TOKEN_FORMAT,
+ TOKEN_NAME,
+ TOKEN_COMPOSER,
+ TOKEN_BEAT,
+ TOKEN_STYLE,
+ TOKEN_VOLUME,
+ TOKEN_MELODY,
+ TOKEN_END,
+ TOKEN_INVALID
+} ENUM_IMELODY_TOKENS;
+
+/* lookup table for note values */
+static const EAS_I8 noteTable[] = { 9, 11, 0, 2, 4, 5, 7 };
+
+/* inline functions */
+#ifdef _DEBUG_IMELODY
+static void PutBackChar (S_IMELODY_DATA *pData)
+{
+ if (pData->index)
+ pData->index--;
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "PutBackChar '%c'\n", pData->buffer[pData->index]); */ }
+}
+#else
+EAS_INLINE void PutBackChar (S_IMELODY_DATA *pData) { if (pData->index) pData->index--; }
+#endif
+
+
+/* local prototypes */
+static EAS_RESULT IMY_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+static EAS_RESULT IMY_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT IMY_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+static EAS_RESULT IMY_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+static EAS_RESULT IMY_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT IMY_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT IMY_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT IMY_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT IMY_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT IMY_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT IMY_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_BOOL IMY_PlayNote (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData, EAS_I8 note, EAS_INT parserMode);
+static EAS_BOOL IMY_PlayRest (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData);
+static EAS_BOOL IMY_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_I32 *pDuration);
+static EAS_BOOL IMY_GetLEDState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData);
+static EAS_BOOL IMY_GetVibeState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData);
+static EAS_BOOL IMY_GetBackState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData);
+static EAS_BOOL IMY_GetVolume (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader);
+static EAS_BOOL IMY_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_INT *temp, EAS_BOOL inHeader);
+static EAS_RESULT IMY_ParseHeader (S_EAS_DATA *pEASData, S_IMELODY_DATA* pData);
+static EAS_I8 IMY_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader);
+static EAS_RESULT IMY_ReadLine (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I8 *buffer, EAS_I32 *pStartLine);
+static EAS_INT IMY_ParseLine (EAS_I8 *buffer, EAS_U8 *pIndex);
+
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_iMelody_Parser
+ *
+ * This structure contains the functional interface for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_iMelody_Parser =
+{
+ IMY_CheckFileType,
+ IMY_Prepare,
+ IMY_Time,
+ IMY_Event,
+ IMY_State,
+ IMY_Close,
+ IMY_Reset,
+ IMY_Pause,
+ IMY_Resume,
+ NULL,
+ IMY_SetData,
+ IMY_GetData,
+ NULL
+};
+
+/*----------------------------------------------------------------------------
+ * IMY_CheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset)
+{
+ S_IMELODY_DATA* pData;
+ EAS_I8 buffer[MAX_LINE_SIZE+1];
+ EAS_U8 index;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_CheckFileType\n"); */ }
+#endif
+
+ /* read the first line of the file */
+ *ppHandle = NULL;
+ if (IMY_ReadLine(pEASData->hwInstData, fileHandle, buffer, NULL) != EAS_SUCCESS)
+ return EAS_SUCCESS;
+
+ /* check for header string */
+ if (IMY_ParseLine(buffer, &index) == TOKEN_BEGIN)
+ {
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pData = EAS_CMEnumData(EAS_CM_IMELODY_DATA);
+ else
+ pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_IMELODY_DATA));
+ if (!pData)
+ return EAS_ERROR_MALLOC_FAILED;
+ EAS_HWMemSet(pData, 0, sizeof(S_IMELODY_DATA));
+
+ /* initialize */
+ pData->fileHandle = fileHandle;
+ pData->fileOffset = offset;
+ pData->state = EAS_STATE_ERROR;
+ pData->state = EAS_STATE_OPEN;
+
+ /* return a pointer to the instance data */
+ *ppHandle = pData;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Prepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_IMELODY_DATA* pData;
+ EAS_RESULT result;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_Prepare\n"); */ }
+#endif
+
+ /* check for valid state */
+ pData = (S_IMELODY_DATA*) pInstData;
+ if (pData->state != EAS_STATE_OPEN)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* instantiate a synthesizer */
+ if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ }
+ return result;
+ }
+
+ /* parse the header */
+ if ((result = IMY_ParseHeader(pEASData, pData)) != EAS_SUCCESS)
+ return result;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Prepare: state set to EAS_STATE_READY\n"); */ }
+#endif
+
+ pData ->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Time()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the time of the next event in msecs
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pTime - pointer to variable to hold time of next event (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMY_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime)
+{
+ S_IMELODY_DATA *pData;
+
+ pData = (S_IMELODY_DATA*) pInstData;
+
+ /* return time in milliseconds */
+ /*lint -e{704} use shift instead of division */
+ *pTime = pData->time >> 8;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Event()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the next event in the file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode)
+{
+ S_IMELODY_DATA* pData;
+ EAS_RESULT result;
+ EAS_I8 c;
+ EAS_BOOL eof;
+ EAS_INT temp;
+
+ pData = (S_IMELODY_DATA*) pInstData;
+ if (pData->state >= EAS_STATE_OPEN)
+ return EAS_SUCCESS;
+
+ if (pData->state == EAS_STATE_READY) {
+ pData->state = EAS_STATE_PLAY;
+ }
+
+ /* initialize MIDI channel when the track starts playing */
+ if (pData->time == 0)
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: Reset\n"); */ }
+#endif
+ /* set program to square lead */
+ VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, IMELODY_PROGRAM);
+
+ /* set channel volume to max */
+ VMControlChange(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, 7, 127);
+ }
+
+ /* check for end of note */
+ if (pData->note)
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Stopping note %d\n", pData->note); */ }
+#endif
+ /* stop the note */
+ VMStopNote(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, pData->note, 0);
+ pData->note = 0;
+
+ /* check for rest between notes */
+ if (pData->restTicks)
+ {
+ pData->time += pData->restTicks;
+ pData->restTicks = 0;
+ return EAS_SUCCESS;
+ }
+ }
+
+ /* parse the next event */
+ eof = EAS_FALSE;
+ while (!eof)
+ {
+
+ /* get next character */
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE);
+
+ switch (c)
+ {
+ /* start repeat */
+ case '(':
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter repeat section\n", c); */ }
+#endif
+
+ if (pData->repeatOffset < 0)
+ {
+ pData->repeatOffset = pData->startLine + (EAS_I32) pData->index;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat offset = %d\n", pData->repeatOffset); */ }
+#endif
+ }
+ else
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring nested repeat section\n"); */ }
+ break;
+
+ /* end repeat */
+ case ')':
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "End repeat section, repeat offset = %d\n", pData->repeatOffset); */ }
+#endif
+ /* ignore invalid repeats */
+ if (pData->repeatCount >= 0)
+ {
+
+ /* decrement repeat count (repeatCount == 0 means infinite loop) */
+ if (pData->repeatCount > 0)
+ {
+ if (--pData->repeatCount == 0)
+ {
+ pData->repeatCount = -1;
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat loop complete\n"); */ }
+#endif
+ }
+ }
+
+//2 TEMPORARY FIX: If locating, don't do infinite loops.
+//3 We need a different mode for metadata parsing where we don't loop at all
+ if ((parserMode == eParserModePlay) || (pData->repeatCount != 0))
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Rewinding file for repeat\n"); */ }
+#endif
+ /* rewind to start of loop */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->repeatOffset)) != EAS_SUCCESS)
+ return result;
+ IMY_ReadLine(pEASData->hwInstData, pData->fileHandle, pData->buffer, &pData->startLine);
+ pData->index = 0;
+
+ /* if last loop, prevent future loops */
+ if (pData->repeatCount == -1)
+ pData->repeatOffset = -1;
+ }
+ }
+ break;
+
+ /* repeat count */
+ case '@':
+ if (!IMY_GetNumber(pEASData->hwInstData, pData, &temp, EAS_FALSE))
+ eof = EAS_TRUE;
+ else if (pData->repeatOffset > 0)
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Repeat count = %dt", pData->repeatCount); */ }
+#endif
+ if (pData->repeatCount < 0)
+ pData->repeatCount = (EAS_I16) temp;
+ }
+ break;
+
+ /* volume */
+ case 'V':
+ if (!IMY_GetVolume(pEASData->hwInstData, pData, EAS_FALSE))
+ eof = EAS_TRUE;
+ break;
+
+ /* flat */
+ case '&':
+ pData->noteModifier = -1;
+ break;
+
+ /* sharp */
+ case '#':
+ pData->noteModifier = +1;
+ break;
+
+ /* octave */
+ case '*':
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE);
+ if (IsDigit(c))
+ pData->octave = (EAS_U8) ((c - '0' + 1) * 12);
+ else if (!c)
+ eof = EAS_TRUE;
+ break;
+
+ /* ledon or ledoff */
+ case 'l':
+ if (!IMY_GetLEDState(pEASData, pData))
+ eof = EAS_TRUE;
+ break;
+
+ /* vibeon or vibeoff */
+ case 'v':
+ if (!IMY_GetVibeState(pEASData, pData))
+ eof = EAS_TRUE;
+ break;
+
+ /* either a B note or backon or backoff */
+ case 'b':
+ if (IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE) == 'a')
+ {
+ if (!IMY_GetBackState(pEASData, pData))
+ eof = EAS_TRUE;
+ }
+ else
+ {
+ PutBackChar(pData);
+ if (IMY_PlayNote(pEASData, pData, c, parserMode))
+ return EAS_SUCCESS;
+ eof = EAS_TRUE;
+ }
+ break;
+
+ /* rest */
+ case 'r':
+ case 'R':
+ if (IMY_PlayRest(pEASData, pData))
+ return EAS_SUCCESS;
+ eof = EAS_TRUE;
+ break;
+
+ /* EOF */
+ case 0:
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: end of iMelody file detected\n"); */ }
+#endif
+ eof = EAS_TRUE;
+ break;
+
+ /* must be a note */
+ default:
+ c = ToLower(c);
+ if ((c >= 'a') && (c <= 'g'))
+ {
+ if (IMY_PlayNote(pEASData, pData, c, parserMode))
+ return EAS_SUCCESS;
+ eof = EAS_TRUE;
+ }
+ else
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unexpected character '%c' [0x%02x]\n", c, c); */ }
+ break;
+ }
+ }
+
+ /* handle EOF */
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Event: state set to EAS_STATE_STOPPING\n"); */ }
+#endif
+ pData->state = EAS_STATE_STOPPING;
+ VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_State()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMY_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState)
+{
+ S_IMELODY_DATA* pData;
+
+ /* establish pointer to instance data */
+ pData = (S_IMELODY_DATA*) pInstData;
+
+ /* if stopping, check to see if synth voices are active */
+ if (pData->state == EAS_STATE_STOPPING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ {
+ pData->state = EAS_STATE_STOPPED;
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_State: state set to EAS_STATE_STOPPED\n"); */ }
+#endif
+ }
+ }
+
+ if (pData->state == EAS_STATE_PAUSING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_State: state set to EAS_STATE_PAUSED\n"); */ }
+#endif
+ pData->state = EAS_STATE_PAUSED;
+ }
+ }
+
+ /* return current state */
+ *pState = pData->state;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Close()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_IMELODY_DATA* pData;
+ EAS_RESULT result;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Close: close file\n"); */ }
+#endif
+
+ pData = (S_IMELODY_DATA*) pInstData;
+
+ /* close the file */
+ if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS)
+ return result;
+
+ /* free the synth */
+ if (pData->pSynth != NULL)
+ VMMIDIShutdown(pEASData, pData->pSynth);
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pData);
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_IMELODY_DATA* pData;
+ EAS_RESULT result;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Reset: reset file\n"); */ }
+#endif
+ pData = (S_IMELODY_DATA*) pInstData;
+
+ /* reset the synth */
+ VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE);
+
+ /* reset time to zero */
+ pData->time = 0;
+ pData->note = 0;
+
+ /* reset file position and re-parse header */
+ pData->state = EAS_STATE_ERROR;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+ if ((result = IMY_ParseHeader (pEASData, pData)) != EAS_SUCCESS)
+ return result;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Reset: state set to EAS_STATE_ERROR\n"); */ }
+#endif
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Pause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Pauses the sequencer. Mutes all voices and sets state to pause.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_IMELODY_DATA *pData;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Pause: pause file\n"); */ }
+#endif
+
+ /* can't pause a stopped stream */
+ pData = (S_IMELODY_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* mute the synthesizer */
+ VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ pData->state = EAS_STATE_PAUSING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_Resume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume playing after a pause, sets state back to playing.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMY_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_IMELODY_DATA *pData;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_Resume: resume file\n"); */ }
+#endif
+
+ /* can't resume a stopped stream */
+ pData = (S_IMELODY_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* nothing to do but resume playback */
+ pData->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_SetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Adjust tempo relative to song tempo
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pInstData - pointer to iMelody instance data
+ * rate - rate (28-bit fractional amount)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMY_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_IMELODY_DATA *pData;
+
+ pData = (S_IMELODY_DATA*) pInstData;
+ switch (param)
+ {
+
+ /* set metadata callback */
+ case PARSER_DATA_METADATA_CB:
+ EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB));
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return the file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pInstData - pointer to iMelody instance data
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT IMY_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_IMELODY_DATA *pData;
+
+ pData = (S_IMELODY_DATA*) pInstData;
+
+ switch (param)
+ {
+ /* return file type as iMelody */
+ case PARSER_DATA_FILE_TYPE:
+ *pValue = EAS_FILE_IMELODY;
+ break;
+
+ case PARSER_DATA_SYNTH_HANDLE:
+ *pValue = (EAS_I32) pData->pSynth;
+ break;
+
+ case PARSER_DATA_GAIN_OFFSET:
+ *pValue = IMELODY_GAIN_OFFSET;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_PlayNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_PlayNote (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData, EAS_I8 note, EAS_INT parserMode)
+{
+ EAS_I32 duration;
+ EAS_U8 velocity;
+
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayNote: start note %d\n", note); */ }
+#endif
+
+ /* get the duration */
+ if (!IMY_GetDuration(pEASData->hwInstData, pData, &duration))
+ return EAS_FALSE;
+
+ /* save note value */
+ pData->note = (EAS_U8) (pData->octave + noteTable[note - 'a'] + pData->noteModifier);
+ velocity = (EAS_U8) (pData->volume ? pData->volume * IMELODY_VEL_MUL + IMELODY_VEL_OFS : 0);
+
+ /* start note only if in play mode */
+ if (parserMode == eParserModePlay)
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, IMELODY_CHANNEL, pData->note, velocity);
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayNote: Start note %d, duration %d\n", pData->note, duration); */ }
+#endif
+
+ /* determine note length */
+ switch (pData->style)
+ {
+ case 0:
+ /*lint -e{704} shift for performance */
+ pData->restTicks = duration >> 4;
+ break;
+ case 1:
+ pData->restTicks = 0;
+ break;
+ case 2:
+ /*lint -e{704} shift for performance */
+ pData->restTicks = duration >> 1;
+ break;
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "IMY_PlayNote: Note style out of range: %d\n", pData->style); */ }
+ /*lint -e{704} shift for performance */
+ pData->restTicks = duration >> 4;
+ break;
+ }
+
+ /* next event is at end of this note */
+ pData->time += duration - pData->restTicks;
+
+ /* reset the flat/sharp modifier */
+ pData->noteModifier = 0;
+
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_PlayRest()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_PlayRest (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData)
+{
+ EAS_I32 duration;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_PlayRest]n"); */ }
+#endif
+
+ /* get the duration */
+ if (!IMY_GetDuration(pEASData->hwInstData, pData, &duration))
+ return EAS_FALSE;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_PlayRest: note duration %d\n", duration); */ }
+#endif
+
+ /* next event is at end of this note */
+ pData->time += duration;
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetDuration()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+
+static EAS_BOOL IMY_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_I32 *pDuration)
+{
+ EAS_I32 duration;
+ EAS_I8 c;
+
+ /* get the duration */
+ *pDuration = 0;
+ c = IMY_GetNextChar(hwInstData, pData, EAS_FALSE);
+ if (!c)
+ return EAS_FALSE;
+ if ((c < '0') || (c > '5'))
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetDuration: error in duration '%c'\n", c); */ }
+#endif
+ return EAS_FALSE;
+ }
+
+ /* calculate total length of note */
+ duration = pData->tick * (1 << ('5' - c));
+
+ /* check for duration modifier */
+ c = IMY_GetNextChar(hwInstData, pData, EAS_FALSE);
+ if (c)
+ {
+ if (c == '.')
+ /*lint -e{704} shift for performance */
+ duration += duration >> 1;
+ else if (c == ':')
+ /*lint -e{704} shift for performance */
+ duration += (duration >> 1) + (duration >> 2);
+ else if (c == ';')
+ /*lint -e{704} shift for performance */
+ duration = (duration * TRIPLET_MULTIPLIER) >> TRIPLET_SHIFT;
+ else
+ PutBackChar(pData);
+ }
+
+ *pDuration = duration;
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetLEDState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetLEDState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData)
+{
+ EAS_I8 c;
+ EAS_INT i;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetLEDState\n"); */ }
+#endif
+
+ for (i = 0; i < 5; i++)
+ {
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE);
+ if (!c)
+ return EAS_FALSE;
+ switch (i)
+ {
+ case 3:
+ if (c == 'n')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetLEDState: LED on\n"); */ }
+#endif
+ EAS_HWLED(pEASData->hwInstData, EAS_TRUE);
+ return EAS_TRUE;
+ }
+ else if (c != 'f')
+ return EAS_FALSE;
+ break;
+
+ case 4:
+ if (c == 'f')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetLEDState: LED off\n"); */ }
+#endif
+ EAS_HWLED(pEASData->hwInstData, EAS_FALSE);
+ return EAS_TRUE;
+ }
+ return EAS_FALSE;
+
+ default:
+ if (c != ledStr[i])
+ return EAS_FALSE;
+ break;
+ }
+ }
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetVibeState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetVibeState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData)
+{
+ EAS_I8 c;
+ EAS_INT i;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetVibeState\n"); */ }
+#endif
+
+ for (i = 0; i < 6; i++)
+ {
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE);
+ if (!c)
+ return EAS_FALSE;
+ switch (i)
+ {
+ case 4:
+ if (c == 'n')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVibeState: vibrate on\n"); */ }
+#endif
+ EAS_HWVibrate(pEASData->hwInstData, EAS_TRUE);
+ return EAS_TRUE;
+ }
+ else if (c != 'f')
+ return EAS_FALSE;
+ break;
+
+ case 5:
+ if (c == 'f')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVibeState: vibrate off\n"); */ }
+#endif
+ EAS_HWVibrate(pEASData->hwInstData, EAS_FALSE);
+ return EAS_TRUE;
+ }
+ return EAS_FALSE;
+
+ default:
+ if (c != vibeStr[i])
+ return EAS_FALSE;
+ break;
+ }
+ }
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetBackState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetBackState (S_EAS_DATA *pEASData, S_IMELODY_DATA *pData)
+{
+ EAS_I8 c;
+ EAS_INT i;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetBackState\n"); */ }
+#endif
+
+ for (i = 0; i < 5; i++)
+ {
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_FALSE);
+ if (!c)
+ return EAS_FALSE;
+ switch (i)
+ {
+ case 3:
+ if (c == 'n')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetBackState: backlight on\n"); */ }
+#endif
+ EAS_HWBackLight(pEASData->hwInstData, EAS_TRUE);
+ return EAS_TRUE;
+ }
+ else if (c != 'f')
+ return EAS_FALSE;
+ break;
+
+ case 4:
+ if (c == 'f')
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetBackState: backlight off\n"); */ }
+#endif
+ EAS_HWBackLight(pEASData->hwInstData, EAS_FALSE);
+ return EAS_TRUE;
+ }
+ return EAS_FALSE;
+
+ default:
+ if (c != backStr[i])
+ return EAS_FALSE;
+ break;
+ }
+ }
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetVolume (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader)
+{
+ EAS_INT temp;
+ EAS_I8 c;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetVolume\n"); */ }
+#endif
+
+ c = IMY_GetNextChar(hwInstData, pData, inHeader);
+ if (c == '+')
+ {
+ if (pData->volume < 15)
+ pData->volume++;
+ return EAS_TRUE;
+ }
+ else if (c == '-')
+ {
+ if (pData->volume > 0)
+ pData->volume--;
+ return EAS_TRUE;
+ }
+ else if (IsDigit(c))
+ temp = c - '0';
+ else
+ return EAS_FALSE;
+
+ c = IMY_GetNextChar(hwInstData, pData, inHeader);
+ if (IsDigit(c))
+ temp = temp * 10 + c - '0';
+ else if (c)
+ PutBackChar(pData);
+ if ((temp >= 0) && (temp <= 15))
+ {
+ if (inHeader && (temp == 0))
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring V0 encountered in header\n"); */ }
+ else
+ pData->volume = (EAS_U8) temp;
+ }
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetNumber()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_INT *temp, EAS_BOOL inHeader)
+{
+ EAS_BOOL ok;
+ EAS_I8 c;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_GetNumber\n"); */ }
+#endif
+
+ *temp = 0;
+ ok = EAS_FALSE;
+ for (;;)
+ {
+ c = IMY_GetNextChar(hwInstData, pData, inHeader);
+ if (IsDigit(c))
+ {
+ *temp = *temp * 10 + c - '0';
+ ok = EAS_TRUE;
+ }
+ else
+ {
+ if (c)
+ PutBackChar(pData);
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNumber: value %d\n", *temp); */ }
+#endif
+
+ return ok;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetVersion()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL IMY_GetVersion (S_IMELODY_DATA *pData, EAS_INT *pVersion)
+{
+ EAS_I8 c;
+ EAS_INT temp;
+ EAS_INT version;
+
+ version = temp = 0;
+ for (;;)
+ {
+ c = pData->buffer[pData->index++];
+ if ((c == 0) || (c == '.'))
+ {
+ /*lint -e{701} use shift for performance */
+ version = (version << 8) + temp;
+ if (c == 0)
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetVersion: version 0x%04x\n", version); */ }
+#endif
+
+ *pVersion = version;
+ return EAS_TRUE;
+ }
+ temp = 0;
+ }
+ else if (IsDigit(c))
+ temp = (temp * 10) + c - '0';
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_MetaData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static void IMY_MetaData (S_IMELODY_DATA *pData, E_EAS_METADATA_TYPE metaType, EAS_I8 *buffer)
+{
+ EAS_I32 len;
+
+ /* check for callback */
+ if (!pData->metadata.callback)
+ return;
+
+ /* copy data to host buffer */
+ len = (EAS_I32) strlen((char*) buffer);
+ if (len >pData->metadata.bufferSize)
+ len = pData->metadata.bufferSize;
+ strncpy((char*) pData->metadata.buffer, (char*) buffer, (size_t) len);
+ pData->metadata.buffer[len] = 0;
+
+ /* callback to host */
+ pData->metadata.callback(metaType, pData->metadata.buffer, pData->metadata.pUserData);
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_ParseHeader()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_ParseHeader (S_EAS_DATA *pEASData, S_IMELODY_DATA* pData)
+{
+ EAS_RESULT result;
+ EAS_INT token;
+ EAS_INT temp;
+ EAS_I8 c;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Enter IMY_ParseHeader\n"); */ }
+#endif
+
+ /* initialize some defaults */
+ pData->time = 0;
+ pData->tick = DEFAULT_TICK_CONV;
+ pData->note = 0;
+ pData->noteModifier = 0;
+ pData ->restTicks = 0;
+ pData->volume = 7;
+ pData->octave = 60;
+ pData->repeatOffset = -1;
+ pData->repeatCount = -1;
+ pData->style = 0;
+
+ /* force the read of the first line */
+ pData->index = 1;
+
+ /* read data until we get to melody */
+ for (;;)
+ {
+ /* read a line from the file and parse the token */
+ if (pData->index != 0)
+ {
+ if ((result = IMY_ReadLine(pEASData->hwInstData, pData->fileHandle, pData->buffer, &pData->startLine)) != EAS_SUCCESS)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseHeader: IMY_ReadLine returned %d\n", result); */ }
+#endif
+ return result;
+ }
+ }
+ token = IMY_ParseLine(pData->buffer, &pData->index);
+
+ switch (token)
+ {
+ /* ignore these valid tokens */
+ case TOKEN_BEGIN:
+ break;
+
+ case TOKEN_FORMAT:
+ if (!IMY_GetVersion(pData, &temp))
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid FORMAT field '%s'\n", pData->buffer); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+ if ((temp != 0x0100) && (temp != 0x0200))
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unsupported FORMAT %02x\n", temp); */ }
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+ }
+ break;
+
+ case TOKEN_VERSION:
+ if (!IMY_GetVersion(pData, &temp))
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid VERSION field '%s'\n", pData->buffer); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+ if ((temp != 0x0100) && (temp != 0x0102))
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unsupported VERSION %02x\n", temp); */ }
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+ }
+ break;
+
+ case TOKEN_NAME:
+ IMY_MetaData(pData, EAS_METADATA_TITLE, pData->buffer + pData->index);
+ break;
+
+ case TOKEN_COMPOSER:
+ IMY_MetaData(pData, EAS_METADATA_AUTHOR, pData->buffer + pData->index);
+ break;
+
+ /* handle beat */
+ case TOKEN_BEAT:
+ IMY_GetNumber(pEASData->hwInstData, pData, &temp, EAS_TRUE);
+ if ((temp >= 25) && (temp <= 900))
+ pData->tick = TICK_CONVERT / temp;
+ break;
+
+ /* handle style */
+ case TOKEN_STYLE:
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE);
+ if (c == 'S')
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE);
+ if ((c >= '0') && (c <= '2'))
+ pData->style = (EAS_U8) (c - '0');
+ else
+ {
+ PutBackChar(pData);
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Error in style command: %s\n", pData->buffer); */ }
+ }
+ break;
+
+ /* handle volume */
+ case TOKEN_VOLUME:
+ c = IMY_GetNextChar(pEASData->hwInstData, pData, EAS_TRUE);
+ if (c != 'V')
+ {
+ PutBackChar(pData);
+ if (!IsDigit(c))
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Error in volume command: %s\n", pData->buffer); */ }
+ break;
+ }
+ }
+ IMY_GetVolume(pEASData->hwInstData, pData, EAS_TRUE);
+ break;
+
+ case TOKEN_MELODY:
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Header successfully parsed\n"); */ }
+#endif
+ return EAS_SUCCESS;
+
+ case TOKEN_END:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unexpected END:IMELODY encountered\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+
+ default:
+ /* force a read of the next line */
+ pData->index = 1;
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unrecognized token in iMelody file: %s\n", pData->buffer); */ }
+ break;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_GetNextChar()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_I8 IMY_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_IMELODY_DATA *pData, EAS_BOOL inHeader)
+{
+ EAS_I8 c;
+ EAS_U8 index;
+
+ for (;;)
+ {
+ /* get next character */
+ c = pData->buffer[pData->index++];
+
+ /* buffer empty, read more */
+ if (!c)
+ {
+ /* don't read the next line in the header */
+ if (inHeader)
+ return 0;
+
+ pData->index = 0;
+ pData->buffer[0] = 0;
+ if (IMY_ReadLine(hwInstData, pData->fileHandle, pData->buffer, &pData->startLine) != EAS_SUCCESS)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar: EOF\n"); */ }
+#endif
+ return 0;
+ }
+
+ /* check for END:IMELODY token */
+ if (IMY_ParseLine(pData->buffer, &index) == TOKEN_END)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar: found END:IMELODY\n"); */ }
+#endif
+ pData->buffer[0] = 0;
+ return 0;
+ }
+ continue;
+ }
+
+ /* ignore white space */
+ if (!IsSpace(c))
+ {
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_GetNextChar returned '%c'\n", c); */ }
+#endif
+ return c;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_ReadLine()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reads a line of input from the file, discarding the CR/LF
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT IMY_ReadLine (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I8 *buffer, EAS_I32 *pStartLine)
+{
+ EAS_RESULT result;
+ EAS_INT i;
+ EAS_I8 c;
+
+ /* fetch current file position and save it */
+ if (pStartLine != NULL)
+ {
+ if ((result = EAS_HWFilePos(hwInstData, fileHandle, pStartLine)) != EAS_SUCCESS)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseHeader: EAS_HWFilePos returned %d\n", result); */ }
+#endif
+ return result;
+ }
+ }
+
+ buffer[0] = 0;
+ for (i = 0; i < MAX_LINE_SIZE; )
+ {
+ if ((result = EAS_HWGetByte(hwInstData, fileHandle, &c)) != EAS_SUCCESS)
+ {
+ if ((result == EAS_EOF) && (i > 0))
+ break;
+ return result;
+ }
+
+ /* return on LF or end of data */
+ if (c == '\n')
+ break;
+
+ /* store characters in buffer */
+ if (c != '\r')
+ buffer[i++] = c;
+ }
+ buffer[i] = 0;
+
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ReadLine read %s\n", buffer); */ }
+#endif
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * IMY_ParseLine()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_INT IMY_ParseLine (EAS_I8 *buffer, EAS_U8 *pIndex)
+{
+ EAS_INT i;
+ EAS_INT j;
+
+ /* there's no strnicmp() in stdlib, so we have to roll our own */
+ for (i = 0; i < TOKEN_INVALID; i++)
+ {
+ for (j = 0; ; j++)
+ {
+ /* end of token, must be a match */
+ if (tokens[i][j] == 0)
+ {
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseLine found token %d\n", i); */ }
+#endif
+ *pIndex = (EAS_U8) j;
+ return i;
+ }
+ if (tokens[i][j] != ToUpper(buffer[j]))
+ break;
+ }
+ }
+#ifdef _DEBUG_IMELODY
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IMY_ParseLine: no token found\n"); */ }
+#endif
+ return TOKEN_INVALID;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_imelodydata.c b/arm-wt-22k/lib_src/eas_imelodydata.c new file mode 100644 index 0000000..e72dc0b --- /dev/null +++ b/arm-wt-22k/lib_src/eas_imelodydata.c @@ -0,0 +1,43 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imelodydata.c
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_imelodydata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_iMelodyData
+ *
+ * Static memory allocation for iMelody parser
+ *----------------------------------------------------------------------------
+*/
+S_IMELODY_DATA eas_iMelodyData;
+
diff --git a/arm-wt-22k/lib_src/eas_imelodydata.h b/arm-wt-22k/lib_src/eas_imelodydata.h new file mode 100644 index 0000000..303b8f6 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_imelodydata.h @@ -0,0 +1,73 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_imelodydata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data declarations for the iMelody parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 778 $
+ * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_IMELODYDATA_H
+#define EAS_IMELODYDATA_H
+
+#include "eas_data.h"
+
+/* maximum line size as specified in iMelody V1.2 spec */
+#define MAX_LINE_SIZE 75
+
+/*----------------------------------------------------------------------------
+ *
+ * S_IMELODY_DATA
+ *
+ * This structure contains the state data for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+ EAS_FILE_HANDLE fileHandle; /* file handle */
+ S_SYNTH *pSynth; /* pointer to synth */
+ EAS_I32 fileOffset; /* offset to start of data */
+ EAS_I32 time; /* current time in 256ths of a msec */
+ EAS_I32 tickBase; /* basline length of 32nd note in 256th of a msec */
+ EAS_I32 tick; /* actual length of 32nd note in 256th of a msec */
+ EAS_I32 restTicks; /* ticks to rest after current note */
+ EAS_I32 startLine; /* file offset at start of line (for repeats) */
+ EAS_I32 repeatOffset; /* file offset to start of repeat section */
+ S_METADATA_CB metadata; /* metadata callback */
+ EAS_I16 repeatCount; /* repeat counter */
+ EAS_U8 state; /* current state EAS_STATE_XXXX */
+ EAS_U8 style; /* from STYLE */
+ EAS_U8 index; /* index into buffer */
+ EAS_U8 octave; /* octave prefix */
+ EAS_U8 volume; /* current volume */
+ EAS_U8 note; /* MIDI note number */
+ EAS_I8 noteModifier; /* sharp or flat */
+ EAS_I8 buffer[MAX_LINE_SIZE+1]; /* buffer for ASCII data */
+} S_IMELODY_DATA;
+
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_math.c b/arm-wt-22k/lib_src/eas_math.c new file mode 100644 index 0000000..12d788e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_math.c @@ -0,0 +1,168 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_math.c
+ *
+ * Contents and purpose:
+ * Contains common math routines for the various audio engines.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 586 $
+ * $Date: 2007-03-08 20:33:04 -0800 (Thu, 08 Mar 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas.h"
+#include "eas_math.h"
+
+/* anything less than this converts to a fraction too small to represent in 32-bits */
+#define MIN_CENTS -18000
+
+/*----------------------------------------------------------------------------
+ * EAS_Calculate2toX()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate 2^x
+ *
+ * Inputs:
+ * nCents - measured in cents
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_Calculate2toX (EAS_I32 nCents)
+{
+ EAS_I32 nDents;
+ EAS_I32 nExponentInt, nExponentFrac;
+ EAS_I32 nTemp1, nTemp2;
+ EAS_I32 nResult;
+
+ /* check for minimum value */
+ if (nCents < MIN_CENTS)
+ return 0;
+
+ /* for the time being, convert cents to dents */
+ nDents = FMUL_15x15(nCents, CENTS_TO_DENTS);
+
+ nExponentInt = GET_DENTS_INT_PART(nDents);
+ nExponentFrac = GET_DENTS_FRAC_PART(nDents);
+
+ /*
+ implement 2^(fracPart) as a power series
+ */
+ nTemp1 = GN2_TO_X2 + MULT_DENTS_COEF(nExponentFrac, GN2_TO_X3);
+ nTemp2 = GN2_TO_X1 + MULT_DENTS_COEF(nExponentFrac, nTemp1);
+ nTemp1 = GN2_TO_X0 + MULT_DENTS_COEF(nExponentFrac, nTemp2);
+
+ /*
+ implement 2^(intPart) as
+ a left shift for intPart >= 0 or
+ a left shift for intPart < 0
+ */
+ if (nExponentInt >= 0)
+ {
+ /* left shift for positive exponents */
+ /*lint -e{703} <avoid multiply for performance>*/
+ nResult = nTemp1 << nExponentInt;
+ }
+ else
+ {
+ /* right shift for negative exponents */
+ nExponentInt = -nExponentInt;
+ nResult = nTemp1 >> nExponentInt;
+ }
+
+ return nResult;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_LogToLinear16()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform log value to linear gain multiplier using piece-wise linear
+ * approximation
+ *
+ * Inputs:
+ * nGain - log scale value in 20.10 format. Even though gain is normally
+ * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate
+ * the need for saturation checking when combining gain values.
+ *
+ * Outputs:
+ * Returns a 16-bit linear value approximately equal to 2^(nGain/1024)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain)
+{
+ EAS_INT nExp;
+ EAS_U16 nTemp;
+
+ /* bias to positive */
+ nGain += 32767;
+
+ /* check for infinite attenuation */
+ if (nGain < 0)
+ return 0;
+
+ /* extract the exponent */
+ nExp = 31 - (nGain >> 10);
+
+ /* check for maximum output */
+ if (nExp < 0)
+ return 0x7fff;
+
+ /* extract mantissa and restore implied 1 bit */
+ nTemp = (EAS_U16)((((nGain & 0x3ff) << 4) | 0x4000) >> nExp);
+
+ /* use shift to approximate power-of-2 operation */
+ return nTemp;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_VolumeToGain()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform volume control in 1dB increments to gain multiplier
+ *
+ * Inputs:
+ * volume - 100 = 0dB, 99 = -1dB, 0 = -inf
+ *
+ * Outputs:
+ * Returns a 16-bit linear value
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 EAS_VolumeToGain (EAS_INT volume)
+{
+ /* check for limits */
+ if (volume <= 0)
+ return 0;
+ if (volume >= 100)
+ return 0x7fff;
+
+ /*lint -e{702} use shift instead of division */
+ return (EAS_I16) EAS_Calculate2toX((((volume - EAS_MAX_VOLUME) * 204099) >> 10) - 1);
+}
+
diff --git a/arm-wt-22k/lib_src/eas_math.h b/arm-wt-22k/lib_src/eas_math.h new file mode 100644 index 0000000..719270b --- /dev/null +++ b/arm-wt-22k/lib_src/eas_math.h @@ -0,0 +1,412 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_math.h
+ *
+ * Contents and purpose:
+ * Contains common math routines for the various audio engines.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 584 $
+ * $Date: 2007-03-08 09:49:24 -0800 (Thu, 08 Mar 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MATH_H
+#define _EAS_MATH_H
+
+
+/** coefs for pan, generates sin, cos */
+#define COEFF_PAN_G2 -27146 /* -0.82842712474619 = 2 - 4/sqrt(2) */
+#define COEFF_PAN_G0 23170 /* 0.707106781186547 = 1/sqrt(2) */
+
+/*
+coefficients for approximating
+2^x = gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3
+where x is a int.frac number representing number of octaves.
+Actually, we approximate only the 2^(frac) using the power series
+and implement the 2^(int) as a shift, so that
+2^x == 2^(int.frac) == 2^(int) * 2^(fract)
+ == (gn2toX0 + gn2toX1*x + gn2toX2*x^2 + gn2toX3*x^3) << (int)
+
+The gn2toX.. were generated using a best fit for a 3rd
+order polynomial, instead of taking the coefficients from
+a truncated Taylor (or Maclaurin?) series.
+*/
+
+#define GN2_TO_X0 32768 /* 1 */
+#define GN2_TO_X1 22833 /* 0.696807861328125 */
+#define GN2_TO_X2 7344 /* 0.22412109375 */
+#define GN2_TO_X3 2588 /* 0.0789794921875 */
+
+/*----------------------------------------------------------------------------
+ * Fixed Point Math
+ *----------------------------------------------------------------------------
+ * These macros are used for fixed point multiplies. If the processor
+ * supports fixed point multiplies, replace these macros with inline
+ * assembly code to improve performance.
+ *----------------------------------------------------------------------------
+*/
+
+/* Fixed point multiply 0.15 x 0.15 = 0.15 returned as 32-bits */
+#define FMUL_15x15(a,b) \
+ /*lint -e(704) <avoid multiply for performance>*/ \
+ (((EAS_I32)(a) * (EAS_I32)(b)) >> 15)
+
+/* Fixed point multiply 0.7 x 0.7 = 0.15 returned as 32-bits */
+#define FMUL_7x7(a,b) \
+ /*lint -e(704) <avoid multiply for performance>*/ \
+ (((EAS_I32)(a) * (EAS_I32)(b) ) << 1)
+
+/* Fixed point multiply 0.8 x 0.8 = 0.15 returned as 32-bits */
+#define FMUL_8x8(a,b) \
+ /*lint -e(704) <avoid multiply for performance>*/ \
+ (((EAS_I32)(a) * (EAS_I32)(b) ) >> 1)
+
+/* Fixed point multiply 0.8 x 1.15 = 0.15 returned as 32-bits */
+#define FMUL_8x15(a,b) \
+ /*lint -e(704) <avoid divide for performance>*/ \
+ (((EAS_I32)((a) << 7) * (EAS_I32)(b)) >> 15)
+
+/* macros for fractional phase accumulator */
+/*
+Note: changed the _U32 to _I32 on 03/14/02. This should not
+affect the phase calculations, and should allow us to reuse these
+macros for other audio sample related math.
+*/
+#define HARDWARE_BIT_WIDTH 32
+
+#define NUM_PHASE_INT_BITS 1
+#define NUM_PHASE_FRAC_BITS 15
+
+#define PHASE_FRAC_MASK (EAS_U32) ((0x1L << NUM_PHASE_FRAC_BITS) -1)
+
+#define GET_PHASE_INT_PART(x) (EAS_U32)((EAS_U32)(x) >> NUM_PHASE_FRAC_BITS)
+#define GET_PHASE_FRAC_PART(x) (EAS_U32)((EAS_U32)(x) & PHASE_FRAC_MASK)
+
+#define DEFAULT_PHASE_FRAC 0
+#define DEFAULT_PHASE_INT 0
+
+/*
+Linear interpolation calculates:
+output = (1-frac) * sample[n] + (frac) * sample[n+1]
+
+where conceptually 0 <= frac < 1
+
+For a fixed point implementation, frac is actually an integer value
+with an implied binary point one position to the left. The value of
+one (unity) is given by PHASE_ONE
+one half and one quarter are useful for 4-point linear interp.
+*/
+#define PHASE_ONE (EAS_I32) (0x1L << NUM_PHASE_FRAC_BITS)
+
+/*
+ Multiply the signed audio sample by the unsigned fraction.
+- a is the signed audio sample
+- b is the unsigned fraction (cast to signed int as long as coef
+ uses (n-1) or less bits, where n == hardware bit width)
+*/
+#define MULT_AUDIO_COEF(audio,coef) /*lint -e704 <avoid divide for performance>*/ \
+ (EAS_I32)( \
+ ( \
+ ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \
+ ) \
+ >> NUM_PHASE_FRAC_BITS \
+ ) \
+ /* lint +704 <restore checking>*/
+
+/* wet / dry calculation macros */
+#define NUM_WET_DRY_FRAC_BITS 7 // 15
+#define NUM_WET_DRY_INT_BITS 9 // 1
+
+/* define a 1.0 */
+#define WET_DRY_ONE (EAS_I32) ((0x1L << NUM_WET_DRY_FRAC_BITS))
+#define WET_DRY_MINUS_ONE (EAS_I32) (~WET_DRY_ONE)
+#define WET_DRY_FULL_SCALE (EAS_I32) (WET_DRY_ONE - 1)
+
+#define MULT_AUDIO_WET_DRY_COEF(audio,coef) /*lint -e(702) <avoid divide for performance>*/ \
+ (EAS_I32)( \
+ ( \
+ ((EAS_I32)(audio)) * ((EAS_I32)(coef)) \
+ ) \
+ >> NUM_WET_DRY_FRAC_BITS \
+ )
+
+/* Envelope 1 (EG1) calculation macros */
+#define NUM_EG1_INT_BITS 1
+#define NUM_EG1_FRAC_BITS 15
+
+/* the max positive gain used in the synth for EG1 */
+/* SYNTH_FULL_SCALE_EG1_GAIN must match the value in the dls2eas
+converter, otherwise, the values we read from the .eas file are bogus. */
+#define SYNTH_FULL_SCALE_EG1_GAIN (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS) -1)
+
+/* define a 1.0 */
+#define EG1_ONE (EAS_I32) ((0x1L << NUM_EG1_FRAC_BITS))
+#define EG1_MINUS_ONE (EAS_I32) (~SYNTH_FULL_SCALE_EG1_GAIN)
+
+#define EG1_HALF (EAS_I32) (EG1_ONE/2)
+#define EG1_MINUS_HALF (EAS_I32) (EG1_MINUS_ONE/2)
+
+/*
+We implement the EG1 using a linear gain value, which means that the
+attack segment is handled by incrementing (adding) the linear gain.
+However, EG1 treats the Decay, Sustain, and Release differently than
+the Attack portion. For Decay, Sustain, and Release, the gain is
+linear on dB scale, which is equivalent to exponential damping on
+a linear scale. Because we use a linear gain for EG1, we implement
+the Decay and Release as multiplication (instead of incrementing
+as we did for the attack segment).
+Therefore, we need the following macro to implement the multiplication
+(i.e., exponential damping) during the Decay and Release segments of
+the EG1
+*/
+#define MULT_EG1_EG1(gain,damping) /*lint -e(704) <avoid divide for performance>*/ \
+ (EAS_I32)( \
+ ( \
+ ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \
+ ) \
+ >> NUM_EG1_FRAC_BITS \
+ )
+
+// Use the following macro specifically for the filter, when multiplying
+// the b1 coefficient. The 0 <= |b1| < 2, which therefore might overflow
+// in certain conditions because we store b1 as a 1.15 value.
+// Instead, we could store b1 as b1p (b1' == b1 "prime") where
+// b1p == b1/2, thus ensuring no potential overflow for b1p because
+// 0 <= |b1p| < 1
+// However, during the filter calculation, we must account for the fact
+// that we are using b1p instead of b1, and thereby multiply by
+// an extra factor of 2. Rather than multiply by an extra factor of 2,
+// we can instead shift the result right by one less, hence the
+// modified shift right value of (NUM_EG1_FRAC_BITS -1)
+#define MULT_EG1_EG1_X2(gain,damping) /*lint -e(702) <avoid divide for performance>*/ \
+ (EAS_I32)( \
+ ( \
+ ((EAS_I32)(gain)) * ((EAS_I32)(damping)) \
+ ) \
+ >> (NUM_EG1_FRAC_BITS -1) \
+ )
+
+#define SATURATE_EG1(x) /*lint -e{734} saturation operation */ \
+ ((EAS_I32)(x) > SYNTH_FULL_SCALE_EG1_GAIN) ? (SYNTH_FULL_SCALE_EG1_GAIN) : \
+ ((EAS_I32)(x) < EG1_MINUS_ONE) ? (EG1_MINUS_ONE) : (x);
+
+
+/* use "digital cents" == "dents" instead of cents */
+/* we coudl re-use the phase frac macros, but if we do,
+we must change the phase macros to cast to _I32 instead of _U32,
+because using a _U32 cast causes problems when shifting the exponent
+for the 2^x calculation, because right shift a negative values MUST
+be sign extended, or else the 2^x calculation is wrong */
+
+/* use "digital cents" == "dents" instead of cents */
+#define NUM_DENTS_FRAC_BITS 12
+#define NUM_DENTS_INT_BITS (HARDWARE_BIT_WIDTH - NUM_DENTS_FRAC_BITS)
+
+#define DENTS_FRAC_MASK (EAS_I32) ((0x1L << NUM_DENTS_FRAC_BITS) -1)
+
+#define GET_DENTS_INT_PART(x) /*lint -e(704) <avoid divide for performance>*/ \
+ (EAS_I32)((EAS_I32)(x) >> NUM_DENTS_FRAC_BITS)
+
+#define GET_DENTS_FRAC_PART(x) (EAS_I32)((EAS_I32)(x) & DENTS_FRAC_MASK)
+
+#define DENTS_ONE (EAS_I32) (0x1L << NUM_DENTS_FRAC_BITS)
+
+/* use CENTS_TO_DENTS to convert a value in cents to dents */
+#define CENTS_TO_DENTS (EAS_I32) (DENTS_ONE * (0x1L << NUM_EG1_FRAC_BITS) / 1200L) \
+
+
+/*
+For gain, the LFO generates a value that modulates in terms
+of dB. However, we use a linear gain value, so we must convert
+the LFO value in dB to a linear gain. Normally, we would use
+linear gain = 10^x, where x = LFO value in dB / 20.
+Instead, we implement 10^x using our 2^x approximation.
+because
+
+ 10^x = 2^(log2(10^x)) = 2^(x * log2(10))
+
+so we need to multiply by log2(10) which is just a constant.
+Ah, but just wait -- our 2^x actually doesn't exactly implement
+2^x, but it actually assumes that the input is in cents, and within
+the 2^x approximation converts its input from cents to octaves
+by dividing its input by 1200.
+
+So, in order to convert the LFO gain value in dB to something
+that our existing 2^x approximation can use, multiply the LFO gain
+by log2(10) * 1200 / 20
+
+The divide by 20 helps convert dB to linear gain, and we might
+as well incorporate that operation into this conversion.
+Of course, we need to keep some fractional bits, so multiply
+the constant by NUM_EG1_FRAC_BITS
+*/
+
+/* use LFO_GAIN_TO_CENTS to convert the LFO gain value to cents */
+#if 0
+#define DOUBLE_LOG2_10 (double) (3.32192809488736) /* log2(10) */
+
+#define DOUBLE_LFO_GAIN_TO_CENTS (double) \
+ ( \
+ (DOUBLE_LOG2_10) * \
+ 1200.0 / \
+ 20.0 \
+ )
+
+#define LFO_GAIN_TO_CENTS (EAS_I32) \
+ ( \
+ DOUBLE_LFO_GAIN_TO_CENTS * \
+ (0x1L << NUM_EG1_FRAC_BITS) \
+ )
+#endif
+
+#define LFO_GAIN_TO_CENTS (EAS_I32) (1671981156L >> (23 - NUM_EG1_FRAC_BITS))
+
+
+#define MULT_DENTS_COEF(dents,coef) /*lint -e704 <avoid divide for performance>*/ \
+ (EAS_I32)( \
+ ( \
+ ((EAS_I32)(dents)) * ((EAS_I32)(coef)) \
+ ) \
+ >> NUM_DENTS_FRAC_BITS \
+ ) \
+ /* lint +e704 <restore checking>*/
+
+/* we use 16-bits in the PC per audio sample */
+#define BITS_PER_AUDIO_SAMPLE 16
+
+/* we define 1 as 1.0 - 1 LSbit */
+#define DISTORTION_ONE (EAS_I32)((0x1L << (BITS_PER_AUDIO_SAMPLE-1)) -1)
+#define DISTORTION_MINUS_ONE (EAS_I32)(~DISTORTION_ONE)
+
+/* drive coef is given as int.frac */
+#define NUM_DRIVE_COEF_INT_BITS 1
+#define NUM_DRIVE_COEF_FRAC_BITS 4
+
+#define MULT_AUDIO_DRIVE(audio,drive) /*lint -e(702) <avoid divide for performance>*/ \
+ (EAS_I32) ( \
+ ( \
+ ((EAS_I32)(audio)) * ((EAS_I32)(drive)) \
+ ) \
+ >> NUM_DRIVE_COEF_FRAC_BITS \
+ )
+
+#define MULT_AUDIO_AUDIO(audio1,audio2) /*lint -e(702) <avoid divide for performance>*/ \
+ (EAS_I32) ( \
+ ( \
+ ((EAS_I32)(audio1)) * ((EAS_I32)(audio2)) \
+ ) \
+ >> (BITS_PER_AUDIO_SAMPLE-1) \
+ )
+
+#define SATURATE(x) \
+ ((((EAS_I32)(x)) > DISTORTION_ONE) ? (DISTORTION_ONE) : \
+ (((EAS_I32)(x)) < DISTORTION_MINUS_ONE) ? (DISTORTION_MINUS_ONE) : ((EAS_I32)(x)));
+
+
+
+/*----------------------------------------------------------------------------
+ * EAS_Calculate2toX()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate 2^x
+ *
+ * Inputs:
+ * nCents - measured in cents
+ *
+ * Outputs:
+ * nResult - int.frac result (where frac has NUM_DENTS_FRAC_BITS)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_Calculate2toX (EAS_I32 nCents);
+
+/*----------------------------------------------------------------------------
+ * EAS_LogToLinear16()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform log value to linear gain multiplier using piece-wise linear
+ * approximation
+ *
+ * Inputs:
+ * nGain - log scale value in 20.10 format. Even though gain is normally
+ * stored in 6.10 (16-bit) format we use 32-bit numbers here to eliminate
+ * the need for saturation checking when combining gain values.
+ *
+ * Outputs:
+ * Returns a 16-bit linear value approximately equal to 2^(nGain/1024)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_LogToLinear16 (EAS_I32 nGain);
+
+/*----------------------------------------------------------------------------
+ * EAS_VolumeToGain()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform volume control in 1dB increments to gain multiplier
+ *
+ * Inputs:
+ * volume - 100 = 0dB, 99 = -1dB, 0 = -inf
+ *
+ * Outputs:
+ * Returns a 16-bit linear value
+ *----------------------------------------------------------------------------
+*/
+EAS_I16 EAS_VolumeToGain (EAS_INT volume);
+
+/*----------------------------------------------------------------------------
+ * EAS_fsqrt()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the square root of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the square root of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_U16 EAS_fsqrt (EAS_U32 n);
+
+/*----------------------------------------------------------------------------
+ * EAS_flog2()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the log2 of a 32-bit fixed point value
+ *
+ * Inputs:
+ * n = value of interest
+ *
+ * Outputs:
+ * returns the log2 of n
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_flog2 (EAS_U32 n);
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_mdls.c b/arm-wt-22k/lib_src/eas_mdls.c new file mode 100644 index 0000000..3da3b06 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mdls.c @@ -0,0 +1,2668 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_mdls.c + * + * Contents and purpose: + * This file contains DLS to EAS converter. + * + * Copyright (c) 2005 Sonic Network Inc. + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 818 $ + * $Date: 2007-08-02 15:19:41 -0700 (Thu, 02 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +/* + * NOTES: + * + * Processor Endian-ness: + * + * We use the EAS_HWGetDWord() and EAS_HWGetWord () functions + * extensively in this module. It would probably be faster to read + * an entire data structure, but this introduces the problem of + * sensitivity to processor endian-ness to the parser. By utlilizing + * the host wrapper functions, we avoid having to flip bytes around + * for big-endian processors. The default host wrapper versions of + * these functions are insensitive to processor endian-ness due to + * the fact that they read the file as a byte stream. + * + * Dynamic Memory: + * + * Dynamic memory allocation is a risky proposition in a mobile + * device. The memory can become fragmented, resulting in an + * inability to allocate a memory block, or garbage collection + * routines can use many CPU cycles. Either can contribute to + * failures of critical systems. Therefore, we try to minimize the + * number of memory allocations we make. + * + * We allocate a single large block of memory for the entire + * converted DLS collection, including the articulation data and + * samples. This block is then sub-allocated for the various + * data structures. + * + * Parser Overview: + * + * We make two passes through the file, the first pass to count the + * number of instruments, regions, etc. and allocate memory for + * them. The second pass parses the data into the allocated data + * structures. + * + * Conditional chunks are challenging in that they can occur + * anywhere in the list chunk that contains them. To simplify, we + * parse the blocks in a list in specific order, no matter which + * order they appear in the file. This way we don't allocate memory + * and parse a block that we end up throwing away later due to + * a conditional chunk. + * + * Assumptions that may bite us in the future: + * + * We make some assumptions to simplify things. The most fundamental + * assumption is that there will be no more than one of any type of + * chunk in a list. While this is consistent with the block diagram + * of the file layout in the mDLS spec, there is nothing in the + * spec that precludes having mulitple lar2 or rgn2 chunks, with + * conditional blocks that dictate their usage. + * + * DLS -> EAS Conversion Process: + * + * Another challenge is that the DLS structure does not map well to + * the current EAS sound library structure. Not all DLS constructs + * are supported, and data from DLS structures must sometimes be + * mapped to multiple EAS data structures. To simplify the process, + * the EAS region, articulation, and envelopes are treated as a + * single combined unit. Thus for each region, there must be one + * articulation element and two envelope elements. + * + * The sample processing is also a multi-step process. First the + * ptbl chunk is pre-parsed to determine the number of samples + * in the collection. The next step is to parse the instrument data + * to determine which samples are actually used by instruments. + * Some samples may not be used because they are used only in + * conditional blocks that the synthesizer cannot parse, or the + * author neglected to remove unused samples from the collection. + * In the next step, the active samples are read into memory and + * converted to the appropriate playback format. Finally, as the + * instruments are processed, the links are made to the samples and + * wsmp data is extracted for the region and articulation data + * structures. +*/ + +#ifndef _FILTER_ENABLED +#error "Filter must be enabled if DLS_SYNTHESIZER is enabled" +#endif + +/*------------------------------------ + * includes + *------------------------------------ +*/ + +/* this define allows us to use the sndlib.h structures as RW memory */ +#define SCNST + +#include "eas_data.h" +#include "eas_host.h" +#include "eas_mdls.h" +#include "eas_math.h" +#include "dls.h" +#include "dls2.h" +#include "eas_report.h" + +//2 we should replace log10() function with fixed point routine in ConvertSampleRate() +/* lint is choking on the ARM math.h file, so we declare the log10 function here */ +extern double log10(double x); + +/*------------------------------------ + * defines + *------------------------------------ +*/ + +// #define _DEBUG_DLS + +#define DLS_MAX_WAVE_COUNT 1024 +#define DLS_MAX_ART_COUNT 2048 +#define DLS_MAX_REGION_COUNT 2048 +#define DLS_MAX_INST_COUNT 256 +#define MAX_DLS_WAVE_SIZE (1024*1024) + +/*------------------------------------ + * typedefs + *------------------------------------ +*/ + +/* offsets to articulation data */ +typedef enum +{ + PARAM_MODIFIED = 0, + PARAM_MOD_LFO_FREQ, + PARAM_MOD_LFO_DELAY, + + PARAM_VIB_LFO_FREQ, + PARAM_VIB_LFO_DELAY, + + PARAM_VOL_EG_DELAY, + PARAM_VOL_EG_ATTACK, + PARAM_VOL_EG_HOLD, + PARAM_VOL_EG_DECAY, + PARAM_VOL_EG_SUSTAIN, + PARAM_VOL_EG_RELEASE, + PARAM_VOL_EG_SHUTDOWN, + PARAM_VOL_EG_VEL_TO_ATTACK, + PARAM_VOL_EG_KEY_TO_DECAY, + PARAM_VOL_EG_KEY_TO_HOLD, + + PARAM_MOD_EG_DELAY, + PARAM_MOD_EG_ATTACK, + PARAM_MOD_EG_HOLD, + PARAM_MOD_EG_DECAY, + PARAM_MOD_EG_SUSTAIN, + PARAM_MOD_EG_RELEASE, + PARAM_MOD_EG_VEL_TO_ATTACK, + PARAM_MOD_EG_KEY_TO_DECAY, + PARAM_MOD_EG_KEY_TO_HOLD, + + PARAM_INITIAL_FC, + PARAM_INITIAL_Q, + PARAM_MOD_LFO_TO_FC, + PARAM_MOD_LFO_CC1_TO_FC, + PARAM_MOD_LFO_CHAN_PRESS_TO_FC, + PARAM_MOD_EG_TO_FC, + PARAM_VEL_TO_FC, + PARAM_KEYNUM_TO_FC, + + PARAM_MOD_LFO_TO_GAIN, + PARAM_MOD_LFO_CC1_TO_GAIN, + PARAM_MOD_LFO_CHAN_PRESS_TO_GAIN, + PARAM_VEL_TO_GAIN, + + PARAM_TUNING, + PARAM_KEYNUM_TO_PITCH, + PARAM_VIB_LFO_TO_PITCH, + PARAM_VIB_LFO_CC1_TO_PITCH, + PARAM_VIB_LFO_CHAN_PRESS_TO_PITCH, + PARAM_MOD_LFO_TO_PITCH, + PARAM_MOD_LFO_CC1_TO_PITCH, + PARAM_MOD_LFO_CHAN_PRESS_TO_PITCH, + PARAM_MOD_EG_TO_PITCH, + + PARAM_DEFAULT_PAN, + PARAM_MIDI_CC91_TO_REVERB_SEND, + PARAM_DEFAULT_REVERB_SEND, + PARAM_MIDI_CC93_TO_CHORUS_SEND, + PARAM_DEFAULT_CHORUS_SEND, + PARAM_TABLE_SIZE +} E_ART_INDEX; + +/* temporary data structure combining region, articulation, and envelope data */ +typedef struct s_art_dls_tag +{ + EAS_I16 values[PARAM_TABLE_SIZE]; +} S_DLS_ART_VALUES; + +/* temporary data structure for wlnk chunk data */ +typedef struct +{ + EAS_I32 gain; + EAS_U32 loopStart; + EAS_U32 loopLength; + EAS_U32 sampleRate; + EAS_U16 bitsPerSample; + EAS_I16 fineTune; + EAS_U8 unityNote; +} S_WSMP_DATA; + +/* temporary data structure used while parsing a DLS file */ +typedef struct +{ + S_DLS *pDLS; + EAS_HW_DATA_HANDLE hwInstData; + EAS_FILE_HANDLE fileHandle; + S_WSMP_DATA *wsmpData; + EAS_U32 instCount; + EAS_U32 regionCount; + EAS_U32 artCount; + EAS_U32 waveCount; + EAS_U32 wavePoolSize; + EAS_U32 wavePoolOffset; + EAS_BOOL bigEndian; + EAS_BOOL filterUsed; +} SDLS_SYNTHESIZER_DATA; + +/* connection lookup table */ +typedef struct s_connection_tag +{ + EAS_U16 source; + EAS_U16 control; + EAS_U16 destination; + EAS_U16 connection; +} S_CONNECTION; + +static const S_CONNECTION connTable[] = +{ + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_LFO_FREQUENCY, PARAM_MOD_LFO_FREQ }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_LFO_STARTDELAY, PARAM_MOD_LFO_DELAY}, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_VIB_FREQUENCY, PARAM_VIB_LFO_FREQ }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_VIB_STARTDELAY, PARAM_VIB_LFO_DELAY }, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_DELAYTIME, PARAM_VOL_EG_DELAY }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_ATTACKTIME, PARAM_VOL_EG_ATTACK }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_HOLDTIME, PARAM_VOL_EG_HOLD }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_DECAYTIME, PARAM_VOL_EG_DECAY }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_SUSTAINLEVEL, PARAM_VOL_EG_SUSTAIN }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_RELEASETIME, PARAM_VOL_EG_RELEASE }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG1_SHUTDOWNTIME, PARAM_VOL_EG_SHUTDOWN }, + { CONN_SRC_KEYONVELOCITY, CONN_SRC_NONE, CONN_DST_EG1_ATTACKTIME, PARAM_VOL_EG_VEL_TO_ATTACK }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_EG1_DECAYTIME, PARAM_VOL_EG_KEY_TO_DECAY }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_EG1_HOLDTIME, PARAM_VOL_EG_KEY_TO_HOLD }, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_DELAYTIME, PARAM_MOD_EG_DELAY }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_ATTACKTIME, PARAM_MOD_EG_ATTACK }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_HOLDTIME, PARAM_MOD_EG_HOLD }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_DECAYTIME, PARAM_MOD_EG_DECAY }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_SUSTAINLEVEL, PARAM_MOD_EG_SUSTAIN }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_EG2_RELEASETIME, PARAM_MOD_EG_RELEASE }, + { CONN_SRC_KEYONVELOCITY, CONN_SRC_NONE, CONN_DST_EG2_ATTACKTIME, PARAM_MOD_EG_VEL_TO_ATTACK }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_EG2_DECAYTIME, PARAM_MOD_EG_KEY_TO_DECAY }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_EG2_HOLDTIME, PARAM_MOD_EG_KEY_TO_HOLD }, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_FILTER_CUTOFF, PARAM_INITIAL_FC }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_FILTER_Q, PARAM_INITIAL_Q }, + { CONN_SRC_LFO, CONN_SRC_NONE, CONN_DST_FILTER_CUTOFF, PARAM_MOD_LFO_TO_FC }, + { CONN_SRC_LFO, CONN_SRC_CC1, CONN_DST_FILTER_CUTOFF, PARAM_MOD_LFO_CC1_TO_FC }, + { CONN_SRC_LFO, CONN_SRC_CHANNELPRESSURE, CONN_DST_FILTER_CUTOFF, PARAM_MOD_LFO_CHAN_PRESS_TO_FC }, + { CONN_SRC_EG2, CONN_SRC_NONE, CONN_DST_FILTER_CUTOFF, PARAM_MOD_EG_TO_FC }, + { CONN_SRC_KEYONVELOCITY, CONN_SRC_NONE, CONN_DST_FILTER_CUTOFF, PARAM_VEL_TO_FC }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_FILTER_CUTOFF, PARAM_KEYNUM_TO_FC }, + + { CONN_SRC_LFO, CONN_SRC_NONE, CONN_DST_GAIN, PARAM_MOD_LFO_TO_GAIN }, + { CONN_SRC_LFO, CONN_SRC_CC1, CONN_DST_GAIN, PARAM_MOD_LFO_CC1_TO_GAIN }, + { CONN_SRC_LFO, CONN_SRC_CHANNELPRESSURE, CONN_DST_GAIN, PARAM_MOD_LFO_CHAN_PRESS_TO_GAIN }, + { CONN_SRC_KEYONVELOCITY, CONN_SRC_NONE, CONN_DST_GAIN, PARAM_VEL_TO_GAIN }, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_PITCH, PARAM_TUNING }, + { CONN_SRC_KEYNUMBER, CONN_SRC_NONE, CONN_DST_PITCH, PARAM_KEYNUM_TO_PITCH }, + { CONN_SRC_VIBRATO, CONN_SRC_NONE, CONN_DST_PITCH, PARAM_VIB_LFO_TO_PITCH }, + { CONN_SRC_VIBRATO, CONN_SRC_CC1, CONN_DST_PITCH, PARAM_VIB_LFO_CC1_TO_PITCH }, + { CONN_SRC_VIBRATO, CONN_SRC_CHANNELPRESSURE, CONN_DST_PITCH, PARAM_VIB_LFO_CHAN_PRESS_TO_PITCH }, + { CONN_SRC_LFO, CONN_SRC_NONE, CONN_DST_PITCH, PARAM_MOD_LFO_TO_PITCH }, + { CONN_SRC_LFO, CONN_SRC_CC1, CONN_DST_PITCH, PARAM_MOD_LFO_CC1_TO_PITCH }, + { CONN_SRC_LFO, CONN_SRC_CHANNELPRESSURE, CONN_DST_PITCH, PARAM_MOD_LFO_CHAN_PRESS_TO_PITCH }, + { CONN_SRC_EG2, CONN_SRC_NONE, CONN_DST_PITCH, PARAM_MOD_EG_TO_PITCH }, + + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_PAN, PARAM_DEFAULT_PAN }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_REVERB, PARAM_DEFAULT_REVERB_SEND }, + { CONN_SRC_CC91, CONN_SRC_NONE, CONN_DST_REVERB, PARAM_MIDI_CC91_TO_REVERB_SEND }, + { CONN_SRC_NONE, CONN_SRC_NONE, CONN_DST_CHORUS, PARAM_DEFAULT_CHORUS_SEND }, + { CONN_SRC_CC93, CONN_SRC_NONE, CONN_DST_REVERB, PARAM_MIDI_CC93_TO_CHORUS_SEND } +}; +#define ENTRIES_IN_CONN_TABLE (sizeof(connTable)/sizeof(S_CONNECTION)) + +static const S_DLS_ART_VALUES defaultArt = +{ + 0, /* not modified */ + -851, /* Mod LFO frequency: 5 Hz */ + -7973, /* Mod LFO delay: 10 milliseconds */ + + -851, /* Vib LFO frequency: 5 Hz */ + -7973, /* Vib LFO delay: 10 milliseconds */ + + -32768, /* EG1 delay time: 0 secs */ + -32768, /* EG1 attack time: 0 secs */ + -32768, /* EG1 hold time: 0 secs */ + -32768, /* EG1 decay time: 0 secs */ + 1000, /* EG1 sustain level: 100.0% */ + -32768, /* EG1 release time: 0 secs */ + -7271, /* EG1 shutdown time: 15 msecs */ + 0, /* EG1 velocity to attack: 0 time cents */ + 0, /* EG1 key number to decay: 0 time cents */ + 0, /* EG1 key number to hold: 0 time cents */ + + -32768, /* EG2 delay time: 0 secs */ + -32768, /* EG2 attack time: 0 secs */ + -32768, /* EG2 hold time: 0 secs */ + -32768, /* EG2 decay time: 0 secs */ + 1000, /* EG2 sustain level: 100.0% */ + -32768, /* EG2 release time: 0 secs */ + 0, /* EG2 velocity to attack: 0 time cents */ + 0, /* EG2 key number to decay: 0 time cents */ + 0, /* EG2 key number to hold: 0 time cents */ + + 0x7fff, /* Initial Fc: Disabled */ + 0, /* Initial Q: 0 dB */ + 0, /* Mod LFO to Fc: 0 cents */ + 0, /* Mod LFO CC1 to Fc: 0 cents */ + 0, /* Mod LFO channel pressure to Fc: 0 cents */ + 0, /* EG2 to Fc: 0 cents */ + 0, /* Velocity to Fc: 0 cents */ + 0, /* Key number to Fc: 0 cents */ + + 0, /* Mod LFO to gain: 0 dB */ + 0, /* Mod LFO CC1 to gain: 0 dB */ + 0, /* Mod LFO channel pressure to gain: 0 dB */ + 960, /* Velocity to gain: 96 dB */ + + 0, /* Tuning: 0 cents */ + 12800, /* Key number to pitch: 12,800 cents */ + 0, /* Vibrato to pitch: 0 cents */ + 0, /* Vibrato CC1 to pitch: 0 cents */ + 0, /* Vibrato channel pressure to pitch: 0 cents */ + 0, /* Mod LFO to pitch: 0 cents */ + 0, /* Mod LFO CC1 to pitch: 0 cents */ + 0, /* Mod LFO channel pressure to pitch: 0 cents */ + 0, /* Mod EG to pitch: 0 cents */ + + 0, /* Default pan: 0.0% */ + 0, /* Default reverb send: 0.0% */ + 1000, /* Default CC91 to reverb send: 100.0% */ + 0, /* Default chorus send: 0.0% */ + 1000 /* Default CC93 to chorus send: 100.0% */ +}; + +/*------------------------------------ + * local variables + *------------------------------------ +*/ + +#if defined(_8_BIT_SAMPLES) +static const EAS_INT bitDepth = 8; +#elif defined(_16_BIT_SAMPLES) +static const EAS_INT bitDepth = 16; +#else +#error "Must define _8_BIT_SAMPLES or _16_BIT_SAMPLES" +#endif + +static const EAS_U32 outputSampleRate = _OUTPUT_SAMPLE_RATE; +static const EAS_I32 dlsRateConvert = DLS_RATE_CONVERT; +static const EAS_I32 dlsLFOFrequencyConvert = DLS_LFO_FREQUENCY_CONVERT; + +/*------------------------------------ + * inline functions + *------------------------------------ +*/ +EAS_INLINE void *PtrOfs (void *p, EAS_I32 offset) +{ + return (void*) (((EAS_U8*) p) + offset); +} + +/*------------------------------------ + * prototypes + *------------------------------------ +*/ +static EAS_RESULT NextChunk (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 *pPos, EAS_U32 *pChunkType, EAS_I32 *pSize); +static EAS_RESULT Parse_ptbl (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 wsmpPos, EAS_I32 wsmpSize); +static EAS_RESULT Parse_wave (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U16 waveIndex); +static EAS_RESULT Parse_wsmp (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_WSMP_DATA *p); +static EAS_RESULT Parse_fmt (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_WSMP_DATA *p); +static EAS_RESULT Parse_data (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, S_WSMP_DATA *p, EAS_SAMPLE *pSample); +static EAS_RESULT Parse_lins(SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size); +static EAS_RESULT Parse_ins (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size); +static EAS_RESULT Parse_insh (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U32 *pRgnCount, EAS_U32 *pLocale); +static EAS_RESULT Parse_lrgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, EAS_U16 artIndex, EAS_U32 numRegions); +static EAS_RESULT Parse_rgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, EAS_U16 artIndex); +static EAS_RESULT Parse_rgnh (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_DLS_REGION *pRgn); +static EAS_RESULT Parse_lart (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, S_DLS_ART_VALUES *pArt); +static EAS_RESULT Parse_art (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_DLS_ART_VALUES *pArt); +static EAS_RESULT Parse_wlnk (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U32 *pWaveIndex); +static EAS_RESULT Parse_cdl (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 size, EAS_U32 *pValue); +static void Convert_rgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_U16 regionIndex, EAS_U16 artIndex, EAS_U16 waveIndex, S_WSMP_DATA *pWsmp); +static void Convert_art (SDLS_SYNTHESIZER_DATA *pDLSData, const S_DLS_ART_VALUES *pDLSArt, EAS_U16 artIndex); +static EAS_I16 ConvertSampleRate (EAS_U32 sampleRate); +static EAS_I16 ConvertSustain (EAS_I32 sustain); +static EAS_I16 ConvertLFOPhaseIncrement (EAS_I32 pitchCents); +static EAS_I8 ConvertPan (EAS_I32 pan); +static EAS_U8 ConvertQ (EAS_I32 q); + +#ifdef _DEBUG_DLS +static void DumpDLS (S_EAS *pEAS); +#endif + + +/*---------------------------------------------------------------------------- + * DLSParser () + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * pEASData - pointer to over EAS data instance + * fileHandle - file handle for input file + * offset - offset into file where DLS data starts + * + * Outputs: + * EAS_RESULT + * ppEAS - address of pointer to alternate EAS wavetable + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT DLSParser (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, EAS_DLSLIB_HANDLE *ppDLS) +{ + EAS_RESULT result; + SDLS_SYNTHESIZER_DATA dls; + EAS_U32 temp; + EAS_I32 pos; + EAS_I32 chunkPos; + EAS_I32 size; + EAS_I32 instSize; + EAS_I32 rgnPoolSize; + EAS_I32 artPoolSize; + EAS_I32 waveLenSize; + EAS_I32 endDLS; + EAS_I32 wvplPos; + EAS_I32 wvplSize; + EAS_I32 linsPos; + EAS_I32 linsSize; + EAS_I32 ptblPos; + EAS_I32 ptblSize; + void *p; + + /* zero counts and pointers */ + EAS_HWMemSet(&dls, 0, sizeof(dls)); + + /* save file handle and hwInstData to save copying pointers around */ + dls.hwInstData = hwInstData; + dls.fileHandle = fileHandle; + + /* NULL return value in case of error */ + *ppDLS = NULL; + + /* seek to start of DLS and read in RIFF tag and set processor endian flag */ + if ((result = EAS_HWFileSeek(dls.hwInstData, dls.fileHandle, offset)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWReadFile(dls.hwInstData, dls.fileHandle, &temp, sizeof(temp), &size)) != EAS_SUCCESS) + return result; + + /* check for processor endian-ness */ + dls.bigEndian = (temp == CHUNK_RIFF); + + /* first chunk should be DLS */ + pos = offset; + if ((result = NextChunk(&dls, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + if (temp != CHUNK_DLS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected DLS chunk, got %08lx\n", temp); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* no instrument or wavepool chunks */ + linsSize = wvplSize = ptblSize = linsPos = wvplPos = ptblPos = 0; + + /* scan the chunks in the DLS list */ + endDLS = offset + size; + pos = offset + 12; + while (pos < endDLS) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(&dls, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* parse useful chunks */ + switch (temp) + { + case CHUNK_CDL: + if ((result = Parse_cdl(&dls, size, &temp)) != EAS_SUCCESS) + return result; + if (!temp) + return EAS_ERROR_UNRECOGNIZED_FORMAT; + break; + + case CHUNK_LINS: + linsPos = chunkPos + 12; + linsSize = size - 4; + break; + + case CHUNK_WVPL: + wvplPos = chunkPos + 12; + wvplSize = size - 4; + break; + + case CHUNK_PTBL: + ptblPos = chunkPos + 8; + ptblSize = size - 4; + break; + + default: + break; + } + } + + /* must have a lins chunk */ + if (linsSize == 0) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No lins chunk found"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* must have a wvpl chunk */ + if (wvplSize == 0) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No wvpl chunk found"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* must have a ptbl chunk */ + if ((ptblSize == 0) || (ptblSize > DLS_MAX_WAVE_COUNT * sizeof(POOLCUE) + sizeof(POOLTABLE))) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No ptbl chunk found"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* pre-parse the wave pool chunk */ + if ((result = Parse_ptbl(&dls, ptblPos, wvplPos, wvplSize)) != EAS_SUCCESS) + return result; + + /* limit check */ + if ((dls.waveCount == 0) || (dls.waveCount > DLS_MAX_WAVE_COUNT)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS file contains invalid #waves [%u]\n", dls.waveCount); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* allocate memory for wsmp data */ + dls.wsmpData = EAS_HWMalloc(dls.hwInstData, (EAS_I32) (sizeof(S_WSMP_DATA) * dls.waveCount)); + if (dls.wsmpData == NULL) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_HWMalloc for wsmp data failed\n"); */ } + return EAS_ERROR_MALLOC_FAILED; + } + EAS_HWMemSet(dls.wsmpData, 0, (EAS_I32) (sizeof(S_WSMP_DATA) * dls.waveCount)); + + /* pre-parse the lins chunk */ + result = Parse_lins(&dls, linsPos, linsSize); + if (result == EAS_SUCCESS) + { + + /* limit check */ + if ((dls.regionCount == 0) || (dls.regionCount > DLS_MAX_REGION_COUNT)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS file contains invalid #regions [%u]\n", dls.regionCount); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* limit check */ + if ((dls.artCount == 0) || (dls.artCount > DLS_MAX_ART_COUNT)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS file contains invalid #articulations [%u]\n", dls.regionCount); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* limit check */ + if ((dls.instCount == 0) || (dls.instCount > DLS_MAX_INST_COUNT)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS file contains invalid #instruments [%u]\n", dls.instCount); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* Allocate memory for the converted DLS data */ + /* calculate size of instrument data */ + instSize = (EAS_I32) (sizeof(S_PROGRAM) * dls.instCount); + + /* calculate size of region pool */ + rgnPoolSize = (EAS_I32) (sizeof(S_DLS_REGION) * dls.regionCount); + + /* calculate size of articulation pool, add one for default articulation */ + dls.artCount++; + artPoolSize = (EAS_I32) (sizeof(S_DLS_ARTICULATION) * dls.artCount); + + /* calculate size of wave length and offset arrays */ + waveLenSize = (EAS_I32) (dls.waveCount * sizeof(EAS_U32)); + + /* calculate final memory size */ + size = (EAS_I32) sizeof(S_EAS) + instSize + rgnPoolSize + artPoolSize + (2 * waveLenSize) + (EAS_I32) dls.wavePoolSize; + if (size <= 0) { + return EAS_ERROR_FILE_FORMAT; + } + + /* allocate the main EAS chunk */ + dls.pDLS = EAS_HWMalloc(dls.hwInstData, size); + if (dls.pDLS == NULL) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_HWMalloc failed for DLS memory allocation size %ld\n", size); */ } + return EAS_ERROR_MALLOC_FAILED; + } + EAS_HWMemSet(dls.pDLS, 0, size); + dls.pDLS->refCount = 1; + p = PtrOfs(dls.pDLS, sizeof(S_EAS)); + + /* setup pointer to programs */ + dls.pDLS->numDLSPrograms = (EAS_U16) dls.instCount; + dls.pDLS->pDLSPrograms = p; + p = PtrOfs(p, instSize); + + /* setup pointer to regions */ + dls.pDLS->pDLSRegions = p; + dls.pDLS->numDLSRegions = (EAS_U16) dls.regionCount; + p = PtrOfs(p, rgnPoolSize); + + /* setup pointer to articulations */ + dls.pDLS->numDLSArticulations = (EAS_U16) dls.artCount; + dls.pDLS->pDLSArticulations = p; + p = PtrOfs(p, artPoolSize); + + /* setup pointer to wave length table */ + dls.pDLS->numDLSSamples = (EAS_U16) dls.waveCount; + dls.pDLS->pDLSSampleLen = p; + p = PtrOfs(p, waveLenSize); + + /* setup pointer to wave offsets table */ + dls.pDLS->pDLSSampleOffsets = p; + p = PtrOfs(p, waveLenSize); + + /* setup pointer to wave pool */ + dls.pDLS->pDLSSamples = p; + + /* clear filter flag */ + dls.filterUsed = EAS_FALSE; + + /* parse the wave pool and load samples */ + result = Parse_ptbl(&dls, ptblPos, wvplPos, wvplSize); + } + + /* create the default articulation */ + Convert_art(&dls, &defaultArt, 0); + dls.artCount = 1; + + /* parse the lins chunk and load instruments */ + dls.regionCount = dls.instCount = 0; + if (result == EAS_SUCCESS) + result = Parse_lins(&dls, linsPos, linsSize); + + /* clean up any temporary objects that were allocated */ + if (dls.wsmpData) + EAS_HWFree(dls.hwInstData, dls.wsmpData); + + /* if successful, return a pointer to the EAS collection */ + if (result == EAS_SUCCESS) + { + *ppDLS = dls.pDLS; +#ifdef _DEBUG_DLS + DumpDLS(dls.pDLS); +#endif + } + + /* something went wrong, deallocate the EAS collection */ + else + DLSCleanup(dls.hwInstData, dls.pDLS); + + return result; +} + +/*---------------------------------------------------------------------------- + * DLSCleanup () + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * pEASData - pointer to over EAS data instance + * pEAS - pointer to alternate EAS wavetable + * + * Outputs: + * EAS_RESULT + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT DLSCleanup (EAS_HW_DATA_HANDLE hwInstData, S_DLS *pDLS) +{ + + /* free the allocated memory */ + if (pDLS) + { + if (pDLS->refCount) + { + if (--pDLS->refCount == 0) + EAS_HWFree(hwInstData, pDLS); + } + } + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * DLSAddRef () + *---------------------------------------------------------------------------- + * Increment reference count + *---------------------------------------------------------------------------- +*/ +void DLSAddRef (S_DLS *pDLS) +{ + if (pDLS) + pDLS->refCount++; +} + +/*---------------------------------------------------------------------------- + * NextChunk () + *---------------------------------------------------------------------------- + * Purpose: + * Returns the type and size of the next chunk in the file + * + * Inputs: + * + * Outputs: + * + * Side Effects: + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT NextChunk (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 *pPos, EAS_U32 *pChunkType, EAS_I32 *pSize) +{ + EAS_RESULT result; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, *pPos)) != EAS_SUCCESS) + return result; + + /* read the chunk type */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, pChunkType, EAS_TRUE)) != EAS_SUCCESS) + return result; + + /* read the chunk size */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, pSize, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get form type for RIFF and LIST types */ + if ((*pChunkType == CHUNK_RIFF) || (*pChunkType == CHUNK_LIST)) + { + + /* read the form type */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, pChunkType, EAS_TRUE)) != EAS_SUCCESS) + return result; + + } + + /* calculate start of next chunk */ + *pPos += *pSize + 8; + + /* adjust to word boundary */ + if (*pPos & 1) + (*pPos)++; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_ptbl () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_ptbl (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 wtblPos, EAS_I32 wtblSize) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_FILE_HANDLE tempFile; + EAS_U16 waveIndex; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get the structure size */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &temp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get the number of waves */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &pDLSData->waveCount, EAS_FALSE)) != EAS_SUCCESS) + return result; + +#if 0 + /* just need the wave count on the first pass */ + if (!pDLSData->pDLS) + return EAS_SUCCESS; +#endif + + /* open duplicate file handle */ + if ((result = EAS_HWDupHandle(pDLSData->hwInstData, pDLSData->fileHandle, &tempFile)) != EAS_SUCCESS) + return result; + + /* read to end of chunk */ + for (waveIndex = 0; waveIndex < pDLSData->waveCount; waveIndex++) + { + + /* get the offset to the wave and make sure it is within the wtbl chunk */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, tempFile, &temp, EAS_FALSE)) != EAS_SUCCESS) + return result; + if (temp > (EAS_U32) wtblSize) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Ptbl offset exceeds size of wtbl\n"); */ } + EAS_HWCloseFile(pDLSData->hwInstData, tempFile); + return EAS_ERROR_FILE_FORMAT; + } + + /* parse the wave */ + if ((result = Parse_wave(pDLSData, wtblPos +(EAS_I32) temp, waveIndex)) != EAS_SUCCESS) + return result; + } + + /* close the temporary handle and return */ + EAS_HWCloseFile(pDLSData->hwInstData, tempFile); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_wave () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_wave (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U16 waveIndex) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 size; + EAS_I32 endChunk; + EAS_I32 chunkPos; + EAS_I32 wsmpPos = 0; + EAS_I32 fmtPos = 0; + EAS_I32 dataPos = 0; + EAS_I32 dataSize = 0; + S_WSMP_DATA *p; + void *pSample; + S_WSMP_DATA wsmp; + + /* seek to start of chunk */ + chunkPos = pos + 12; + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get the chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* make sure it is a wave chunk */ + if (temp != CHUNK_WAVE) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Offset in ptbl does not point to wave chunk\n"); */ } + return EAS_ERROR_FILE_FORMAT; + } + + /* read to end of chunk */ + pos = chunkPos; + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* parse useful chunks */ + switch (temp) + { + case CHUNK_WSMP: + wsmpPos = chunkPos + 8; + break; + + case CHUNK_FMT: + fmtPos = chunkPos + 8; + break; + + case CHUNK_DATA: + dataPos = chunkPos + 8; + dataSize = size; + break; + + default: + break; + } + } + + // limit to reasonable size + if (dataSize > MAX_DLS_WAVE_SIZE) + { + return EAS_ERROR_SOUND_LIBRARY; + } + + /* for first pass, use temporary variable */ + if (pDLSData->pDLS == NULL) + p = &wsmp; + else + p = &pDLSData->wsmpData[waveIndex]; + + /* set the defaults */ + p->fineTune = 0; + p->unityNote = 60; + p->gain = 0; + p->loopStart = 0; + p->loopLength = 0; + + /* must have a fmt chunk */ + if (!fmtPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS wave chunk has no fmt chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* must have a data chunk */ + if (!dataPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS wave chunk has no data chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* parse the wsmp chunk */ + if (wsmpPos) + { + if ((result = Parse_wsmp(pDLSData, wsmpPos, p)) != EAS_SUCCESS) + return result; + } + + /* parse the fmt chunk */ + if ((result = Parse_fmt(pDLSData, fmtPos, p)) != EAS_SUCCESS) + return result; + + /* calculate the size of the wavetable needed. We need only half + * the memory for 16-bit samples when in 8-bit mode, and we need + * double the memory for 8-bit samples in 16-bit mode. For + * unlooped samples, we may use ADPCM. If so, we need only 1/4 + * the memory. + * + * We also need to add one for looped samples to allow for + * the first sample to be copied to the end of the loop. + */ + + /* use ADPCM encode for unlooped 16-bit samples if ADPCM is enabled */ + /*lint -e{506} -e{774} groundwork for future version to support 8 & 16 bit */ + if (bitDepth == 8) + { + if (p->bitsPerSample == 8) + size = dataSize; + else + /*lint -e{704} use shift for performance */ + size = dataSize >> 1; + if (p->loopLength) + size++; + } + + else + { + if (p->bitsPerSample == 16) + size = dataSize; + else + /*lint -e{703} use shift for performance */ + size = dataSize << 1; + if (p->loopLength) + size += 2; + } + + /* for first pass, add size to wave pool size and return */ + if (pDLSData->pDLS == NULL) + { + pDLSData->wavePoolSize += (EAS_U32) size; + return EAS_SUCCESS; + } + + /* allocate memory and read in the sample data */ + pSample = pDLSData->pDLS->pDLSSamples + pDLSData->wavePoolOffset; + pDLSData->pDLS->pDLSSampleOffsets[waveIndex] = pDLSData->wavePoolOffset; + pDLSData->pDLS->pDLSSampleLen[waveIndex] = (EAS_U32) size; + pDLSData->wavePoolOffset += (EAS_U32) size; + if (pDLSData->wavePoolOffset > pDLSData->wavePoolSize) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Wave pool exceeded allocation\n"); */ } + return EAS_ERROR_SOUND_LIBRARY; + } + + if ((result = Parse_data(pDLSData, dataPos, dataSize, p, pSample)) != EAS_SUCCESS) + return result; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_wsmp () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_wsmp (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_WSMP_DATA *p) +{ + EAS_RESULT result; + EAS_U16 wtemp; + EAS_U32 ltemp; + EAS_U32 cbSize; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get structure size */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &cbSize, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get unity note */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &wtemp, EAS_FALSE)) != EAS_SUCCESS) + return result; + if (wtemp <= 127) + p->unityNote = (EAS_U8) wtemp; + else + { + p->unityNote = 60; + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Invalid unity note [%u] in DLS wsmp ignored, set to 60\n", wtemp); */ } + } + + /* get fine tune */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->fineTune, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get gain */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->gain, EAS_FALSE)) != EAS_SUCCESS) + return result; + if (p->gain > 0) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Positive gain [%ld] in DLS wsmp ignored, set to 0dB\n", p->gain); */ } + p->gain = 0; + } + + /* option flags */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, <emp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* sample loops */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, <emp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* if looped sample, get loop data */ + if (ltemp) + { + + if (ltemp > 1) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS sample with %lu loops, ignoring extra loops\n", ltemp); */ } + + /* skip ahead to loop data */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos + (EAS_I32) cbSize)) != EAS_SUCCESS) + return result; + + /* get structure size */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, <emp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get loop type */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, <emp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get loop start */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->loopStart, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get loop length */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->loopLength, EAS_FALSE)) != EAS_SUCCESS) + return result; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_fmt () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_fmt (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_WSMP_DATA *p) +{ + EAS_RESULT result; + EAS_U16 wtemp; + EAS_U32 ltemp; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get format tag */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &wtemp, EAS_FALSE)) != EAS_SUCCESS) + return result; + if (wtemp != WAVE_FORMAT_PCM) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unsupported DLS sample format %04x\n", wtemp); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* get number of channels */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &wtemp, EAS_FALSE)) != EAS_SUCCESS) + return result; + if (wtemp != 1) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No support for DLS multi-channel samples\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* get sample rate */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->sampleRate, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* bytes/sec */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, <emp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* block align */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &wtemp, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* bits/sample */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &p->bitsPerSample, EAS_FALSE)) != EAS_SUCCESS) + return result; + + if ((p->bitsPerSample != 8) && (p->bitsPerSample != 16)) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unsupported DLS bits-per-sample %d\n", p->bitsPerSample); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + return EAS_SUCCESS; +} + +#if defined( _8_BIT_SAMPLES) +/*---------------------------------------------------------------------------- + * Parse_data () + *---------------------------------------------------------------------------- + * Purpose: + * + * NOTE: The optimized assembly versions of the interpolator require + * an extra sample at the end of the loop - a copy of the first + * sample. This routine must allocate an extra sample of data and + * copy the first sample of the loop to the end. + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_data (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, S_WSMP_DATA *pWsmp, EAS_SAMPLE *pSample) +{ + EAS_RESULT result; + EAS_U8 convBuf[SAMPLE_CONVERT_CHUNK_SIZE]; + EAS_I32 count; + EAS_I32 i; + EAS_I8 *p; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* 8-bit samples in an 8-bit synth, just copy the data, and flip bit 7 */ + p = pSample; + if (pWsmp->bitsPerSample == 8) + { + if ((result = EAS_HWReadFile(pDLSData->hwInstData, pDLSData->fileHandle, pSample, size, &count)) != EAS_SUCCESS) + return result; + for (i = 0; i < size; i++) + /*lint -e{734} convert from unsigned to signed audio */ + *p++ ^= 0x80; + } + + /* 16-bit samples, need to convert to 8-bit or ADPCM */ + else + { + + while (size) + { + EAS_I8 *pInput; + + /* for undithered conversion, we're just copying the 8-bit data */ + if (pDLSData->bigEndian) + pInput = (EAS_I8*) convBuf; + else + pInput = (EAS_I8*) convBuf + 1; + + /* read a small chunk of data and convert it */ + count = (size < SAMPLE_CONVERT_CHUNK_SIZE ? size : SAMPLE_CONVERT_CHUNK_SIZE); + if ((result = EAS_HWReadFile(pDLSData->hwInstData, pDLSData->fileHandle, convBuf, count, &count)) != EAS_SUCCESS) + return result; + size -= count; + /*lint -e{704} use shift for performance */ + count = count >> 1; + + while (count--) + { + *p++ = *pInput; + pInput += 2; + } + } + } + + /* for looped samples, copy the last sample to the end */ + if (pWsmp->loopLength) + pSample[pWsmp->loopStart + pWsmp->loopLength] = pSample[pWsmp->loopStart]; + + return EAS_SUCCESS; +} +#elif defined(_16_BIT_SAMPLES) +#error "16-bit DLS conversion not implemented yet" +#else +#error "Must specifiy _8_BIT_SAMPLES or _16_BIT_SAMPLES" +#endif + +/*---------------------------------------------------------------------------- + * Parse_lins () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_lins (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 endChunk; + EAS_I32 chunkPos; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* read to end of chunk */ + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* only instrument chunks are useful */ + if (temp != CHUNK_INS) + continue; + + if ((result = Parse_ins(pDLSData, chunkPos + 12, size)) != EAS_SUCCESS) + return result; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_ins () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_ins (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 chunkPos; + EAS_I32 endChunk; + EAS_I32 lrgnPos; + EAS_I32 lrgnSize; + EAS_I32 lartPos; + EAS_I32 lartSize; + EAS_I32 lar2Pos; + EAS_I32 lar2Size; + EAS_I32 inshPos; + EAS_U32 regionCount; + EAS_U32 locale; + S_DLS_ART_VALUES art; + S_PROGRAM *pProgram; + EAS_U16 artIndex; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* no chunks yet */ + lrgnPos = lrgnSize = lartPos = lartSize = lar2Pos = lar2Size = inshPos = artIndex = 0; + + /* read to end of chunk */ + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* parse useful chunks */ + switch (temp) + { + case CHUNK_INSH: + inshPos = chunkPos + 8; + break; + + case CHUNK_LART: + lartPos = chunkPos + 12; + lartSize = size; + break; + + case CHUNK_LAR2: + lar2Pos = chunkPos + 12; + lar2Size = size; + break; + + case CHUNK_LRGN: + lrgnPos = chunkPos + 12; + lrgnSize = size; + break; + + default: + break; + } + } + + /* must have an lrgn to be useful */ + if (!lrgnPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS ins chunk has no lrgn chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* must have an insh to be useful */ + if (!inshPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS ins chunk has no insh chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* parse the instrument header */ + if ((result = Parse_insh(pDLSData, inshPos, ®ionCount, &locale)) != EAS_SUCCESS) + return result; + + /* initialize and parse the global data first */ + EAS_HWMemCpy(&art, &defaultArt, sizeof(S_DLS_ART_VALUES)); + if (lartPos) + if ((result = Parse_lart(pDLSData, lartPos, lartSize, &art)) != EAS_SUCCESS) + return result; + if (lar2Pos) + if ((result = Parse_lart(pDLSData, lar2Pos, lar2Size, &art)) != EAS_SUCCESS) + return result; + + if (art.values[PARAM_MODIFIED]) + { + artIndex = (EAS_U16) pDLSData->artCount; + pDLSData->artCount++; + } + + /* convert data on second pass */ + if (pDLSData->pDLS) + { + + if (art.values[PARAM_MODIFIED]) + Convert_art(pDLSData, &art, artIndex); + + /* setup pointers */ + pProgram = &pDLSData->pDLS->pDLSPrograms[pDLSData->instCount]; + + /* initialize instrument */ + pProgram->locale = locale; + pProgram->regionIndex = (EAS_U16) pDLSData->regionCount | FLAG_RGN_IDX_DLS_SYNTH; + + } + + /* parse the region data */ + if ((result = Parse_lrgn(pDLSData, lrgnPos, lrgnSize, artIndex, regionCount)) != EAS_SUCCESS) + return result; + + /* bump instrument count */ + pDLSData->instCount++; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_insh () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_insh (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U32 *pRgnCount, EAS_U32 *pLocale) +{ + EAS_RESULT result; + EAS_U32 bank; + EAS_U32 program; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get the region count and locale */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, pRgnCount, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &bank, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &program, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* verify the parameters are valid */ + if (bank & 0x7fff8080) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS bank number is out of range: %08lx\n", bank); */ } + bank &= 0xff7f; + } + if (program > 127) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS program number is out of range: %08lx\n", program); */ } + program &= 0x7f; + } + + /* save the program number */ + *pLocale = (bank << 8) | program; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_lrgn () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_lrgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, EAS_U16 artIndex, EAS_U32 numRegions) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 chunkPos; + EAS_I32 endChunk; + EAS_U16 regionCount; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* read to end of chunk */ + regionCount = 0; + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + if ((temp == CHUNK_RGN) || (temp == CHUNK_RGN2)) + { + if (regionCount == numRegions) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS region count exceeded cRegions value in insh, extra region ignored\n"); */ } + return EAS_SUCCESS; + } + if ((result = Parse_rgn(pDLSData, chunkPos + 12, size, artIndex)) != EAS_SUCCESS) + return result; + regionCount++; + } + } + + /* set a flag in the last region */ + if ((pDLSData->pDLS != NULL) && (regionCount > 0)) + pDLSData->pDLS->pDLSRegions[pDLSData->regionCount - 1].wtRegion.region.keyGroupAndFlags |= REGION_FLAG_LAST_REGION; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_rgn () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_rgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, EAS_U16 artIndex) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 chunkPos; + EAS_I32 endChunk; + EAS_I32 rgnhPos; + EAS_I32 lartPos; + EAS_I32 lartSize; + EAS_I32 lar2Pos; + EAS_I32 lar2Size; + EAS_I32 wlnkPos; + EAS_I32 wsmpPos; + EAS_U32 waveIndex; + S_DLS_ART_VALUES art; + S_WSMP_DATA wsmp; + S_WSMP_DATA *pWsmp; + EAS_U16 regionIndex; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* no chunks found yet */ + rgnhPos = lartPos = lartSize = lar2Pos = lar2Size = wsmpPos = wlnkPos = 0; + regionIndex = (EAS_U16) pDLSData->regionCount; + + /* read to end of chunk */ + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* parse useful chunks */ + switch (temp) + { + case CHUNK_CDL: + if ((result = Parse_cdl(pDLSData, size, &temp)) != EAS_SUCCESS) + return result; + + /* if conditional chunk evaluates false, skip this list */ + if (!temp) + return EAS_SUCCESS; + break; + + case CHUNK_RGNH: + rgnhPos = chunkPos + 8; + break; + + case CHUNK_WLNK: + wlnkPos = chunkPos + 8; + break; + + case CHUNK_WSMP: + wsmpPos = chunkPos + 8; + break; + + case CHUNK_LART: + lartPos = chunkPos + 12; + lartSize = size; + break; + + case CHUNK_LAR2: + lar2Pos = chunkPos + 12; + lar2Size = size; + break; + + default: + break; + } + } + + /* must have a rgnh chunk to be useful */ + if (!rgnhPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS rgn chunk has no rgnh chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* must have a wlnk chunk to be useful */ + if (!wlnkPos) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "DLS rgn chunk has no wlnk chunk\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; + } + + /* parse wlnk chunk */ + if ((result = Parse_wlnk(pDLSData, wlnkPos, &waveIndex)) != EAS_SUCCESS) + return result; + pWsmp = &pDLSData->wsmpData[waveIndex]; + + /* if there is any articulation data, parse it */ + EAS_HWMemCpy(&art, &defaultArt, sizeof(S_DLS_ART_VALUES)); + if (lartPos) + { + if ((result = Parse_lart(pDLSData, lartPos, lartSize, &art)) != EAS_SUCCESS) + return result; + } + + if (lar2Pos) + { + if ((result = Parse_lart(pDLSData, lar2Pos, lar2Size, &art)) != EAS_SUCCESS) + return result; + } + + /* if second pass, process region header */ + if (pDLSData->pDLS) + { + + /* if local data was found convert it */ + if (art.values[PARAM_MODIFIED] == EAS_TRUE) + { + Convert_art(pDLSData, &art, (EAS_U16) pDLSData->artCount); + artIndex = (EAS_U16) pDLSData->artCount; + } + + /* parse region header */ + if ((result = Parse_rgnh(pDLSData, rgnhPos, &pDLSData->pDLS->pDLSRegions[regionIndex & REGION_INDEX_MASK])) != EAS_SUCCESS) + return result; + + /* parse wsmp chunk, copying parameters from original first */ + if (wsmpPos) + { + EAS_HWMemCpy(&wsmp, pWsmp, sizeof(wsmp)); + if ((result = Parse_wsmp(pDLSData, wsmpPos, &wsmp)) != EAS_SUCCESS) + return result; + + pWsmp = &wsmp; + } + + Convert_rgn(pDLSData, regionIndex, artIndex, (EAS_U16) waveIndex, pWsmp); + } + + /* if local articulation, bump count */ + if (art.values[PARAM_MODIFIED]) + pDLSData->artCount++; + + /* increment region count */ + pDLSData->regionCount++; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_rgnh () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_rgnh (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_DLS_REGION *pRgn) +{ + EAS_RESULT result; + EAS_U16 lowKey; + EAS_U16 highKey; + EAS_U16 lowVel; + EAS_U16 highVel; + EAS_U16 optionFlags; + EAS_U16 keyGroup; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get the key range */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &lowKey, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &highKey, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* check the range */ + if (lowKey > 127) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS rgnh: Low key out of range [%u]\n", lowKey); */ } + lowKey = 127; + } + if (highKey > 127) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS rgnh: High key out of range [%u]\n", lowKey); */ } + highKey = 127; + } + + /* get the velocity range */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &lowVel, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &highVel, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* check the range */ + if (lowVel > 127) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS rgnh: Low velocity out of range [%u]\n", lowVel); */ } + lowVel = 127; + } + if (highVel > 127) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS rgnh: High velocity out of range [%u]\n", highVel); */ } + highVel = 127; + } + + /* get the option flags */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &optionFlags, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* get the key group */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &keyGroup, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* save the key range and key group */ + pRgn->wtRegion.region.rangeLow = (EAS_U8) lowKey; + pRgn->wtRegion.region.rangeHigh = (EAS_U8) highKey; + + /*lint -e{734} keyGroup will always be from 0-15 */ + pRgn->wtRegion.region.keyGroupAndFlags = keyGroup << 8; + pRgn->velLow = (EAS_U8) lowVel; + pRgn->velHigh = (EAS_U8) highVel; + if (optionFlags & F_RGN_OPTION_SELFNONEXCLUSIVE) + pRgn->wtRegion.region.keyGroupAndFlags |= REGION_FLAG_NON_SELF_EXCLUSIVE; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_lart () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_lart (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_I32 size, S_DLS_ART_VALUES *pArt) +{ + EAS_RESULT result; + EAS_U32 temp; + EAS_I32 endChunk; + EAS_I32 chunkPos; + EAS_I32 art1Pos; + EAS_I32 art2Pos; + + /* seek to start of chunk */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* no articulation chunks yet */ + art1Pos = art2Pos = 0; + + /* read to end of chunk */ + endChunk = pos + size; + while (pos < endChunk) + { + chunkPos = pos; + + /* get the next chunk type */ + if ((result = NextChunk(pDLSData, &pos, &temp, &size)) != EAS_SUCCESS) + return result; + + /* parse useful chunks */ + switch (temp) + { + case CHUNK_CDL: + if ((result = Parse_cdl(pDLSData, size, &temp)) != EAS_SUCCESS) + return result; + + /* if conditional chunk evaluates false, skip this list */ + if (!temp) + return EAS_SUCCESS; + break; + + case CHUNK_ART1: + art1Pos = chunkPos + 8; + break; + + case CHUNK_ART2: + art2Pos = chunkPos + 8; + break; + + default: + break; + + } + } + + if (art1Pos) + { + if ((result = Parse_art(pDLSData, art1Pos, pArt)) != EAS_SUCCESS) + return result; + } + + if (art2Pos) + { + if ((result = Parse_art(pDLSData, art2Pos, pArt)) != EAS_SUCCESS) + return result; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_art() + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_art (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, S_DLS_ART_VALUES *pArt) +{ + EAS_RESULT result; + EAS_U32 structSize; + EAS_U32 numConnections; + EAS_U16 source; + EAS_U16 control; + EAS_U16 destination; + EAS_U16 transform; + EAS_I32 scale; + EAS_INT i; + + /* seek to start of data */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + /* get the structure size */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &structSize, EAS_FALSE)) != EAS_SUCCESS) + return result; + pos += (EAS_I32) structSize; + + /* get the number of connections */ + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &numConnections, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* skip to start of connections */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos)) != EAS_SUCCESS) + return result; + + while (numConnections--) + { + + /* read the connection data */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &source, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &control, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &destination, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &transform, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &scale, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* look up the connection */ + for (i = 0; i < (EAS_INT) ENTRIES_IN_CONN_TABLE; i++) + { + if ((connTable[i].source == source) && + (connTable[i].destination == destination) && + (connTable[i].control == control)) + { + /*lint -e{704} use shift for performance */ + pArt->values[connTable[i].connection] = (EAS_I16) (scale >> 16); + pArt->values[PARAM_MODIFIED] = EAS_TRUE; + break; + } + } + if (i == PARAM_TABLE_SIZE) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "WARN: Unsupported parameter in DLS file\n"); */ } + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * Parse_wlnk () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_wlnk (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 pos, EAS_U32 *pWaveIndex) +{ + EAS_RESULT result; + + /* we only care about the the index */ + if ((result = EAS_HWFileSeek(pDLSData->hwInstData, pDLSData->fileHandle, pos + 8)) != EAS_SUCCESS) + return result; + + /* read the index */ + return EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle,pWaveIndex, EAS_FALSE); +} + +/*---------------------------------------------------------------------------- + * PopcdlStack () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT PopcdlStack (EAS_U32 *pStack, EAS_INT *pStackPtr, EAS_U32 *pValue) +{ + + /* stack underflow, cdl block has an errorr */ + if (*pStackPtr < 0) + return EAS_ERROR_FILE_FORMAT; + + /* pop the value off the stack */ + *pValue = pStack[*pStackPtr]; + *pStackPtr = *pStackPtr - 1; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * PushcdlStack () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT PushcdlStack (EAS_U32 *pStack, EAS_INT *pStackPtr, EAS_U32 value) +{ + + /* stack overflow, return an error */ + if (*pStackPtr >= CDL_STACK_SIZE) + return EAS_ERROR_FILE_FORMAT; + + /* push the value onto the stack */ + *pStackPtr = *pStackPtr + 1; + pStack[*pStackPtr] = value; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * QueryGUID () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_BOOL QueryGUID (const DLSID *pGUID, EAS_U32 *pValue) +{ + + /* assume false */ + *pValue = 0; + if (EAS_HWMemCmp(&DLSID_GMInHardware, pGUID, sizeof(DLSID)) == 0) + { + *pValue = 0xffffffff; + return EAS_TRUE; + } + + if (EAS_HWMemCmp(&DLSID_GSInHardware, pGUID, sizeof(DLSID)) == 0) + return EAS_TRUE; + + if (EAS_HWMemCmp(&DLSID_XGInHardware, pGUID, sizeof(DLSID)) == 0) + return EAS_TRUE; + + if (EAS_HWMemCmp(&DLSID_SupportsDLS1, pGUID, sizeof(DLSID)) == 0) + { + *pValue = 0xffffffff; + return EAS_TRUE; + } + + if (EAS_HWMemCmp(&DLSID_SupportsDLS2, pGUID, sizeof(DLSID)) == 0) + return EAS_TRUE; + + if (EAS_HWMemCmp(&DLSID_SampleMemorySize, pGUID, sizeof(DLSID)) == 0) + { + *pValue = MAX_DLS_MEMORY; + return EAS_TRUE; + } + + if (EAS_HWMemCmp(&DLSID_ManufacturersID, pGUID, sizeof(DLSID)) == 0) + { + *pValue = 0x0000013A; + return EAS_TRUE; + } + + if (EAS_HWMemCmp(&DLSID_ProductID, pGUID, sizeof(DLSID)) == 0) + { + *pValue = LIB_VERSION; + return EAS_TRUE; + } + + if (EAS_HWMemCmp(&DLSID_SamplePlaybackRate, pGUID, sizeof(DLSID)) == 0) + { + *pValue = (EAS_U32) outputSampleRate; + return EAS_TRUE; + } + + /* unrecognized DLSID */ + return EAS_FALSE; +} + +/*---------------------------------------------------------------------------- + * ReadDLSID () + *---------------------------------------------------------------------------- + * Purpose: + * Reads a DLSID in a manner that is not sensitive to processor endian-ness + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT ReadDLSID (SDLS_SYNTHESIZER_DATA *pDLSData, DLSID *pDLSID) +{ + EAS_RESULT result; + EAS_I32 n; + + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &pDLSID->Data1, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &pDLSID->Data2, EAS_FALSE)) != EAS_SUCCESS) + return result; + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &pDLSID->Data3, EAS_FALSE)) != EAS_SUCCESS) + return result; + return EAS_HWReadFile(pDLSData->hwInstData, pDLSData->fileHandle, pDLSID->Data4, sizeof(pDLSID->Data4), &n); +} + +/*---------------------------------------------------------------------------- + * Parse_cdl () + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT Parse_cdl (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_I32 size, EAS_U32 *pValue) +{ + EAS_RESULT result; + EAS_U32 stack[CDL_STACK_SIZE]; + EAS_U16 opcode; + EAS_INT stackPtr; + EAS_U32 x, y; + DLSID dlsid; + + stackPtr = -1; + *pValue = 0; + x = 0; + while (size) + { + /* read the opcode */ + if ((result = EAS_HWGetWord(pDLSData->hwInstData, pDLSData->fileHandle, &opcode, EAS_FALSE)) != EAS_SUCCESS) + return result; + + /* handle binary opcodes */ + if (opcode <= DLS_CDL_EQ) + { + /* pop X and Y */ + if ((result = PopcdlStack(stack, &stackPtr, &x)) != EAS_SUCCESS) + return result; + if ((result = PopcdlStack(stack, &stackPtr, &y)) != EAS_SUCCESS) + return result; + switch (opcode) + { + case DLS_CDL_AND: + x = x & y; + break; + case DLS_CDL_OR: + x = x | y; + break; + case DLS_CDL_XOR: + x = x ^ y; + break; + case DLS_CDL_ADD: + x = x + y; + break; + case DLS_CDL_SUBTRACT: + x = x - y; + break; + case DLS_CDL_MULTIPLY: + x = x * y; + break; + case DLS_CDL_DIVIDE: + if (!y) + return EAS_ERROR_FILE_FORMAT; + x = x / y; + break; + case DLS_CDL_LOGICAL_AND: + x = (x && y); + break; + case DLS_CDL_LOGICAL_OR: + x = (x || y); + break; + case DLS_CDL_LT: + x = (x < y); + break; + case DLS_CDL_LE: + x = (x <= y); + break; + case DLS_CDL_GT: + x = (x > y); + break; + case DLS_CDL_GE: + x = (x >= y); + break; + case DLS_CDL_EQ: + x = (x == y); + break; + default: + break; + } + } + + else if (opcode == DLS_CDL_NOT) + { + if ((result = PopcdlStack(stack, &stackPtr, &x)) != EAS_SUCCESS) + return result; + x = !x; + } + + else if (opcode == DLS_CDL_CONST) + { + if ((result = EAS_HWGetDWord(pDLSData->hwInstData, pDLSData->fileHandle, &x, EAS_FALSE)) != EAS_SUCCESS) + return result; + } + + else if (opcode == DLS_CDL_QUERY) + { + if ((result = ReadDLSID(pDLSData, &dlsid)) != EAS_SUCCESS) + return result; + QueryGUID(&dlsid, &x); + } + + else if (opcode == DLS_CDL_QUERYSUPPORTED) + { + if ((result = ReadDLSID(pDLSData, &dlsid)) != EAS_SUCCESS) + return result; + x = QueryGUID(&dlsid, &y); + } + else + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unsupported opcode %d in DLS file\n", opcode); */ } + + /* push the result on the stack */ + if ((result = PushcdlStack(stack, &stackPtr, x)) != EAS_SUCCESS) + return result; + } + + /* pop the last result off the stack */ + return PopcdlStack(stack, &stackPtr, pValue); +} + +/*---------------------------------------------------------------------------- + * Convert_rgn() + *---------------------------------------------------------------------------- + * Purpose: + * Convert region data from DLS to EAS + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static void Convert_rgn (SDLS_SYNTHESIZER_DATA *pDLSData, EAS_U16 regionIndex, EAS_U16 artIndex, EAS_U16 waveIndex, S_WSMP_DATA *pWsmp) +{ + S_DLS_REGION *pRgn; + + /* setup pointers to data structures */ + pRgn = &pDLSData->pDLS->pDLSRegions[regionIndex]; + + /* intiailize indices */ + pRgn->wtRegion.artIndex = artIndex; + pRgn->wtRegion.waveIndex = waveIndex; + + /* convert region data */ + /*lint -e{704} use shift for performance */ + pRgn->wtRegion.gain = (EAS_I16) (pWsmp->gain >> 16); + pRgn->wtRegion.loopStart = pWsmp->loopStart; + pRgn->wtRegion.loopEnd = (pWsmp->loopStart + pWsmp->loopLength); + pRgn->wtRegion.tuning = pWsmp->fineTune -(pWsmp->unityNote * 100) + ConvertSampleRate(pWsmp->sampleRate); + if (pWsmp->loopLength != 0) + pRgn->wtRegion.region.keyGroupAndFlags |= REGION_FLAG_IS_LOOPED; +} + +/*---------------------------------------------------------------------------- + * Convert_art() + *---------------------------------------------------------------------------- + * Purpose: + * Convert articulation data from DLS to EAS + * + * Inputs: + * + * + * Outputs: + * + * + *---------------------------------------------------------------------------- +*/ +static void Convert_art (SDLS_SYNTHESIZER_DATA *pDLSData, const S_DLS_ART_VALUES *pDLSArt, EAS_U16 artIndex) +{ + S_DLS_ARTICULATION *pArt; + + /* setup pointers to data structures */ + pArt = &pDLSData->pDLS->pDLSArticulations[artIndex]; + + /* LFO parameters */ + pArt->modLFO.lfoFreq = ConvertLFOPhaseIncrement(pDLSArt->values[PARAM_MOD_LFO_FREQ]); + pArt->modLFO.lfoDelay = -ConvertDelay(pDLSArt->values[PARAM_MOD_LFO_DELAY]); + pArt->vibLFO.lfoFreq = ConvertLFOPhaseIncrement(pDLSArt->values[PARAM_VIB_LFO_FREQ]); + pArt->vibLFO.lfoDelay = -ConvertDelay(pDLSArt->values[PARAM_VIB_LFO_DELAY]); + + /* EG1 parameters */ + pArt->eg1.delayTime = ConvertDelay(pDLSArt->values[PARAM_VOL_EG_DELAY]); + pArt->eg1.attackTime = pDLSArt->values[PARAM_VOL_EG_ATTACK]; + pArt->eg1.holdTime = pDLSArt->values[PARAM_VOL_EG_HOLD]; + pArt->eg1.decayTime = pDLSArt->values[PARAM_VOL_EG_DECAY]; + pArt->eg1.sustainLevel = ConvertSustain(pDLSArt->values[PARAM_VOL_EG_SUSTAIN]); + pArt->eg1.releaseTime = ConvertRate(pDLSArt->values[PARAM_VOL_EG_RELEASE]); + pArt->eg1.velToAttack = pDLSArt->values[PARAM_VOL_EG_VEL_TO_ATTACK]; + pArt->eg1.keyNumToDecay = pDLSArt->values[PARAM_VOL_EG_KEY_TO_DECAY]; + pArt->eg1.keyNumToHold = pDLSArt->values[PARAM_VOL_EG_KEY_TO_HOLD]; + pArt->eg1ShutdownTime = ConvertRate(pDLSArt->values[PARAM_VOL_EG_SHUTDOWN]); + + /* EG2 parameters */ + pArt->eg2.delayTime = ConvertDelay(pDLSArt->values[PARAM_MOD_EG_DELAY]); + pArt->eg2.attackTime = pDLSArt->values[PARAM_MOD_EG_ATTACK]; + pArt->eg2.holdTime = pDLSArt->values[PARAM_MOD_EG_HOLD]; + pArt->eg2.decayTime = pDLSArt->values[PARAM_MOD_EG_DECAY]; + pArt->eg2.sustainLevel = ConvertSustain(pDLSArt->values[PARAM_MOD_EG_SUSTAIN]); + pArt->eg2.releaseTime = ConvertRate(pDLSArt->values[PARAM_MOD_EG_RELEASE]); + pArt->eg2.velToAttack = pDLSArt->values[PARAM_MOD_EG_VEL_TO_ATTACK]; + pArt->eg2.keyNumToDecay = pDLSArt->values[PARAM_MOD_EG_KEY_TO_DECAY]; + pArt->eg2.keyNumToHold = pDLSArt->values[PARAM_MOD_EG_KEY_TO_HOLD]; + + /* filter parameters */ + pArt->filterCutoff = pDLSArt->values[PARAM_INITIAL_FC]; + pArt->filterQandFlags = ConvertQ(pDLSArt->values[PARAM_INITIAL_Q]); + pArt->modLFOToFc = pDLSArt->values[PARAM_MOD_LFO_TO_FC]; + pArt->modLFOCC1ToFc = pDLSArt->values[PARAM_MOD_LFO_CC1_TO_FC]; + pArt->modLFOChanPressToFc = pDLSArt->values[PARAM_MOD_LFO_CHAN_PRESS_TO_FC]; + pArt->eg2ToFc = pDLSArt->values[PARAM_MOD_EG_TO_FC]; + pArt->velToFc = pDLSArt->values[PARAM_VEL_TO_FC]; + pArt->keyNumToFc = pDLSArt->values[PARAM_KEYNUM_TO_FC]; + + /* gain parameters */ + pArt->modLFOToGain = pDLSArt->values[PARAM_MOD_LFO_TO_GAIN]; + pArt->modLFOCC1ToGain = pDLSArt->values[PARAM_MOD_LFO_CC1_TO_GAIN]; + pArt->modLFOChanPressToGain = pDLSArt->values[PARAM_MOD_LFO_CHAN_PRESS_TO_GAIN]; + + /* pitch parameters */ + pArt->tuning = pDLSArt->values[PARAM_TUNING]; + pArt->keyNumToPitch = pDLSArt->values[PARAM_KEYNUM_TO_PITCH]; + pArt->vibLFOToPitch = pDLSArt->values[PARAM_VIB_LFO_TO_PITCH]; + pArt->vibLFOCC1ToPitch = pDLSArt->values[PARAM_VIB_LFO_CC1_TO_PITCH]; + pArt->vibLFOChanPressToPitch = pDLSArt->values[PARAM_VIB_LFO_CHAN_PRESS_TO_PITCH]; + pArt->modLFOToPitch = pDLSArt->values[PARAM_MOD_LFO_TO_PITCH]; + pArt->modLFOCC1ToPitch = pDLSArt->values[PARAM_MOD_LFO_CC1_TO_PITCH]; + pArt->modLFOChanPressToPitch = pDLSArt->values[PARAM_MOD_LFO_CHAN_PRESS_TO_PITCH]; + pArt->eg2ToPitch = pDLSArt->values[PARAM_MOD_EG_TO_PITCH]; + + /* output parameters */ + pArt->pan = ConvertPan(pDLSArt->values[PARAM_DEFAULT_PAN]); + + if (pDLSArt->values[PARAM_VEL_TO_GAIN] != 0) + pArt->filterQandFlags |= FLAG_DLS_VELOCITY_SENSITIVE; + +#ifdef _REVERB + pArt->reverbSend = pDLSArt->values[PARAM_DEFAULT_REVERB_SEND]; + pArt->cc91ToReverbSend = pDLSArt->values[PARAM_MIDI_CC91_TO_REVERB_SEND]; +#endif + +#ifdef _CHORUS + pArt->chorusSend = pDLSArt->values[PARAM_DEFAULT_CHORUS_SEND]; + pArt->cc93ToChorusSend = pDLSArt->values[PARAM_MIDI_CC93_TO_CHORUS_SEND]; +#endif +} + +/*---------------------------------------------------------------------------- + * ConvertSampleRate() + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * + * Outputs: + * + * Side Effects: + *---------------------------------------------------------------------------- +*/ +static EAS_I16 ConvertSampleRate (EAS_U32 sampleRate) +{ + return (EAS_I16) (1200.0 * log10((double) sampleRate / (double) outputSampleRate) / log10(2.0)); +} + +/*---------------------------------------------------------------------------- + * ConvertSustainEG2() + *---------------------------------------------------------------------------- + * Convert sustain level to pitch/Fc multipler for EG2 + *---------------------------------------------------------------------------- +*/ +static EAS_I16 ConvertSustain (EAS_I32 sustain) +{ + /* check for sustain level of zero */ + if (sustain == 0) + return 0; + + /* convert to log2 factor */ + /*lint -e{704} use shift for performance */ + sustain = (sustain * SUSTAIN_LINEAR_CONVERSION_FACTOR) >> 15; + + if (sustain > SYNTH_FULL_SCALE_EG1_GAIN) + return SYNTH_FULL_SCALE_EG1_GAIN; + return (EAS_I16) sustain; +} + +/*---------------------------------------------------------------------------- + * ConvertDelay () + *---------------------------------------------------------------------------- + * Converts timecents to frame count. Used for LFO and envelope + * delay times. + *---------------------------------------------------------------------------- +*/ +EAS_I16 ConvertDelay (EAS_I32 timeCents) +{ + EAS_I32 temp; + + if (timeCents == ZERO_TIME_IN_CENTS) + return 0; + + /* divide time by secs per frame to get number of frames */ + temp = timeCents - dlsRateConvert; + + /* convert from time cents to 10-bit fraction */ + temp = FMUL_15x15(temp, TIME_CENTS_TO_LOG2); + + /* convert to frame count */ + temp = EAS_LogToLinear16(temp - (15 << 10)); + + if (temp < SYNTH_FULL_SCALE_EG1_GAIN) + return (EAS_I16) temp; + return SYNTH_FULL_SCALE_EG1_GAIN; +} + +/*---------------------------------------------------------------------------- + * ConvertRate () + *---------------------------------------------------------------------------- + * Convert timecents to rate + *---------------------------------------------------------------------------- +*/ +EAS_I16 ConvertRate (EAS_I32 timeCents) +{ + EAS_I32 temp; + + if (timeCents == ZERO_TIME_IN_CENTS) + return SYNTH_FULL_SCALE_EG1_GAIN; + + /* divide frame rate by time in log domain to get rate */ + temp = dlsRateConvert - timeCents; + +#if 1 + temp = EAS_Calculate2toX(temp); +#else + /* convert from time cents to 10-bit fraction */ + temp = FMUL_15x15(temp, TIME_CENTS_TO_LOG2); + + /* convert to rate */ + temp = EAS_LogToLinear16(temp); +#endif + + if (temp < SYNTH_FULL_SCALE_EG1_GAIN) + return (EAS_I16) temp; + return SYNTH_FULL_SCALE_EG1_GAIN; +} + + +/*---------------------------------------------------------------------------- + * ConvertLFOPhaseIncrement() + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * + * Outputs: + * + * Side Effects: + *---------------------------------------------------------------------------- +*/ +static EAS_I16 ConvertLFOPhaseIncrement (EAS_I32 pitchCents) +{ + + /* check range */ + if (pitchCents > MAX_LFO_FREQUENCY_IN_PITCHCENTS) + pitchCents = MAX_LFO_FREQUENCY_IN_PITCHCENTS; + if (pitchCents < MIN_LFO_FREQUENCY_IN_PITCHCENTS) + pitchCents = MIN_LFO_FREQUENCY_IN_PITCHCENTS; + + /* double the rate and divide by frame rate by subtracting in log domain */ + pitchCents = pitchCents - dlsLFOFrequencyConvert; + + /* convert to phase increment */ + return (EAS_I16) EAS_Calculate2toX(pitchCents); +} + +/*---------------------------------------------------------------------------- + * ConvertPan() + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * + * Outputs: + * + * Side Effects: + *---------------------------------------------------------------------------- +*/ +static EAS_I8 ConvertPan (EAS_I32 pan) +{ + + /* multiply by conversion factor */ + pan = FMUL_15x15 (PAN_CONVERSION_FACTOR, pan); + if (pan < MIN_PAN_VALUE) + return MIN_PAN_VALUE; + if (pan > MAX_PAN_VALUE) + return MAX_PAN_VALUE; + return (EAS_I8) pan; +} + +/*---------------------------------------------------------------------------- + * ConvertQ() + *---------------------------------------------------------------------------- + * Convert the DLS filter resonance to an index value used by the synth + * that accesses tables of coefficients based on the Q. + *---------------------------------------------------------------------------- +*/ +static EAS_U8 ConvertQ (EAS_I32 q) +{ + + /* apply limits */ + if (q <= 0) + return 0; + + /* convert to table index */ + /*lint -e{704} use shift for performance */ + q = (FILTER_Q_CONVERSION_FACTOR * q + 0x4000) >> 15; + + /* apply upper limit */ + if (q >= FILTER_RESONANCE_NUM_ENTRIES) + q = FILTER_RESONANCE_NUM_ENTRIES - 1; + return (EAS_U8) q; +} + +#ifdef _DEBUG_DLS +/*---------------------------------------------------------------------------- + * DumpDLS() + *---------------------------------------------------------------------------- +*/ +static void DumpDLS (S_EAS *pEAS) +{ + S_DLS_ARTICULATION *pArt; + S_DLS_REGION *pRegion; + EAS_INT i; + EAS_INT j; + + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000022 , pEAS->numPrograms); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000023 , pEAS->numWTRegions); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000024 , pEAS->numDLSArticulations); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000025 , pEAS->numSamples); + + /* dump the instruments */ + for (i = 0; i < pEAS->numPrograms; i++) + { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000026 , + pEAS->pPrograms[i].locale >> 16, + (pEAS->pPrograms[i].locale >> 8) & 0x7f, + pEAS->pPrograms[i].locale & 0x7f); + + for (j = pEAS->pPrograms[i].regionIndex; ; j++) + { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000027 , j); + pRegion = &pEAS->pWTRegions[j]; + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000028 , pRegion->gain); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000029 , pRegion->region.rangeLow, pRegion->region.rangeHigh); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002a , pRegion->region.keyGroupAndFlags); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002b , pRegion->loopStart); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002c , pRegion->loopEnd); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002d , pRegion->tuning); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002e , pRegion->artIndex); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000002f , pRegion->waveIndex); + + if (pRegion->region.keyGroupAndFlags & REGION_FLAG_LAST_REGION) + break; + } + + } + + /* dump the articulation data */ + for (i = 0; i < pEAS->numDLSArticulations; i++) + { + /* articulation data */ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000030 , i); + pArt = &pEAS->pDLSArticulations[i]; + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000031 , pArt->m_nEG2toFilterDepth); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000032 , pArt->m_nEG2toPitchDepth); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000033 , pArt->m_nFilterCutoffFrequency); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000034 , pArt->m_nFilterResonance); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000035 , pArt->m_nLFOAmplitudeDepth); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000036 , pArt->m_nLFODelayTime); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000037 , pArt->m_nLFOFrequency); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000038 , pArt->m_nLFOPitchDepth); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000039 , pArt->m_nPan); + + /* EG1 data */ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003a , pArt->m_sEG1.m_nAttack); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003b , pArt->m_sEG1.m_nDecay); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003c , pArt->m_sEG1.m_nSustain); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003d , pArt->m_sEG1.m_nRelease); + + /* EG2 data */ + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003e , pArt->m_sEG2.m_nAttack); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x0000003f , pArt->m_sEG2.m_nDecay); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000040 , pArt->m_sEG2.m_nSustain); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000041 , pArt->m_sEG2.m_nRelease); + + } + + /* dump the waves */ + for (i = 0; i < pEAS->numSamples; i++) + { + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000042 , i); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000043 , pEAS->pSampleLen[i]); + EAS_ReportEx(_EAS_SEVERITY_NOFILTER, 0x19299ed4, 0x00000044 , pEAS->ppSamples[i]); + } + +} +#endif + diff --git a/arm-wt-22k/lib_src/eas_mdls.h b/arm-wt-22k/lib_src/eas_mdls.h new file mode 100644 index 0000000..f7dbf2f --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mdls.h @@ -0,0 +1,295 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mdls.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for eas_mdls.c
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MDLS_H
+#define _EAS_MDLS_H
+
+/*------------------------------------
+ * includes
+ *------------------------------------
+*/
+#include "eas_data.h"
+
+
+/*------------------------------------
+ * Some defines for dls.h
+ *------------------------------------
+*/
+#ifndef DWORD
+#define DWORD EAS_I32
+#define FAR
+#define SHORT EAS_I16
+#define USHORT EAS_U16
+#define LONG EAS_I32
+#define ULONG EAS_U32
+#endif
+
+
+/* GUID struct (call it DLSID in case GUID is defined elsewhere) */
+typedef struct
+{
+ EAS_U32 Data1;
+ EAS_U16 Data2;
+ EAS_U16 Data3;
+ EAS_U8 Data4[8];
+} DLSID;
+
+#define DEFINE_DLSID(name, l, w1, w2, b1, b2, b3, b4, b5, b6, b7, b8) const DLSID name = { l, w1, w2, { b1, b2, b3, b4, b5, b6, b7, b8 } }
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/* maximum sample memory for DLS query support */
+#ifndef MAX_DLS_MEMORY
+#define MAX_DLS_MEMORY 65536
+#endif
+
+/* size of conditional chunk stack */
+#ifndef CDL_STACK_SIZE
+#define CDL_STACK_SIZE 8
+#endif
+
+/* size of read buffer for sample conversion */
+#ifndef SAMPLE_CONVERT_CHUNK_SIZE
+#define SAMPLE_CONVERT_CHUNK_SIZE 32
+#endif
+
+
+#define ZERO_TIME_IN_CENTS -32768
+
+/* Pan calculation macros */
+#define PAN_CONVERSION_FACTOR 4129
+#define MAX_PAN_VALUE 63
+#define MIN_PAN_VALUE -63
+
+/* multiplier to convert time cents to 10-bit fraction log for EAS_LogToLinear16 */
+#define TIME_CENTS_TO_LOG2 27962
+
+/* conversion factor sustain level from percent to exponent for LogToLinear16 */
+#define SUSTAIN_LOG_CONVERSION_FACTOR 536871
+#define SUSTAIN_LOG_CONVERSION_SHIFT 15
+
+/* conversion factor sustain level from percent to EG full scale */
+#define SUSTAIN_LINEAR_CONVERSION_FACTOR 1073709
+
+/* conversion factor to convert frame period to decay rate */
+#define DECAY_CONVERSION_FACTOR -16
+
+/*----------------------------------------------------------------------------
+ * These macros define the various characteristics of the defined sample rates
+ *----------------------------------------------------------------------------
+ * DLS_ATTACK_TIME_CONVERT log offset for conversion from time cents to attack rate
+ * DLS_LFO_FREQUENCY_CONVERT pitch-cents offset for LFO frequency conversion
+ *----------------------------------------------------------------------------
+*/
+
+#if defined (_SAMPLE_RATE_8000)
+#define DLS_RATE_CONVERT -9559
+#define DLS_LFO_FREQUENCY_CONVERT 5921
+
+#elif defined (_SAMPLE_RATE_16000)
+#define DLS_RATE_CONVERT -9559
+#define DLS_LFO_FREQUENCY_CONVERT 5921
+
+#elif defined (_SAMPLE_RATE_20000)
+#define DLS_RATE_CONVERT -8745
+#define DLS_LFO_FREQUENCY_CONVERT 5108
+
+#elif defined (_SAMPLE_RATE_22050)
+#define DLS_RATE_CONVERT -8914
+#define DLS_LFO_FREQUENCY_CONVERT 5277
+
+#elif defined (_SAMPLE_RATE_24000)
+#define DLS_RATE_CONVERT -9061
+#define DLS_LFO_FREQUENCY_CONVERT 5423
+
+#elif defined (_SAMPLE_RATE_32000)
+#define DLS_RATE_CONVERT -9559
+#define DLS_LFO_FREQUENCY_CONVERT 5921
+
+#elif defined (_SAMPLE_RATE_44100)
+#define DLS_RATE_CONVERT -8914
+#define DLS_LFO_FREQUENCY_CONVERT 5277
+
+#elif defined (_SAMPLE_RATE_48000)
+#define DLS_RATE_CONVERT -9061
+#define DLS_LFO_FREQUENCY_CONVERT 5423
+
+#else
+#error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
+#endif
+
+/*
+ * FILTER_Q_CONVERSION_FACTOR convers the 0.1dB steps in the DLS
+ * file to our internal 0.75 dB steps. The value is calculated
+ * as follows:
+ *
+ * 32768 / (10 * <step-size in dB>)
+ *
+ * FILTER_RESONANCE_NUM_ENTRIES is the number of entries in the table
+*/
+#define FILTER_Q_CONVERSION_FACTOR 4369
+#define FILTER_RESONANCE_NUM_ENTRIES 31
+
+/*
+ * Multiplier to convert DLS gain units (10ths of a dB) to a
+ * power-of-two exponent for conversion to linear gain using our
+ * piece-wise linear approximator. Note that we ignore the lower
+ * 16-bits of the DLS gain value. The result is a 10-bit fraction
+ * that works with the EAS_LogToLinear16 function.
+ *
+ * DLS_GAIN_FACTOR = (2^18) / (200 * log10(2))
+ */
+#define DLS_GAIN_FACTOR 4354
+#define DLS_GAIN_SHIFT 8
+
+/*
+ * Reciprocal of 10 for quick divide by 10's
+ *
+ * DLS_GAIN_FACTOR = (2^18) / (200 * log10(2))
+ */
+#define DLS_DIV_10_FACTOR 3277
+#define DLS_DIV_10_SHIFT 16
+
+/*
+ * Multiplier to convert DLS time cents units to a power-of-two
+ * exponent for conversion to absolute time units using our
+ * piece-wise linear approximator.
+ *
+ * DLS_TIME_FACTOR = (2^22) / 1200
+ */
+#define DLS_TIME_FACTOR 3495
+#define DLS_TIME_SHIFT 22
+
+
+/* LFO limits */
+#define MAX_LFO_FREQUENCY_IN_HERTZ 20
+#define MIN_LFO_FREQUENCY_IN_HERTZ 0.1
+#define MAX_LFO_FREQUENCY_IN_PITCHCENTS 1549
+#define MIN_LFO_FREQUENCY_IN_PITCHCENTS -7624
+#define MAX_LFO_AMPLITUDE_DEPTH 12 /* in dB, DLS2.1 p 31*/
+#define MIN_LFO_AMPLITUDE_DEPTH -12 /* in dB, DLS2.1 p 31*/
+
+/* add to pitch cents before pow(2.0, n) to convert to frequency */
+#define ABSOLUTE_PITCH_BIAS 238395828
+
+#define A5_PITCH_OFFSET 6900
+
+/*
+CHUNK_TYPE is a macro that converts the 4 input args into a 32-bit int
+where
+argument a is placed at the MSB location and
+argument d is placed at the LSB location.
+This is useful for determining the DLS chunk types
+*/
+#define CHUNK_TYPE(a,b,c,d) ( \
+ ( ((EAS_U32)(a) & 0xFF) << 24 ) \
+ + ( ((EAS_U32)(b) & 0xFF) << 16 ) \
+ + ( ((EAS_U32)(c) & 0xFF) << 8 ) \
+ + ( ((EAS_U32)(d) & 0xFF) ) )
+
+#define CHUNK_RIFF CHUNK_TYPE('R','I','F','F')
+#define CHUNK_DLS CHUNK_TYPE('D','L','S',' ')
+#define CHUNK_CDL CHUNK_TYPE('c','d','l',' ')
+#define CHUNK_VERS CHUNK_TYPE('v','e','r','s')
+#define CHUNK_DLID CHUNK_TYPE('d','l','i','d')
+#define CHUNK_LIST CHUNK_TYPE('L','I','S','T')
+#define CHUNK_COLH CHUNK_TYPE('c','o','l','h')
+#define CHUNK_LINS CHUNK_TYPE('l','i','n','s')
+#define CHUNK_PTBL CHUNK_TYPE('p','t','b','l')
+#define CHUNK_WVPL CHUNK_TYPE('w','v','p','l')
+#define CHUNK_INFO CHUNK_TYPE('I','N','F','O')
+#define CHUNK_INAM CHUNK_TYPE('I','N','A','M')
+#define CHUNK_INS CHUNK_TYPE('i','n','s',' ')
+#define CHUNK_INSH CHUNK_TYPE('i','n','s','h')
+#define CHUNK_LRGN CHUNK_TYPE('l','r','g','n')
+#define CHUNK_RGN CHUNK_TYPE('r','g','n',' ')
+#define CHUNK_RGN2 CHUNK_TYPE('r','g','n','2')
+#define CHUNK_RGNH CHUNK_TYPE('r','g','n','h')
+#define CHUNK_WSMP CHUNK_TYPE('w','s','m','p')
+#define CHUNK_WLNK CHUNK_TYPE('w','l','n','k')
+#define CHUNK_LART CHUNK_TYPE('l','a','r','t')
+#define CHUNK_LAR2 CHUNK_TYPE('l','a','r','2')
+#define CHUNK_ART1 CHUNK_TYPE('a','r','t','1')
+#define CHUNK_ART2 CHUNK_TYPE('a','r','t','2')
+#define CHUNK_WAVE CHUNK_TYPE('w','a','v','e')
+#define CHUNK_FMT CHUNK_TYPE('f','m','t',' ')
+#define CHUNK_DATA CHUNK_TYPE('d','a','t','a')
+#define CHUNK_DMPR CHUNK_TYPE('d','m','p','r')
+
+
+#define WAVE_FORMAT_PCM 0x0001 /* Microsoft PCM format, see DLS2.1 p60 */
+#define WAVE_FORMAT_EXTENSIBLE 0xffff
+
+/* defines for wave table structures */
+
+/* initialize each articulation structure to a harmless state */
+/* change art values after we've determined EAS internals */
+#define DEFAULT_DLS_FILTER_CUTOFF_FREQUENCY 0x7FFF /* DLS2.1, p 31 means leave filter off */
+
+/**********/
+
+/* define the waves that we expect to generate instead of store */
+/* NOTE: our comparison routine converts the input string
+to lowercase, so the following comparison values should all
+be in lowercase.
+*/
+#define STRING_NOISE "noise"
+
+
+/*------------------------------------
+ * type definitions
+ *------------------------------------
+*/
+#ifdef _STANDALONE_CONVERTER
+typedef struct s_dls_params
+{
+ EAS_INT sampleRate;
+ EAS_INT samplesPerFrame;
+ EAS_INT bitDepth;
+ double ditherLevel;
+ double ditherFilterCoeff;
+ EAS_BOOL compatibility;
+ EAS_BOOL encodeADPCM;
+} S_DLS_PARAMS;
+#endif
+
+
+/* function prototypes */
+EAS_RESULT DLSParser (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, S_DLS **pDLS);
+EAS_RESULT DLSCleanup (EAS_HW_DATA_HANDLE hwInstData, S_DLS *pDLS);
+void DLSAddRef (S_DLS *pDLS);
+EAS_I16 ConvertDelay (EAS_I32 timeCents);
+EAS_I16 ConvertRate (EAS_I32 timeCents);
+
+
+#ifdef _STANDALONE_CONVERTER
+void DLSConvParams (S_DLS_PARAMS *pParams, EAS_BOOL set);
+#endif
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_midi.c b/arm-wt-22k/lib_src/eas_midi.c new file mode 100644 index 0000000..0c77aea --- /dev/null +++ b/arm-wt-22k/lib_src/eas_midi.c @@ -0,0 +1,569 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midi.c
+ *
+ * Contents and purpose:
+ * This file implements the MIDI stream parser. It is called by eas_smf.c to parse MIDI messages
+ * that are streamed out of the file. It can also parse live MIDI streams.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 794 $
+ * $Date: 2007-08-01 00:08:48 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_report.h"
+#include "eas_miditypes.h"
+#include "eas_midi.h"
+#include "eas_vm_protos.h"
+#include "eas_parser.h"
+
+#ifdef JET_INTERFACE
+#include "jet_data.h"
+#endif
+
+
+/* state enumerations for ProcessSysExMessage */
+typedef enum
+{
+ eSysEx,
+ eSysExUnivNonRealTime,
+ eSysExUnivNrtTargetID,
+ eSysExGMControl,
+ eSysExUnivRealTime,
+ eSysExUnivRtTargetID,
+ eSysExDeviceControl,
+ eSysExMasterVolume,
+ eSysExMasterVolLSB,
+ eSysExSPMIDI,
+ eSysExSPMIDIchan,
+ eSysExSPMIDIMIP,
+ eSysExMfgID1,
+ eSysExMfgID2,
+ eSysExMfgID3,
+ eSysExEnhancer,
+ eSysExEnhancerSubID,
+ eSysExEnhancerFeedback1,
+ eSysExEnhancerFeedback2,
+ eSysExEnhancerDrive,
+ eSysExEnhancerWet,
+ eSysExEOX,
+ eSysExIgnore
+} E_SYSEX_STATES;
+
+/* local prototypes */
+static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode);
+static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode);
+
+/*----------------------------------------------------------------------------
+ * EAS_InitMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the MIDI stream state for parsing.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream)
+{
+ pMIDIStream->byte3 = EAS_FALSE;
+ pMIDIStream->pending = EAS_FALSE;
+ pMIDIStream->runningStatus = 0;
+ pMIDIStream->status = 0;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_ParseMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled)
+ * so the interface works equally well for both file and stream I/O.
+ *
+ * Inputs:
+ * c - character from MIDI stream
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode)
+{
+
+ /* check for new status byte */
+ if (c & 0x80)
+ {
+ /* save new running status */
+ if (c < 0xf8)
+ {
+ pMIDIStream->runningStatus = c;
+ pMIDIStream->byte3 = EAS_FALSE;
+
+ /* deal with SysEx */
+ if ((c == 0xf7) || (c == 0xf0))
+ {
+ if (parserMode == eParserModeMetaData)
+ return EAS_SUCCESS;
+ return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode);
+ }
+
+ /* inform the file parser that we're in the middle of a message */
+ if ((c < 0xf4) || (c > 0xf6))
+ pMIDIStream->pending = EAS_TRUE;
+ }
+
+ /* real-time message - ignore it */
+ return EAS_SUCCESS;
+ }
+
+ /* 3rd byte of a 3-byte message? */
+ if (pMIDIStream->byte3)
+ {
+ pMIDIStream->d2 = c;
+ pMIDIStream->byte3 = EAS_FALSE;
+ pMIDIStream->pending = EAS_FALSE;
+ if (parserMode == eParserModeMetaData)
+ return EAS_SUCCESS;
+ return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode);
+ }
+
+ /* check for status received */
+ if (pMIDIStream->runningStatus)
+ {
+
+ /* save new status and data byte */
+ pMIDIStream->status = pMIDIStream->runningStatus;
+
+ /* check for 3-byte messages */
+ if (pMIDIStream->status < 0xc0)
+ {
+ pMIDIStream->d1 = c;
+ pMIDIStream->pending = EAS_TRUE;
+ pMIDIStream->byte3 = EAS_TRUE;
+ return EAS_SUCCESS;
+ }
+
+ /* check for 2-byte messages */
+ if (pMIDIStream->status < 0xe0)
+ {
+ pMIDIStream->d1 = c;
+ pMIDIStream->pending = EAS_FALSE;
+ if (parserMode == eParserModeMetaData)
+ return EAS_SUCCESS;
+ return ProcessMIDIMessage(pEASData, pSynth, pMIDIStream, parserMode);
+ }
+
+ /* check for more 3-bytes message */
+ if (pMIDIStream->status < 0xf0)
+ {
+ pMIDIStream->d1 = c;
+ pMIDIStream->pending = EAS_TRUE;
+ pMIDIStream->byte3 = EAS_TRUE;
+ return EAS_SUCCESS;
+ }
+
+ /* SysEx message? */
+ if (pMIDIStream->status == 0xF0)
+ {
+ if (parserMode == eParserModeMetaData)
+ return EAS_SUCCESS;
+ return ProcessSysExMessage(pEASData, pSynth, pMIDIStream, c, parserMode);
+ }
+
+ /* remaining messages all clear running status */
+ pMIDIStream->runningStatus = 0;
+
+ /* F2 is 3-byte message */
+ if (pMIDIStream->status == 0xf2)
+ {
+ pMIDIStream->byte3 = EAS_TRUE;
+ return EAS_SUCCESS;
+ }
+ }
+
+ /* no status byte received, provide a warning, but we should be able to recover */
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Received MIDI data without a valid status byte: %d\n",c); */ }
+ pMIDIStream->pending = EAS_FALSE;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * ProcessMIDIMessage()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function processes a typical MIDI message. All of the data has been received, just need
+ * to take appropriate action.
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ProcessMIDIMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_INT parserMode)
+{
+ EAS_U8 channel;
+
+ channel = pMIDIStream->status & 0x0f;
+ switch (pMIDIStream->status & 0xf0)
+ {
+ case 0x80:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ if (parserMode <= eParserModeMute)
+ VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+ break;
+
+ case 0x90:
+ if (pMIDIStream->d2)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOn: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ pMIDIStream->flags |= MIDI_FLAG_FIRST_NOTE;
+ if (parserMode == eParserModePlay)
+ VMStartNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+ }
+ else
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"NoteOff: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ if (parserMode <= eParserModeMute)
+ VMStopNote(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+ }
+ break;
+
+ case 0xa0:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PolyPres: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ break;
+
+ case 0xb0:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Control: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ if (parserMode <= eParserModeMute)
+ VMControlChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+#ifdef JET_INTERFACE
+ if (pMIDIStream->jetData & MIDI_FLAGS_JET_CB)
+ {
+ JET_Event(pEASData, pMIDIStream->jetData & (JET_EVENT_SEG_MASK | JET_EVENT_TRACK_MASK),
+ channel, pMIDIStream->d1, pMIDIStream->d2);
+ }
+#endif
+ break;
+
+ case 0xc0:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Program: %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1); */ }
+ if (parserMode <= eParserModeMute)
+ VMProgramChange(pEASData->pVoiceMgr, pSynth, channel, pMIDIStream->d1);
+ break;
+
+ case 0xd0:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"ChanPres: %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1); */ }
+ if (parserMode <= eParserModeMute)
+ VMChannelPressure(pSynth, channel, pMIDIStream->d1);
+ break;
+
+ case 0xe0:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"PBend: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ if (parserMode <= eParserModeMute)
+ VMPitchBend(pSynth, channel, pMIDIStream->d1, pMIDIStream->d2);
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL,"Unknown: %02x %02x %02x\n",
+ pMIDIStream->status, pMIDIStream->d1, pMIDIStream->d2); */ }
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * ProcessSysExMessage()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Process a SysEx character byte from the MIDI stream. Since we cannot
+ * simply wait for the next character to arrive, we are forced to save
+ * state after each character. It would be easier to parse at the file
+ * level, but then we lose the nice feature of being able to support
+ * these messages in a real-time MIDI stream.
+ *
+ * Inputs:
+ * pEASData - pointer to synthesizer instance data
+ * c - character to be processed
+ * locating - if true, the sequencer is relocating to a new position
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * These are the SysEx messages we can receive:
+ *
+ * SysEx messages
+ * { f0 7e 7f 09 01 f7 } GM 1 On
+ * { f0 7e 7f 09 02 f7 } GM 1/2 Off
+ * { f0 7e 7f 09 03 f7 } GM 2 On
+ * { f0 7f 7f 04 01 lsb msb } Master Volume
+ * { f0 7f 7f 0b 01 ch mip [ch mip ...] f7 } SP-MIDI
+ * { f0 00 01 3a 04 01 fdbk1 fdbk2 drive wet dry f7 } Enhancer
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ProcessSysExMessage (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode)
+{
+
+ /* check for start byte */
+ if (c == 0xf0)
+ {
+ pMIDIStream->sysExState = eSysEx;
+ }
+ /* check for end byte */
+ else if (c == 0xf7)
+ {
+ /* if this was a MIP message, update the MIP table */
+ if ((pMIDIStream->sysExState == eSysExSPMIDIchan) && (parserMode != eParserModeMetaData))
+ VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth);
+ pMIDIStream->sysExState = eSysExIgnore;
+ }
+
+ /* process SysEx message */
+ else
+ {
+ switch (pMIDIStream->sysExState)
+ {
+ case eSysEx:
+
+ /* first byte, determine message class */
+ switch (c)
+ {
+ case 0x7e:
+ pMIDIStream->sysExState = eSysExUnivNonRealTime;
+ break;
+ case 0x7f:
+ pMIDIStream->sysExState = eSysExUnivRealTime;
+ break;
+ case 0x00:
+ pMIDIStream->sysExState = eSysExMfgID1;
+ break;
+ default:
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+ }
+ break;
+
+ /* process GM message */
+ case eSysExUnivNonRealTime:
+ if (c == 0x7f)
+ pMIDIStream->sysExState = eSysExUnivNrtTargetID;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExUnivNrtTargetID:
+ if (c == 0x09)
+ pMIDIStream->sysExState = eSysExGMControl;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExGMControl:
+ if ((c == 1) || (c == 3))
+ {
+ /* GM 1 or GM2 On, reset synth */
+ if (parserMode != eParserModeMetaData)
+ {
+ pMIDIStream->flags |= MIDI_FLAG_GM_ON;
+ VMReset(pEASData->pVoiceMgr, pSynth, EAS_FALSE);
+ VMInitMIPTable(pSynth);
+ }
+ pMIDIStream->sysExState = eSysExEOX;
+ }
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ /* Process Master Volume and SP-MIDI */
+ case eSysExUnivRealTime:
+ if (c == 0x7f)
+ pMIDIStream->sysExState = eSysExUnivRtTargetID;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExUnivRtTargetID:
+ if (c == 0x04)
+ pMIDIStream->sysExState = eSysExDeviceControl;
+ else if (c == 0x0b)
+ pMIDIStream->sysExState = eSysExSPMIDI;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ /* process master volume */
+ case eSysExDeviceControl:
+ if (c == 0x01)
+ pMIDIStream->sysExState = eSysExMasterVolume;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExMasterVolume:
+ /* save LSB */
+ pMIDIStream->d1 = c;
+ pMIDIStream->sysExState = eSysExMasterVolLSB;
+ break;
+
+ case eSysExMasterVolLSB:
+ if (parserMode != eParserModeMetaData)
+ {
+ EAS_I32 gain = ((EAS_I32) c << 8) | ((EAS_I32) pMIDIStream->d1 << 1);
+ gain = (gain * gain) >> 15;
+ VMSetVolume(pSynth, (EAS_U16) gain);
+ }
+ pMIDIStream->sysExState = eSysExEOX;
+ break;
+
+ /* process SP-MIDI MIP message */
+ case eSysExSPMIDI:
+ if (c == 0x01)
+ {
+ /* assume all channels are muted */
+ if (parserMode != eParserModeMetaData)
+ VMInitMIPTable(pSynth);
+ pMIDIStream->d1 = 0;
+ pMIDIStream->sysExState = eSysExSPMIDIchan;
+ }
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExSPMIDIchan:
+ if (c < NUM_SYNTH_CHANNELS)
+ {
+ pMIDIStream->d2 = c;
+ pMIDIStream->sysExState = eSysExSPMIDIMIP;
+ }
+ else
+ {
+ /* bad MIP message - unmute channels */
+ if (parserMode != eParserModeMetaData)
+ VMInitMIPTable(pSynth);
+ pMIDIStream->sysExState = eSysExIgnore;
+ }
+ break;
+
+ case eSysExSPMIDIMIP:
+ /* process MIP entry here */
+ if (parserMode != eParserModeMetaData)
+ VMSetMIPEntry(pEASData->pVoiceMgr, pSynth, pMIDIStream->d2, pMIDIStream->d1, c);
+ pMIDIStream->sysExState = eSysExSPMIDIchan;
+
+ /* if 16 channels received, update MIP table */
+ if (++pMIDIStream->d1 == NUM_SYNTH_CHANNELS)
+ {
+ if (parserMode != eParserModeMetaData)
+ VMUpdateMIPTable(pEASData->pVoiceMgr, pSynth);
+ pMIDIStream->sysExState = eSysExEOX;
+ }
+ break;
+
+ /* process Enhancer */
+ case eSysExMfgID1:
+ if (c == 0x01)
+ pMIDIStream->sysExState = eSysExMfgID1;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExMfgID2:
+ if (c == 0x3a)
+ pMIDIStream->sysExState = eSysExMfgID1;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExMfgID3:
+ if (c == 0x04)
+ pMIDIStream->sysExState = eSysExEnhancer;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExEnhancer:
+ if (c == 0x01)
+ pMIDIStream->sysExState = eSysExEnhancerSubID;
+ else
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExEnhancerSubID:
+ pMIDIStream->sysExState = eSysExEnhancerFeedback1;
+ break;
+
+ case eSysExEnhancerFeedback1:
+ pMIDIStream->sysExState = eSysExEnhancerFeedback2;
+ break;
+
+ case eSysExEnhancerFeedback2:
+ pMIDIStream->sysExState = eSysExEnhancerDrive;
+ break;
+
+ case eSysExEnhancerDrive:
+ pMIDIStream->sysExState = eSysExEnhancerWet;
+ break;
+
+ case eSysExEnhancerWet:
+ pMIDIStream->sysExState = eSysExEOX;
+ break;
+
+ case eSysExEOX:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Expected F7, received %02x\n", c); */ }
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+
+ case eSysExIgnore:
+ break;
+
+ default:
+ pMIDIStream->sysExState = eSysExIgnore;
+ break;
+ }
+ }
+
+ if (pMIDIStream->sysExState == eSysExIgnore)
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Ignoring SysEx byte %02x\n", c); */ }
+ return EAS_SUCCESS;
+} /* end ProcessSysExMessage */
+
diff --git a/arm-wt-22k/lib_src/eas_midi.h b/arm-wt-22k/lib_src/eas_midi.h new file mode 100644 index 0000000..37a03ee --- /dev/null +++ b/arm-wt-22k/lib_src/eas_midi.h @@ -0,0 +1,71 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midi.h
+ *
+ * Contents and purpose:
+ * Prototypes for MIDI stream parsing functions
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDI_H
+#define _EAS_MIDI_H
+
+/*----------------------------------------------------------------------------
+ * EAS_InitMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the MIDI stream state for parsing.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_InitMIDIStream (S_MIDI_STREAM *pMIDIStream);
+
+/*----------------------------------------------------------------------------
+ * EAS_ParseMIDIStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parses a MIDI input stream character by character. Characters are pushed (rather than pulled)
+ * so the interface works equally well for both file and stream I/O.
+ *
+ * Inputs:
+ * c - character from MIDI stream
+ *
+ * Outputs:
+ * returns EAS_RESULT (EAS_SUCCESS is OK)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_ParseMIDIStream (S_EAS_DATA *pEASData, S_SYNTH *pSynth, S_MIDI_STREAM *pMIDIStream, EAS_U8 c, EAS_INT parserMode);
+
+#endif /* #define _EAS_MIDI_H */
+
diff --git a/arm-wt-22k/lib_src/eas_midictrl.h b/arm-wt-22k/lib_src/eas_midictrl.h new file mode 100644 index 0000000..0c4217d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_midictrl.h @@ -0,0 +1,64 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_midictrl.h
+ *
+ * Contents and purpose:
+ * MIDI controller definitions
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDICTRL_H
+#define _EAS_MIDICTRL_H
+
+/* define controller types */
+/*
+ Note that these controller types are specified in base 10 (decimal)
+ and not in hexadecimal. The above midi messages are specified
+ in hexadecimal.
+*/
+#define MIDI_CONTROLLER_BANK_SELECT 0
+#define MIDI_CONTROLLER_BANK_SELECT_MSB 0
+#define MIDI_CONTROLLER_MOD_WHEEL 1
+#define MIDI_CONTROLLER_ENTER_DATA_MSB 6
+#define MIDI_CONTROLLER_VOLUME 7
+#define MIDI_CONTROLLER_PAN 10
+#define MIDI_CONTROLLER_EXPRESSION 11
+#define MIDI_CONTROLLER_BANK_SELECT_LSB 32
+#define MIDI_CONTROLLER_ENTER_DATA_LSB 38 /* 0x26 */
+#define MIDI_CONTROLLER_SUSTAIN_PEDAL 64
+#define MIDI_CONTROLLER_SELECT_NRPN_LSB 98
+#define MIDI_CONTROLLER_SELECT_NRPN_MSB 99
+#define MIDI_CONTROLLER_SELECT_RPN_LSB 100 /* 0x64 */
+#define MIDI_CONTROLLER_SELECT_RPN_MSB 101 /* 0x65 */
+#define MIDI_CONTROLLER_ALL_SOUND_OFF 120
+#define MIDI_CONTROLLER_RESET_CONTROLLERS 121
+#define MIDI_CONTROLLER_ALL_NOTES_OFF 123
+#define MIDI_CONTROLLER_OMNI_OFF 124
+#define MIDI_CONTROLLER_OMNI_ON 125
+#define MIDI_CONTROLLER_MONO_ON_POLY_OFF 126
+#define MIDI_CONTROLLER_POLY_ON_MONO_OFF 127
+
+#endif /* #ifndef _EAS_MIDICTRL_H */
diff --git a/arm-wt-22k/lib_src/eas_mididata.c b/arm-wt-22k/lib_src/eas_mididata.c new file mode 100644 index 0000000..2ee907e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mididata.c @@ -0,0 +1,34 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mididata.c
+ *
+ * Contents and purpose:
+ * Data module for MIDI stream interface
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_miditypes.h"
+
+S_INTERACTIVE_MIDI eas_MIDIData;
+
diff --git a/arm-wt-22k/lib_src/eas_miditypes.h b/arm-wt-22k/lib_src/eas_miditypes.h new file mode 100644 index 0000000..0b7f96e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_miditypes.h @@ -0,0 +1,138 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_miditypes.h
+ *
+ * Contents and purpose:
+ * Contains declarations for the MIDI stream parser.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 778 $
+ * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIDITYPES_H
+#define _EAS_MIDITYPES_H
+
+#include "eas_data.h"
+#include "eas_parser.h"
+
+/*----------------------------------------------------------------------------
+ * S_MIDI_STREAM
+ *
+ * Maintains parser state for the MIDI stream parser
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_midi_stream_tag
+{
+ EAS_BOOL8 byte3; /* flag indicates 3rd byte expected */
+ EAS_BOOL8 pending; /* flag indicates more data expected */
+ EAS_U8 sysExState; /* maintains the SysEx state */
+ EAS_U8 runningStatus; /* last running status received */
+ EAS_U8 status; /* status byte */
+ EAS_U8 d1; /* first data byte */
+ EAS_U8 d2; /* second data byte */
+ EAS_U8 flags; /* flags - see below for definition */
+#ifdef JET_INTERFACE
+ EAS_U32 jetData; /* JET data */
+#endif
+} S_MIDI_STREAM;
+
+/* flags for S_MIDI_STREAM.flags */
+#define MIDI_FLAG_GM_ON 0x01 /* GM System On message received */
+#define MIDI_FLAG_FIRST_NOTE 0x02 /* first note received */
+
+/* flags for S_MIDI_STREAM.jetFlags */
+#define MIDI_FLAGS_JET_MUTE 0x00000001 /* track is muted */
+#define MIDI_FLAGS_JET_CB 0x00000002 /* JET callback enabled */
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_STREAM
+ *
+ * This structure contains data required to parse an SMF stream. For SMF0 files, there
+ * will be a single instance of this per file. For SMF1 files, there will be multiple instance,
+ * one for each separate stream in the file.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_smf_stream_tag
+{
+ EAS_FILE_HANDLE fileHandle; /* host wrapper file handle */
+ EAS_U32 ticks; /* time of next event in stream */
+ EAS_I32 startFilePos; /* start location of track within file */
+ S_MIDI_STREAM midiStream; /* MIDI stream state */
+} S_SMF_STREAM;
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_DATA
+ *
+ * This structure contains the instance data required to parse an SMF stream.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_smf_data_tag
+{
+#ifdef _CHECKED_BUILD
+ EAS_U32 handleCheck; /* signature check for checked build */
+#endif
+ S_SMF_STREAM *streams; /* pointer to individual streams in file */
+ S_SMF_STREAM *nextStream; /* pointer to next stream with event */
+ S_SYNTH *pSynth; /* pointer to synth */
+ EAS_FILE_HANDLE fileHandle; /* file handle */
+ S_METADATA_CB metadata; /* metadata callback */
+ EAS_I32 fileOffset; /* for embedded files */
+ EAS_I32 time; /* current time in milliseconds/256 */
+ EAS_U16 numStreams; /* actual number of streams */
+ EAS_U16 tickConv; /* current MIDI tick to msec conversion */
+ EAS_U16 ppqn; /* ticks per quarter note */
+ EAS_U8 state; /* current state EAS_STATE_XXXX */
+ EAS_U8 flags; /* flags - see definitions below */
+} S_SMF_DATA;
+
+#define SMF_FLAGS_CHASE_MODE 0x01 /* chase mode - skip to first note */
+#define SMF_FLAGS_HAS_TIME_SIG 0x02 /* time signature encountered at time 0 */
+#define SMF_FLAGS_HAS_TEMPO 0x04 /* tempo encountered at time 0 */
+#define SMF_FLAGS_HAS_GM_ON 0x08 /* GM System On encountered at time 0 */
+#define SMF_FLAGS_JET_STREAM 0x80 /* JET in use - keep strict timing */
+
+/* combo flags indicate setup bar */
+#define SMF_FLAGS_SETUP_BAR (SMF_FLAGS_HAS_TIME_SIG | SMF_FLAGS_HAS_TEMPO | SMF_FLAGS_HAS_GM_ON)
+
+/*----------------------------------------------------------------------------
+ * Interactive MIDI structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_interactive_midi_tag
+{
+#ifdef _CHECKED_BUILD
+ EAS_U32 handleCheck; /* signature check for checked build */
+#endif
+ S_SYNTH *pSynth; /* pointer to synth */
+ S_MIDI_STREAM stream; /* stream data */
+} S_INTERACTIVE_MIDI;
+
+#endif /* #ifndef _EAS_MIDITYPES_H */
+
diff --git a/arm-wt-22k/lib_src/eas_mixbuf.c b/arm-wt-22k/lib_src/eas_mixbuf.c new file mode 100644 index 0000000..73e969a --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mixbuf.c @@ -0,0 +1,36 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixbuf.c
+ *
+ * Contents and purpose:
+ * Contains a data allocation for synthesizer
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+// includes
+#include "eas_data.h"
+#include "eas_mixer.h"
+
+// globals
+EAS_I32 eas_MixBuffer[BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS];
+
diff --git a/arm-wt-22k/lib_src/eas_mixer.c b/arm-wt-22k/lib_src/eas_mixer.c new file mode 100644 index 0000000..c4a2f9f --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mixer.c @@ -0,0 +1,464 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixer.c
+ *
+ * Contents and purpose:
+ * This file contains the critical components of the mix engine that
+ * must be optimized for best performance.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 706 $
+ * $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+//3 dls: This module is in the midst of being converted from a synth
+//3 specific module to a general purpose mix engine
+
+/*------------------------------------
+ * includes
+ *------------------------------------
+*/
+#include "eas_data.h"
+#include "eas_host.h"
+#include "eas_math.h"
+#include "eas_mixer.h"
+#include "eas_config.h"
+#include "eas_report.h"
+
+#ifdef _MAXIMIZER_ENABLED
+EAS_I32 MaximizerProcess (EAS_VOID_PTR pInstData, EAS_I32 *pSrc, EAS_I32 *pDst, EAS_I32 numSamples);
+#endif
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/* need to boost stereo by ~3dB to compensate for the panner */
+#define STEREO_3DB_GAIN_BOOST 512
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepares the mix engine for work, allocates buffers, locates effects modules, etc.
+ *
+ * Inputs:
+ * pEASData - instance data
+ * pInstData - pointer to variable to receive instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineInit (S_EAS_DATA *pEASData)
+{
+
+ /* check Configuration Module for mix buffer allocation */
+ if (pEASData->staticMemoryModel)
+ pEASData->pMixBuffer = EAS_CMEnumData(EAS_CM_MIX_BUFFER);
+ else
+ pEASData->pMixBuffer = EAS_HWMalloc(pEASData->hwInstData, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
+ if (pEASData->pMixBuffer == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate mix buffer memory\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+ EAS_HWMemSet((void *)(pEASData->pMixBuffer), 0, BUFFER_SIZE_IN_MONO_SAMPLES * NUM_OUTPUT_CHANNELS * sizeof(EAS_I32));
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePrep()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Performs prep before synthesize a buffer of audio, such as clearing
+ * audio buffers, etc.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePrep (S_EAS_DATA *pEASData, EAS_I32 numSamples)
+{
+
+ /* clear the mix buffer */
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_HWMemSet(pEASData->pMixBuffer, 0, numSamples * (EAS_I32) sizeof(long) * 2);
+#else
+ EAS_HWMemSet(pEASData->pMixBuffer, 0, (EAS_I32) numSamples * (EAS_I32) sizeof(long));
+#endif
+
+ /* need to clear other side-chain effect buffers (chorus & reverb) */
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePost
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine does the post-processing after all voices have been
+ * synthesized. It calls any sweeteners and does the final mixdown to
+ * the output buffer.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Notes:
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePost (S_EAS_DATA *pEASData, EAS_I32 numSamples)
+{
+ EAS_U16 gain;
+
+//3 dls: Need to restore the mix engine metrics
+
+ /* calculate the gain multiplier */
+#ifdef _MAXIMIZER_ENABLED
+ if (pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effect)
+ {
+ EAS_I32 temp;
+ temp = MaximizerProcess(pEASData->effectsModules[EAS_MODULE_MAXIMIZER].effectData, pEASData->pMixBuffer, pEASData->pMixBuffer, numSamples);
+ temp = (temp * pEASData->masterGain) >> 15;
+ if (temp > 32767)
+ gain = 32767;
+ else
+ gain = (EAS_U16) temp;
+ }
+ else
+ gain = (EAS_U16) pEASData->masterGain;
+#else
+ gain = (EAS_U16) pEASData->masterGain;
+#endif
+
+ /* Not using all the gain bits for now
+ * Reduce the input to the compressor by 6dB to prevent saturation
+ */
+#ifdef _COMPRESSOR_ENABLED
+ if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
+ gain = gain >> 5;
+ else
+ gain = gain >> 4;
+#else
+ gain = gain >> 4;
+#endif
+
+ /* convert 32-bit mix buffer to 16-bit output format */
+#if (NUM_OUTPUT_CHANNELS == 2)
+ SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) ((EAS_U16) numSamples * 2));
+#else
+ SynthMasterGain(pEASData->pMixBuffer, pEASData->pOutputAudioBuffer, gain, (EAS_U16) numSamples);
+#endif
+
+#ifdef _ENHANCER_ENABLED
+ /* enhancer effect */
+ if (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_ENHANCER].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_ENHANCER].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+#ifdef _GRAPHIC_EQ_ENABLED
+ /* graphic EQ effect */
+ if (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_GRAPHIC_EQ].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+#ifdef _COMPRESSOR_ENABLED
+ /* compressor effect */
+ if (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_COMPRESSOR].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+#ifdef _WOW_ENABLED
+ /* WOW requires a 32-bit buffer, borrow the mix buffer and
+ * pass it as the destination buffer
+ */
+ /*lint -e{740} temporarily passing a parameter through an existing I/F */
+ if (pEASData->effectsModules[EAS_MODULE_WOW].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_WOW].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_WOW].effectData,
+ pEASData->pOutputAudioBuffer,
+ (EAS_PCM*) pEASData->pMixBuffer,
+ numSamples);
+#endif
+
+#ifdef _TONECONTROLEQ_ENABLED
+ /* ToneControlEQ effect */
+ if (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_TONECONTROLEQ].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+#ifdef _REVERB_ENABLED
+ /* Reverb effect */
+ if (pEASData->effectsModules[EAS_MODULE_REVERB].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_REVERB].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_REVERB].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+#ifdef _CHORUS_ENABLED
+ /* Chorus effect */
+ if (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData)
+ (*pEASData->effectsModules[EAS_MODULE_CHORUS].effect->pfProcess)
+ (pEASData->effectsModules[EAS_MODULE_CHORUS].effectData,
+ pEASData->pOutputAudioBuffer,
+ pEASData->pOutputAudioBuffer,
+ numSamples);
+#endif
+
+}
+
+#ifndef NATIVE_EAS_KERNEL
+/*----------------------------------------------------------------------------
+ * SynthMasterGain
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mixes down audio from 32-bit to 16-bit target buffer
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void SynthMasterGain (long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 numSamples) {
+
+ /* loop through the buffer */
+ while (numSamples--) {
+ long s;
+
+ /* read a sample from the input buffer and add some guard bits */
+ s = *pInputBuffer++;
+
+ /* add some guard bits */
+ /*lint -e{704} <avoid divide for performance>*/
+ s = s >> 7;
+
+ /* apply master gain */
+ s *= (long) nGain;
+
+ /* shift to lower 16-bits */
+ /*lint -e{704} <avoid divide for performance>*/
+ s = s >> 9;
+
+ /* saturate */
+ s = SATURATE(s);
+
+ *pOutputBuffer++ = (EAS_PCM)s;
+ }
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down effects modules and deallocates memory
+ *
+ * Inputs:
+ * pEASData - instance data
+ * pInstData - instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineShutdown (S_EAS_DATA *pEASData)
+{
+
+ /* check Configuration Module for static memory allocation */
+ if (!pEASData->staticMemoryModel && (pEASData->pMixBuffer != NULL))
+ EAS_HWFree(pEASData->hwInstData, pEASData->pMixBuffer);
+
+ return EAS_SUCCESS;
+}
+
+#ifdef UNIFIED_MIXER
+#ifndef NATIVE_MIX_STREAM
+/*----------------------------------------------------------------------------
+ * EAS_MixStream
+ *----------------------------------------------------------------------------
+ * Mix a 16-bit stream into a 32-bit buffer
+ *
+ * pInputBuffer 16-bit input buffer
+ * pMixBuffer 32-bit mix buffer
+ * numSamples number of samples to mix
+ * gainLeft initial gain left or mono
+ * gainRight initial gain right
+ * gainLeft left gain increment per sample
+ * gainRight right gain increment per sample
+ * flags bit 0 = stereo source
+ * bit 1 = stereo output
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags)
+{
+ EAS_I32 temp;
+ EAS_INT src, dest;
+
+ /* NOTE: There are a lot of optimizations that can be done
+ * in the native implementations based on register
+ * availability, etc. For example, it may make sense to
+ * break this down into 8 separate routines:
+ *
+ * 1. Mono source to mono output
+ * 2. Mono source to stereo output
+ * 3. Stereo source to mono output
+ * 4. Stereo source to stereo output
+ * 5. Mono source to mono output - no gain change
+ * 6. Mono source to stereo output - no gain change
+ * 7. Stereo source to mono output - no gain change
+ * 8. Stereo source to stereo output - no gain change
+ *
+ * Other possibilities include loop unrolling, skipping
+ * a gain calculation every 2 or 4 samples, etc.
+ */
+
+ /* no gain change, use fast loops */
+ if ((gainIncLeft == 0) && (gainIncRight == 0))
+ {
+ switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
+ {
+ /* mono to mono */
+ case 0:
+ gainLeft >>= 15;
+ for (src = dest = 0; src < numSamples; src++, dest++)
+ {
+
+ pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+
+ /* mono to stereo */
+ case MIX_FLAGS_STEREO_OUTPUT:
+ gainLeft >>= 15;
+ gainRight >>= 15;
+ for (src = dest = 0; src < numSamples; src++, dest+=2)
+ {
+ pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+ pMixBuffer[dest+1] += (pInputBuffer[src] * gainRight) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+
+ /* stereo to mono */
+ case MIX_FLAGS_STEREO_SOURCE:
+ gainLeft >>= 15;
+ gainRight >>= 15;
+ for (src = dest = 0; src < numSamples; src+=2, dest++)
+ {
+ temp = (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+ temp += ((pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS);
+ pMixBuffer[dest] += temp;
+ }
+ break;
+
+ /* stereo to stereo */
+ case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
+ gainLeft >>= 15;
+ gainRight >>= 15;
+ for (src = dest = 0; src < numSamples; src+=2, dest+=2)
+ {
+ pMixBuffer[dest] += (pInputBuffer[src] * gainLeft) >> NUM_MIXER_GUARD_BITS;
+ pMixBuffer[dest+1] += (pInputBuffer[src+1] * gainRight) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+ }
+ }
+
+ /* gain change - do gain increment */
+ else
+ {
+ switch (flags & (MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT))
+ {
+ /* mono to mono */
+ case 0:
+ for (src = dest = 0; src < numSamples; src++, dest++)
+ {
+ gainLeft += gainIncLeft;
+ pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+
+ /* mono to stereo */
+ case MIX_FLAGS_STEREO_OUTPUT:
+ for (src = dest = 0; src < numSamples; src++, dest+=2)
+ {
+ gainLeft += gainIncLeft;
+ gainRight += gainIncRight;
+ pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+ pMixBuffer[dest+1] += (pInputBuffer[src] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+
+ /* stereo to mono */
+ case MIX_FLAGS_STEREO_SOURCE:
+ for (src = dest = 0; src < numSamples; src+=2, dest++)
+ {
+ gainLeft += gainIncLeft;
+ gainRight += gainIncRight;
+ temp = (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+ temp += ((pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS);
+ pMixBuffer[dest] += temp;
+ }
+ break;
+
+ /* stereo to stereo */
+ case MIX_FLAGS_STEREO_SOURCE | MIX_FLAGS_STEREO_OUTPUT:
+ for (src = dest = 0; src < numSamples; src+=2, dest+=2)
+ {
+ gainLeft += gainIncLeft;
+ gainRight += gainIncRight;
+ pMixBuffer[dest] += (pInputBuffer[src] * (gainLeft >> 15)) >> NUM_MIXER_GUARD_BITS;
+ pMixBuffer[dest+1] += (pInputBuffer[src+1] * (gainRight >> 15)) >> NUM_MIXER_GUARD_BITS;
+ }
+ break;
+ }
+ }
+}
+#endif
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_mixer.h b/arm-wt-22k/lib_src/eas_mixer.h new file mode 100644 index 0000000..2ba2d3d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_mixer.h @@ -0,0 +1,137 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_mixer.h
+ *
+ * Contents and purpose:
+ * This file contains the critical components of the mix engine that
+ * must be optimized for best performance.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 706 $
+ * $Date: 2007-05-31 17:22:51 -0700 (Thu, 31 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_MIXER_H
+#define _EAS_MIXER_H
+
+//3 dls: This module is in the midst of being converted from a synth
+//3 specific module to a general purpose mix engine
+
+#define MIX_FLAGS_STEREO_SOURCE 1
+#define MIX_FLAGS_STEREO_OUTPUT 2
+#define NUM_MIXER_GUARD_BITS 4
+
+#include "eas_effects.h"
+
+extern void SynthMasterGain( long *pInputBuffer, EAS_PCM *pOutputBuffer, EAS_U16 nGain, EAS_U16 nNumLoopSamples);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepares the mix engine for work, allocates buffers, locates effects modules, etc.
+ *
+ * Inputs:
+ * pEASData - instance data
+ * pInstData - pointer to variable to receive instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineInit (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePrep()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Performs prep before synthesize a buffer of audio, such as clearing
+ * audio buffers, etc.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePrep (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEnginePost
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine does the post-processing after all voices have been
+ * synthesized. It calls any sweeteners and does the final mixdown to
+ * the output buffer.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Notes:
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixEnginePost (EAS_DATA_HANDLE pEASData, EAS_I32 nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * EAS_MixEngineShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down effects modules and deallocates memory
+ *
+ * Inputs:
+ * pEASData - instance data
+ * pInstData - instance data handle
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_MixEngineShutdown (EAS_DATA_HANDLE pEASData);
+
+#ifdef UNIFIED_MIXER
+/*----------------------------------------------------------------------------
+ * EAS_MixStream
+ *----------------------------------------------------------------------------
+ * Mix a 16-bit stream into a 32-bit buffer
+ *
+ * pInputBuffer 16-bit input buffer
+ * pMixBuffer 32-bit mix buffer
+ * numSamples number of samples to mix
+ * gainLeft initial gain left or mono
+ * gainRight initial gain right
+ * gainLeft left gain increment per sample
+ * gainRight right gain increment per sample
+ * flags bit 0 = stereo source
+ * bit 1 = stereo output
+ *----------------------------------------------------------------------------
+*/
+void EAS_MixStream (EAS_PCM *pInputBuffer, EAS_I32 *pMixBuffer, EAS_I32 numSamples, EAS_I32 gainLeft, EAS_I32 gainRight, EAS_I32 gainIncLeft, EAS_I32 gainIncRight, EAS_I32 flags);
+#endif
+
+#endif /* #ifndef _EAS_MIXER_H */
+
diff --git a/arm-wt-22k/lib_src/eas_ota.c b/arm-wt-22k/lib_src/eas_ota.c new file mode 100644 index 0000000..fb81d62 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_ota.c @@ -0,0 +1,1077 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_ota.c
+ *
+ * Contents and purpose:
+ * OTA parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 795 $
+ * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_miditypes.h"
+#include "eas_parser.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_midi.h"
+#include "eas_config.h"
+#include "eas_vm_protos.h"
+#include "eas_otadata.h"
+
+/* increase gain for mono ringtones */
+#define OTA_GAIN_OFFSET 8
+
+/* file definitions */
+#define OTA_RINGTONE 0x25
+#define OTA_SOUND 0x1d
+#define OTA_UNICODE 0x22
+
+/* song type definitions */
+#define OTA_BASIC_SONG_TYPE 0x01
+#define OTA_TEMPORARY_SONG_TYPE 0x02
+
+/* instruction ID coding */
+#define OTA_PATTERN_HEADER_ID 0x00
+#define OTA_NOTE_INST_ID 0x01
+#define OTA_SCALE_INST_ID 0x02
+#define OTA_STYLE_INST_ID 0x03
+#define OTA_TEMPO_INST_ID 0x04
+#define OTA_VOLUME_INST_ID 0x05
+
+/* note durations */
+#define OTA_NORMAL_DURATION 0x00
+#define OTA_DOTTED_NOTE 0x01
+#define OTA_DOUBLE_DOTTED_NOTE 0x02
+#define OTA_TRIPLET_NOTE 0x03
+
+/* loop count value for infinite loop */
+#define OTA_INFINITE_LOOP 0x0f
+
+/* length of 32nd note in 1/256ths of a msec for 63 BPM tempo */
+#define DEFAULT_TICK_CONV 30476
+
+/* default channel and program for OTA playback */
+#define OTA_CHANNEL 0
+#define OTA_PROGRAM 80
+#define OTA_VEL_MUL 4
+#define OTA_VEL_OFS 67
+#define OTA_VEL_DEFAULT 95
+
+/* multiplier for fixed point triplet conversion */
+#define TRIPLET_MULTIPLIER 683
+#define TRIPLET_SHIFT 10
+
+/* local prototypes */
+static EAS_RESULT OTA_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+static EAS_RESULT OTA_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT OTA_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+static EAS_RESULT OTA_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+static EAS_RESULT OTA_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT OTA_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT OTA_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT OTA_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT OTA_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT OTA_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT OTA_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT OTA_ParseHeader (S_EAS_DATA *pEASData, S_OTA_DATA* pData);
+static EAS_RESULT OTA_FetchBitField (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, EAS_I32 numBits, EAS_U8 *pValue);
+static EAS_RESULT OTA_SavePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc);
+static EAS_RESULT OTA_RestorePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc);
+
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_OTA_Parser
+ *
+ * This structure contains the functional interface for the OTA parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_OTA_Parser =
+{
+ OTA_CheckFileType,
+ OTA_Prepare,
+ OTA_Time,
+ OTA_Event,
+ OTA_State,
+ OTA_Close,
+ OTA_Reset,
+ OTA_Pause,
+ OTA_Resume,
+ NULL,
+ OTA_SetData,
+ OTA_GetData,
+ NULL
+};
+
+/*----------------------------------------------------------------------------
+ *
+ * bpmTable
+ *
+ * BPM conversion table. Converts bpm values to 256ths of a millisecond for a 32nd note
+ *----------------------------------------------------------------------------
+*/
+static const EAS_U32 bpmTable[32] =
+{
+ 76800, 68571, 61935, 54857,
+ 48000, 42667, 38400, 34286,
+ 30476, 27429, 24000, 21333,
+ 19200, 17143, 15360, 13714,
+ 12000, 10667, 9600, 8533,
+ 7680, 6737, 6000, 5408,
+ 4800, 4267, 3840, 3398,
+ 3024, 2685, 2400, 2133
+};
+
+/*----------------------------------------------------------------------------
+ * OTA_CheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset)
+{
+ S_OTA_DATA* pData;
+ EAS_RESULT result;
+ EAS_INT cmdLen;
+ EAS_INT state;
+ EAS_U8 temp;
+
+ /* read the first byte, should be command length */
+ *ppHandle = NULL;
+ if ((result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* read all the commands */
+ cmdLen = temp;
+ state = 0;
+ while (cmdLen--)
+ {
+
+ /* read the command, upper 7 bits */
+ if ((result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &temp)) != EAS_SUCCESS)
+ return result;
+ temp = temp >> 1;
+
+ if (state == 0)
+ {
+ if (temp != OTA_RINGTONE)
+ break;
+ state++;
+ }
+ else
+ {
+
+ if (temp == OTA_SOUND)
+ {
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pData = EAS_CMEnumData(EAS_CM_OTA_DATA);
+ else
+ pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_OTA_DATA));
+ if (!pData)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Malloc failed in OTA_Prepare\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+ EAS_HWMemSet(pData, 0, sizeof(S_OTA_DATA));
+
+ /* return a pointer to the instance data */
+ pData->fileHandle = fileHandle;
+ pData->fileOffset = offset;
+ pData->state = EAS_STATE_OPEN;
+ *ppHandle = pData;
+ break;
+ }
+
+ if (temp != OTA_UNICODE)
+ break;
+ }
+ }
+
+ /* not recognized */
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Prepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_OTA_DATA* pData;
+ EAS_RESULT result;
+
+ /* check for valid state */
+ pData = (S_OTA_DATA*) pInstData;
+ if (pData->state != EAS_STATE_OPEN)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* instantiate a synthesizer */
+ if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ }
+ return result;
+ }
+
+ pData->state = EAS_STATE_ERROR;
+ if ((result = OTA_ParseHeader(pEASData, pData)) != EAS_SUCCESS)
+ return result;
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Time()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the time of the next event in msecs
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pTime - pointer to variable to hold time of next event (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT OTA_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime)
+{
+ S_OTA_DATA *pData;
+
+ pData = (S_OTA_DATA*) pInstData;
+
+ /* return time in milliseconds */
+ /*lint -e{704} use shift instead of division */
+ *pTime = pData->time >> 8;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Event()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the next event in the file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode)
+{
+ S_OTA_DATA* pData;
+ EAS_RESULT result;
+ EAS_U32 duration;
+ EAS_U8 temp;
+
+ pData = (S_OTA_DATA*) pInstData;
+ if (pData->state >= EAS_STATE_OPEN)
+ return EAS_SUCCESS;
+
+ /* initialize MIDI channel when the track starts playing */
+ if (pData->time == 0)
+ {
+ /* set program to square lead */
+ if (parserMode != eParserModeMetaData)
+ VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, OTA_PROGRAM);
+
+ /* set channel volume to max */
+ if (parserMode != eParserModeMetaData)
+ VMControlChange(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, 7, 127);
+ }
+
+ /* check for end of note */
+ if (pData->note)
+ {
+ /* stop the note */
+ VMStopNote(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, pData->note, 0);
+ pData->note = 0;
+
+ /* check for rest between notes */
+ if (pData->restTicks)
+ {
+ pData->time += (EAS_I32) pData->restTicks;
+ pData->restTicks = 0;
+ return EAS_SUCCESS;
+ }
+ }
+
+ /* if not in a pattern, read the pattern header */
+ while (pData->current.patternLen == 0)
+ {
+
+ /* check for loop - don't do infinite loops when locating */
+ if (pData->loopCount && ((parserMode == eParserModePlay) || (pData->loopCount != OTA_INFINITE_LOOP)))
+ {
+ /* if not infinite loop, decrement loop count */
+ if (pData->loopCount != OTA_INFINITE_LOOP)
+ pData->loopCount--;
+
+ /* back to start of pattern*/
+ if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* if no previous position to restore, continue forward */
+ else if (pData->restore.fileOffset < 0)
+ {
+
+ /* check for end of song */
+ if (pData->numPatterns == 0)
+ {
+ pData->state = EAS_STATE_STOPPING;
+ VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ return EAS_SUCCESS;
+ }
+
+ /* read the next pattern header */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS)
+ return result;
+ if (temp != OTA_PATTERN_HEADER_ID)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA pattern header\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* get the pattern ID */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &pData->currentPattern)) != EAS_SUCCESS)
+ return result;
+
+ /* get the loop count */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &pData->loopCount)) != EAS_SUCCESS)
+ return result;
+
+ /* get the pattern length */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &pData->current.patternLen)) != EAS_SUCCESS)
+ return result;
+
+ /* if pattern definition, save the current position */
+ if (pData->current.patternLen)
+ {
+ if ((result = OTA_SavePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* if pattern length is zero, repeat a previous pattern */
+ else
+ {
+ /* make sure it's a valid pattern */
+ if (pData->patterns[pData->currentPattern].fileOffset < 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "OTA pattern error, invalid pattern specified\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* save current position and data */
+ if ((result = OTA_SavePosition(pEASData->hwInstData, pData, &pData->restore)) != EAS_SUCCESS)
+ return result;
+
+ /* seek to the pattern in the file */
+ if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->patterns[pData->currentPattern])) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* decrement pattern count */
+ pData->numPatterns--;
+ }
+
+ /* restore previous position */
+ else
+ {
+ if ((result = OTA_RestorePosition(pEASData->hwInstData, pData, &pData->restore)) != EAS_SUCCESS)
+ return result;
+ }
+ }
+
+ /* get the next event */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS)
+ return result;
+
+ switch (temp)
+ {
+ case OTA_NOTE_INST_ID:
+ /* fetch note value */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &pData->note)) != EAS_SUCCESS)
+ return result;
+
+ /* fetch note duration */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS)
+ return result;
+ duration = pData->tick * (0x20 >> temp);
+
+ /* fetch note duration modifier */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &temp)) != EAS_SUCCESS)
+ return result;
+ switch (temp)
+ {
+ case OTA_NORMAL_DURATION:
+ break;
+
+ case OTA_DOTTED_NOTE:
+ duration += duration >> 1;
+ break;
+
+ case OTA_DOUBLE_DOTTED_NOTE:
+ duration += (duration >> 1) + (duration >> 2);
+ break;
+
+ case OTA_TRIPLET_NOTE:
+ duration = (duration * TRIPLET_MULTIPLIER) >> TRIPLET_SHIFT;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unrecognized note duration ignored\n"); */ }
+ break;
+ }
+
+ /* check for note */
+ if (pData->note)
+ {
+
+ /* determine note length based on style */
+ switch (pData->style)
+ {
+ case 0:
+ pData->restTicks = duration >> 4;
+ break;
+ case 1:
+ pData->restTicks = 0;
+ break;
+ case 2:
+ pData->restTicks = duration >> 1;
+ break;
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Unrecognized note style ignored\n"); */ }
+ }
+
+ /* add octave */
+ pData->note += pData->octave;
+ if (parserMode == eParserModePlay)
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, OTA_CHANNEL, pData->note, pData->velocity);
+ pData->time += (EAS_I32) duration - (EAS_I32) pData->restTicks;
+ }
+
+ /* this is a rest */
+ else
+ pData->time += (EAS_I32) duration;
+ break;
+
+ case OTA_SCALE_INST_ID:
+ /* fetch octave */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &temp)) != EAS_SUCCESS)
+ return result;
+ pData->octave = (EAS_U8) (temp * 12 + 59);
+ break;
+
+ case OTA_STYLE_INST_ID:
+ /* fetch note style */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 2, &pData->style)) != EAS_SUCCESS)
+ return result;
+ break;
+
+ case OTA_TEMPO_INST_ID:
+ /* fetch tempo */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 5, &temp)) != EAS_SUCCESS)
+ return result;
+ pData->tick = bpmTable[temp];
+ break;
+
+ case OTA_VOLUME_INST_ID:
+ /* fetch volume */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &temp)) != EAS_SUCCESS)
+ return result;
+ pData->velocity = temp ? (EAS_U8) (temp * OTA_VEL_MUL + OTA_VEL_OFS) : 0;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unexpected instruction ID in OTA stream\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* decrement pattern length */
+ pData->current.patternLen--;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_State()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT OTA_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState)
+{
+ S_OTA_DATA* pData;
+
+ /* establish pointer to instance data */
+ pData = (S_OTA_DATA*) pInstData;
+
+ /* if stopping, check to see if synth voices are active */
+ if (pData->state == EAS_STATE_STOPPING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_STOPPED;
+ }
+
+ if (pData->state == EAS_STATE_PAUSING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_PAUSED;
+ }
+
+ /* return current state */
+ *pState = pData->state;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Close()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_OTA_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_OTA_DATA*) pInstData;
+
+ /* close the file */
+ if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS)
+ return result;
+
+ /* free the synth */
+ if (pData->pSynth != NULL)
+ VMMIDIShutdown(pEASData, pData->pSynth);
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pData);
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_OTA_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_OTA_DATA*) pInstData;
+
+ /* reset the synth */
+ VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE);
+ pData->note = 0;
+
+ /* reset file position and re-parse header */
+ pData->state = EAS_STATE_ERROR;
+ if ((result = OTA_ParseHeader (pEASData, pData)) != EAS_SUCCESS)
+ return result;
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Pause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Pauses the sequencer. Mutes all voices and sets state to pause.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_OTA_DATA *pData;
+
+ /* can't pause a stopped stream */
+ pData = (S_OTA_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* mute the synthesizer */
+ VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ pData->state = EAS_STATE_PAUSING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_Resume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume playing after a pause, sets state back to playing.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT OTA_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_OTA_DATA *pData;
+
+ /* can't resume a stopped stream */
+ pData = (S_OTA_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* nothing to do but resume playback */
+ pData->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_SetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT OTA_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_OTA_DATA *pData;
+
+ pData = (S_OTA_DATA *) pInstData;
+ switch (param)
+ {
+
+ /* set metadata callback */
+ case PARSER_DATA_METADATA_CB:
+ EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB));
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_GetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) common decoder interface - pEASData not used */
+static EAS_RESULT OTA_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_OTA_DATA *pData;
+
+ pData = (S_OTA_DATA*) pInstData;
+ switch (param)
+ {
+ /* return file type as OTA */
+ case PARSER_DATA_FILE_TYPE:
+ *pValue = EAS_FILE_OTA;
+ break;
+
+#if 0
+ /* set transposition */
+ case PARSER_DATA_TRANSPOSITION:
+ *pValue = pData->transposition;
+ break;
+#endif
+
+ case PARSER_DATA_SYNTH_HANDLE:
+ *pValue = (EAS_I32) pData->pSynth;
+ break;
+
+ case PARSER_DATA_GAIN_OFFSET:
+ *pValue = OTA_GAIN_OFFSET;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_ParseHeader()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_ParseHeader (S_EAS_DATA *pEASData, S_OTA_DATA* pData)
+{
+ EAS_RESULT result;
+ EAS_INT i;
+ EAS_INT state;
+ EAS_U8 temp;
+ EAS_U8 titleLen;
+
+ /* initialize some data */
+ pData->flags = 0;
+ pData->time = 0;
+ pData->tick = DEFAULT_TICK_CONV;
+ pData->patterns[0].fileOffset = pData->patterns[1].fileOffset =
+ pData->patterns[2].fileOffset = pData->patterns[3].fileOffset = -1;
+ pData->current.bitCount = 0;
+ pData->current.patternLen = 0;
+ pData->loopCount = 0;
+ pData->restore.fileOffset = -1;
+ pData->note = 0;
+ pData->restTicks = 0;
+ pData->velocity = OTA_VEL_DEFAULT;
+ pData->style = 0;
+ pData->octave = 59;
+
+ /* seek to start of data */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* read the first byte, should be command length */
+ if ((result = EAS_HWGetByte(pEASData->hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* read all the commands */
+ i = temp;
+ state = 0;
+ while (i--)
+ {
+
+ /* fetch command, always starts on byte boundary */
+ pData->current.bitCount = 0;
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 7, &temp)) != EAS_SUCCESS)
+ return result;
+
+ if (state == 0)
+ {
+ if (temp != OTA_RINGTONE)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA Ring Tone Programming type\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+ state++;
+ }
+ else
+ {
+
+ if (temp == OTA_SOUND)
+ break;
+
+ if (temp == OTA_UNICODE)
+ pData->flags |= OTA_FLAGS_UNICODE;
+ else
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA Sound or Unicode type\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+ }
+ }
+
+ /* get song type */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 3, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for basic song type */
+ if (temp == OTA_BASIC_SONG_TYPE)
+ {
+ /* fetch title length */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 4, &titleLen)) != EAS_SUCCESS)
+ return result;
+
+ /* if unicode, double the length */
+ if (pData->flags & OTA_FLAGS_UNICODE)
+ titleLen = (EAS_U8) (titleLen << 1);
+
+ /* zero the metadata buffer */
+ if (pData->metadata.buffer)
+ EAS_HWMemSet(pData->metadata.buffer, 0, pData->metadata.bufferSize);
+
+ /* read the song title */
+ for (i = 0; i < titleLen; i++)
+ {
+ /* fetch character */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for metadata callback */
+ if (pData->metadata.callback)
+ {
+ if (i < (pData->metadata.bufferSize - 1))
+ pData->metadata.buffer[i] = (char) temp;
+ }
+ }
+
+ /* if host has registered callback, call it now */
+ if (pData->metadata.callback)
+ (*pData->metadata.callback)(EAS_METADATA_TITLE, pData->metadata.buffer, pData->metadata.pUserData);
+ }
+
+ /* must be temporary song */
+ else if (temp != OTA_TEMPORARY_SONG_TYPE)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Expected OTA basic or temporary song type\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* get the song length */
+ if ((result = OTA_FetchBitField(pEASData->hwInstData, pData, 8, &pData->numPatterns)) != EAS_SUCCESS)
+ return result;
+
+ /* sanity check */
+ if (pData->numPatterns == 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "OTA number of patterns is zero\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* at start of first pattern */
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_FetchBitField()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Fetch a specified number of bits from the input stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_FetchBitField (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, EAS_I32 numBits, EAS_U8 *pValue)
+{
+ EAS_RESULT result;
+ EAS_I32 bitsLeft;
+ EAS_U8 value;
+
+ value = 0;
+
+ /* do we have enough bits? */
+ bitsLeft = pData->current.bitCount - numBits;
+
+ /* not enough bits, assemble them from 2 characters */
+ if (bitsLeft < 0)
+ {
+ /* grab the remaining bits from the previous byte */
+ if (pData->current.bitCount)
+ /*lint -e{504,734} this is a legitimate shift operation */
+ value = pData->current.dataByte << -bitsLeft;
+
+ /* read the next byte */
+ if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &pData->current.dataByte)) != EAS_SUCCESS)
+ return result;
+ bitsLeft += 8;
+ }
+
+ /* more bits than needed? */
+ if (bitsLeft > 0)
+ {
+ value |= pData->current.dataByte >> bitsLeft;
+ pData->current.bitCount = (EAS_U8) bitsLeft;
+ pData->current.dataByte = pData->current.dataByte & (0xff >> (8 - bitsLeft));
+ }
+
+ /* exactly the right number of bits */
+ else
+ {
+ value |= pData->current.dataByte;
+ pData->current.bitCount = 0;
+ }
+
+ *pValue = value;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_SavePosition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_SavePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc)
+{
+ EAS_HWMemCpy(pLoc, &pData->current, sizeof(S_OTA_LOC));
+ return EAS_HWFilePos(hwInstData, pData->fileHandle, &pLoc->fileOffset);
+}
+
+/*----------------------------------------------------------------------------
+ * OTA_RestorePosition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT OTA_RestorePosition (EAS_HW_DATA_HANDLE hwInstData, S_OTA_DATA *pData, S_OTA_LOC *pLoc)
+{
+ EAS_HWMemCpy(&pData->current, pLoc, sizeof(S_OTA_LOC));
+ pData->restore.fileOffset = -1;
+ return EAS_HWFileSeek(hwInstData, pData->fileHandle, pLoc->fileOffset);
+}
+
diff --git a/arm-wt-22k/lib_src/eas_otadata.c b/arm-wt-22k/lib_src/eas_otadata.c new file mode 100644 index 0000000..237f832 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_otadata.c @@ -0,0 +1,41 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_otadata..c
+ *
+ * Contents and purpose:
+ * OTA Stream Parser data module for static memory model
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_otadata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_OTAData
+ *
+ * Static memory allocation for OTA parser
+ *----------------------------------------------------------------------------
+*/
+S_OTA_DATA eas_OTAData;
+
diff --git a/arm-wt-22k/lib_src/eas_otadata.h b/arm-wt-22k/lib_src/eas_otadata.h new file mode 100644 index 0000000..63e963f --- /dev/null +++ b/arm-wt-22k/lib_src/eas_otadata.h @@ -0,0 +1,81 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_otadata.h
+ *
+ * Contents and purpose:
+ * OTA File Parser
+ *
+ * This file contains data declarations for the OTA parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_OTADATA_H
+#define EAS_OTADATA_H
+
+#include "eas_data.h"
+
+/* definition for state flags */
+#define OTA_FLAGS_UNICODE 0x01 /* unicode text */
+
+/*----------------------------------------------------------------------------
+ *
+ * S_OTA_DATA
+ *
+ * This structure contains the state data for the OTA parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+ EAS_I32 fileOffset; /* offset to location in file */
+ EAS_U8 patternLen; /* length of current pattern */
+ EAS_U8 dataByte; /* previous char from file */
+ EAS_U8 bitCount; /* bit count in char */
+} S_OTA_LOC;
+
+typedef struct
+{
+ EAS_FILE_HANDLE fileHandle; /* file handle */
+ S_SYNTH *pSynth; /* synth handle */
+ EAS_I32 fileOffset; /* offset to start of data */
+ EAS_I32 time; /* current time in 256ths of a msec */
+ EAS_U32 tick; /* length of 32nd note in 256th of a msec */
+ EAS_U32 restTicks; /* ticks to rest after current note */
+ S_OTA_LOC patterns[4]; /* pattern locations */
+ S_OTA_LOC current; /* current location */
+ S_OTA_LOC restore; /* previous location */
+ S_METADATA_CB metadata; /* metadata callback */
+ EAS_U8 flags; /* bit flags */
+ EAS_U8 numPatterns; /* number of patterns left in song */
+ EAS_U8 currentPattern; /* current pattern for loop */
+ EAS_U8 note; /* MIDI note number */
+ EAS_U8 octave; /* octave modifier */
+ EAS_U8 style; /* from STYLE */
+ EAS_U8 velocity; /* current volume */
+ EAS_U8 state; /* current state EAS_STATE_XXXX */
+ EAS_U8 loopCount; /* loop count for pattern */
+} S_OTA_DATA;
+
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_pan.c b/arm-wt-22k/lib_src/eas_pan.c new file mode 100644 index 0000000..373d90e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pan.c @@ -0,0 +1,98 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pan.c
+ *
+ * Contents and purpose:
+ * Calculates left and right gain multipliers based on a pan value from -63 to +63
+ *
+ * NOTES:
+ * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine
+ * whether the parser works for those particular file formats.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_pan.h"
+#include "eas_math.h"
+
+/*----------------------------------------------------------------------------
+ * EAS_CalcPanControl()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Assign the left and right gain values corresponding to the given pan value.
+ *
+ * This routine uses sin/cos approximations for an equal power curve:
+ *
+ * sin(x) = (2-4*c)*x^2 + c + x
+ * cos(x) = (2-4*c)*x^2 + c - x
+ *
+ * where c = 1/sqrt(2)
+ * using the a0 + x*(a1 + x*a2) approach
+ *
+ * Inputs:
+ * pan - pan value (-63 to + 63)
+ *
+ * Outputs:
+ * pGainLeft linear gain multiplier for left channel (15-bit fraction)
+ * pGainRight linear gain multiplier for left channel (15-bit fraction)
+ *
+ * Side Effects:
+ *----------------------------------------------------------------------------
+*/
+void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight)
+{
+ EAS_INT temp;
+ EAS_INT netAngle;
+
+ /* impose hard limit */
+ if (pan < -63)
+ netAngle = -63;
+ else if (pan > 63)
+ netAngle = 63;
+ else
+ netAngle = pan;
+
+ /*lint -e{701} <avoid multiply for performance reasons>*/
+ netAngle = netAngle << 8;
+
+ /* calculate sin */
+ temp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle);
+ temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle);
+
+ if (temp > SYNTH_FULL_SCALE_EG1_GAIN)
+ temp = SYNTH_FULL_SCALE_EG1_GAIN;
+ else if (temp < 0)
+ temp = 0;
+
+ *pGainRight = (EAS_I16) temp;
+
+ /* calculate cos */
+ temp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, netAngle);
+ temp = COEFF_PAN_G0 + FMUL_15x15(temp, netAngle);
+ if (temp > SYNTH_FULL_SCALE_EG1_GAIN)
+ temp = SYNTH_FULL_SCALE_EG1_GAIN;
+ else if (temp < 0)
+ temp = 0;
+
+ *pGainLeft = (EAS_I16) temp;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_pan.h b/arm-wt-22k/lib_src/eas_pan.h new file mode 100644 index 0000000..cefa650 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pan.h @@ -0,0 +1,66 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pan.h
+ *
+ * Contents and purpose:
+ * Calculates left and right gain multipliers based on a pan value from -63 to +63
+ *
+ * NOTES:
+ * The _CMX_PARSER and _MFI_PARSER preprocessor symbols determine
+ * whether the parser works for those particular file formats.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_PAN_H
+#define _EAS_PAN_H
+
+#include "eas_types.h"
+
+/*----------------------------------------------------------------------------
+ * EAS_CalcPanControl()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Assign the left and right gain values corresponding to the given pan value.
+ *
+ * This routine uses sin/cos approximations for an equal power curve:
+ *
+ * sin(x) = (2-4*c)*x^2 + c + x
+ * cos(x) = (2-4*c)*x^2 + c - x
+ *
+ * where c = 1/sqrt(2)
+ * using the a0 + x*(a1 + x*a2) approach
+ *
+ * Inputs:
+ * pan - pan value (-63 to + 63)
+ *
+ * Outputs:
+ * pGainLeft linear gain multiplier for left channel (15-bit fraction)
+ * pGainRight linear gain multiplier for left channel (15-bit fraction)
+ *
+ * Side Effects:
+ *----------------------------------------------------------------------------
+*/
+void EAS_CalcPanControl (EAS_INT pan, EAS_I16 *pGainLeft, EAS_I16 *pGainRight);
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_parser.h b/arm-wt-22k/lib_src/eas_parser.h new file mode 100644 index 0000000..5512c82 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_parser.h @@ -0,0 +1,98 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_parser.h
+ *
+ * Contents and purpose:
+ * Interface declarations for the generic parser interface
+ *
+ * This header only contains declarations that are specific
+ * to this implementation.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 767 $
+ * $Date: 2007-07-19 13:47:31 -0700 (Thu, 19 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PARSER_H
+#define _EAS_PARSER_H
+
+#include "eas_types.h"
+
+
+/* metadata callback */
+typedef struct s_metadata_cb_tag
+{
+ EAS_METADATA_CBFUNC callback;
+ char *buffer;
+ EAS_VOID_PTR pUserData;
+ EAS_I32 bufferSize;
+} S_METADATA_CB;
+
+/* generic parser interface */
+typedef struct
+{
+ EAS_RESULT (* EAS_CONST pfCheckFileType)(struct s_eas_data_tag *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+ EAS_RESULT (* EAS_CONST pfPrepare)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (* EAS_CONST pfTime)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+ EAS_RESULT (* EAS_CONST pfEvent)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_INT parseMode);
+ EAS_RESULT (* EAS_CONST pfState)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+ EAS_RESULT (* EAS_CONST pfClose)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (* EAS_CONST pfReset)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (* EAS_CONST pfPause)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (* EAS_CONST pfResume)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData);
+ EAS_RESULT (* EAS_CONST pfLocate)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate);
+ EAS_RESULT (* EAS_CONST pfSetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+ EAS_RESULT (* EAS_CONST pfGetData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+ EAS_RESULT (* EAS_CONST pfGetMetaData)(struct s_eas_data_tag *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength);
+} S_FILE_PARSER_INTERFACE;
+
+typedef enum
+{
+ eParserModePlay,
+ eParserModeLocate,
+ eParserModeMute,
+ eParserModeMetaData
+} E_PARSE_MODE;
+
+typedef enum
+{
+ PARSER_DATA_FILE_TYPE,
+ PARSER_DATA_PLAYBACK_RATE,
+ PARSER_DATA_TRANSPOSITION,
+ PARSER_DATA_VOLUME,
+ PARSER_DATA_SYNTH_HANDLE,
+ PARSER_DATA_METADATA_CB,
+ PARSER_DATA_DLS_COLLECTION,
+ PARSER_DATA_EAS_LIBRARY,
+ PARSER_DATA_POLYPHONY,
+ PARSER_DATA_PRIORITY,
+ PARSER_DATA_FORMAT,
+ PARSER_DATA_MEDIA_LENGTH,
+ PARSER_DATA_JET_CB,
+ PARSER_DATA_MUTE_FLAGS,
+ PARSER_DATA_SET_MUTE,
+ PARSER_DATA_CLEAR_MUTE,
+ PARSER_DATA_NOTE_COUNT,
+ PARSER_DATA_MAX_PCM_STREAMS,
+ PARSER_DATA_GAIN_OFFSET,
+ PARSER_DATA_PLAY_MODE
+} E_PARSER_DATA;
+
+#endif /* #ifndef _EAS_PARSER_H */
diff --git a/arm-wt-22k/lib_src/eas_pcm.c b/arm-wt-22k/lib_src/eas_pcm.c new file mode 100644 index 0000000..64b8f71 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pcm.c @@ -0,0 +1,1482 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcm.c
+ *
+ * Contents and purpose:
+ * Implements the PCM engine including ADPCM decode for SMAF and CMX audio playback.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 849 $
+ * $Date: 2007-08-28 08:59:11 -0700 (Tue, 28 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_config.h"
+#include "eas_parser.h"
+#include "eas_pcm.h"
+#include "eas_math.h"
+#include "eas_mixer.h"
+
+#define PCM_MIXER_GUARD_BITS (NUM_MIXER_GUARD_BITS + 1)
+
+/*----------------------------------------------------------------------------
+ * Decoder interfaces
+ *----------------------------------------------------------------------------
+*/
+
+static EAS_RESULT LinearPCMDecode (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+static EAS_RESULT LinearPCMLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time);
+
+static const S_DECODER_INTERFACE PCMDecoder =
+{
+ NULL,
+ LinearPCMDecode,
+ LinearPCMLocate,
+};
+
+/* SMAF ADPCM decoder */
+#ifdef _SMAF_PARSER
+extern S_DECODER_INTERFACE SmafDecoder;
+#define SMAF_DECODER &SmafDecoder
+extern S_DECODER_INTERFACE Smaf7BitDecoder;
+#define SMAF_7BIT_DECODER &Smaf7BitDecoder
+#else
+#define SMAF_DECODER NULL
+#define SMAF_7BIT_DECODER NULL
+#endif
+
+/* IMA ADPCM decoder */
+#ifdef _IMA_DECODER
+extern S_DECODER_INTERFACE IMADecoder;
+#define IMA_DECODER &IMADecoder
+#else
+#define IMA_DECODER NULL
+#endif
+
+static const S_DECODER_INTERFACE * const decoders[] =
+{
+ &PCMDecoder,
+ SMAF_DECODER,
+ IMA_DECODER,
+ SMAF_7BIT_DECODER
+};
+
+/*----------------------------------------------------------------------------
+ * Sample rate conversion
+ *----------------------------------------------------------------------------
+*/
+
+#define SRC_RATE_MULTIPLER (0x40000000 / _OUTPUT_SAMPLE_RATE)
+
+#ifdef _LOOKUP_SAMPLE_RATE
+static const EAS_U32 srcConvRate[][2] =
+{
+ 4000L, (4000L << 15) / _OUTPUT_SAMPLE_RATE,
+ 8000L, (8000L << 15) / _OUTPUT_SAMPLE_RATE,
+ 11025L, (11025L << 15) / _OUTPUT_SAMPLE_RATE,
+ 12000L, (12000L << 15) / _OUTPUT_SAMPLE_RATE,
+ 16000L, (16000L << 15) / _OUTPUT_SAMPLE_RATE,
+ 22050L, (22050L << 15) / _OUTPUT_SAMPLE_RATE,
+ 24000L, (24000L << 15) / _OUTPUT_SAMPLE_RATE,
+ 32000L, (32000L << 15) / _OUTPUT_SAMPLE_RATE
+};
+static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate);
+#define SRC_CONV_RATE_ENTRIES (sizeof(srcConvRate)/sizeof(EAS_U32)/2)
+#endif
+
+
+/* interface prototypes */
+static EAS_RESULT RenderPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 numSamples);
+
+
+/* local prototypes */
+static S_PCM_STATE *FindSlot (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_PCM_CALLBACK pCallbackFunc, EAS_VOID_PTR cbInstData);
+static EAS_RESULT InitPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEInit (S_EAS_DATA *pEASData)
+{
+ S_PCM_STATE *pState;
+ EAS_INT i;
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pEASData->pPCMStreams = EAS_CMEnumData(EAS_CM_PCM_DATA);
+ /* allocate dynamic memory */
+ else
+ pEASData->pPCMStreams = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_PCM_STATE) * MAX_PCM_STREAMS);
+
+ if (!pEASData->pPCMStreams)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate memory for PCM streams\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+
+ //zero the memory to insure complete initialization
+ EAS_HWMemSet((void *)(pEASData->pPCMStreams),0, sizeof(S_PCM_STATE) * MAX_PCM_STREAMS);
+
+ /* initialize the state data */
+ for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++)
+ pState->fileHandle = NULL;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEShutdown (S_EAS_DATA *pEASData)
+{
+
+ /* free any dynamic memory */
+ if (!pEASData->staticMemoryModel)
+ {
+ if (pEASData->pPCMStreams)
+ {
+ EAS_HWFree(pEASData->hwInstData, pEASData->pPCMStreams);
+ pEASData->pPCMStreams = NULL;
+ }
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PERender()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Render a buffer of PCM audio
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PERender (S_EAS_DATA* pEASData, EAS_I32 numSamples)
+{
+ S_PCM_STATE *pState;
+ EAS_RESULT result;
+ EAS_INT i;
+
+ /* render all the active streams */
+ for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++)
+ {
+ if ((pState->fileHandle) && (pState->state != EAS_STATE_STOPPED) && (pState->state != EAS_STATE_PAUSED))
+ if ((result = RenderPCMStream(pEASData, pState, numSamples)) != EAS_SUCCESS)
+ return result;
+ }
+ return EAS_SUCCESS;
+}
+
+
+/*----------------------------------------------------------------------------
+ * EAS_PEState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * This interface is also exposed in the internal library for use by the other modules.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEState (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pInstData, EAS_STATE *pState)
+{
+ /* return current state */
+ *pState = pInstData->state;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEClose()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEClose (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState)
+{
+ EAS_RESULT result;
+
+ if ((result = EAS_HWCloseFile(pEASData->hwInstData, pState->fileHandle)) != EAS_SUCCESS)
+ return result;
+
+ pState->fileHandle = NULL;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * PCM_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEReset (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState)
+{
+ EAS_RESULT result;
+
+ /* reset file position to first byte of data in the stream */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, pState->startPos)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %d seeking to start of PCM file\n", result); */ }
+ return result;
+ }
+
+ /* re-initialize stream */
+ return InitPCMStream(pEASData, pState);
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEOpenStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Starts up a PCM playback
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEOpenStream (S_EAS_DATA *pEASData, S_PCM_OPEN_PARAMS *pParams, EAS_PCM_HANDLE *pHandle)
+{
+ EAS_RESULT result;
+ S_PCM_STATE *pState;
+ EAS_I32 filePos;
+
+ /* make sure we support this decoder */
+ if (pParams->decoder >= NUM_DECODER_MODULES)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Decoder selector out of range\n"); */ }
+ return EAS_ERROR_PARAMETER_RANGE;
+ }
+ if (decoders[pParams->decoder] == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Decoder module not available\n"); */ }
+ return EAS_ERROR_FEATURE_NOT_AVAILABLE;
+ }
+
+ /* find a slot for the new stream */
+ if ((pState = FindSlot(pEASData, pParams->fileHandle, pParams->pCallbackFunc, pParams->cbInstData)) == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unable to open ADPCM stream, too many streams open\n"); */ }
+ return EAS_ERROR_MAX_PCM_STREAMS;
+ }
+
+ /* get the current file position */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, pState->fileHandle, &filePos)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_HWFilePos returned %ld\n",result); */ }
+ pState->fileHandle = NULL;
+ return result;
+ }
+
+ pState->pDecoder = decoders[pParams->decoder];
+ pState->startPos = filePos;
+ pState->bytesLeftLoop = pState->byteCount = pParams->size;
+ pState->loopStart = pParams->loopStart;
+ pState->samplesTilLoop = (EAS_I32) pState->loopStart;
+ pState->loopSamples = pParams->loopSamples;
+ pState->samplesInLoop = 0;
+ pState->blockSize = (EAS_U16) pParams->blockSize;
+ pState->flags = pParams->flags;
+ pState->envData = pParams->envData;
+ pState->volume = pParams->volume;
+ pState->sampleRate = (EAS_U16) pParams->sampleRate;
+
+ /* set the base frequency */
+ pState->basefreq = (SRC_RATE_MULTIPLER * (EAS_U32) pParams->sampleRate) >> 15;
+
+ /* calculate shift for frequencies > 1.0 */
+ pState->rateShift = 0;
+ while (pState->basefreq > 32767)
+ {
+ pState->basefreq = pState->basefreq >> 1;
+ pState->rateShift++;
+ }
+
+ /* initialize */
+ if ((result = InitPCMStream(pEASData, pState)) != EAS_SUCCESS)
+ return result;
+
+ *pHandle = pState;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PEOpenStream: StartPos=%d, byteCount = %d, loopSamples=%d\n",
+ pState->startPos, pState->byteCount, pState->loopSamples); */ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEContinueStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Continues a PCM stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -e{715} reserved for future use */
+EAS_RESULT EAS_PEContinueStream (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_I32 size)
+{
+
+ /* add new samples to count */
+ pState->bytesLeft += size;
+ if (pState->bytesLeft > 0)
+ pState->flags &= ~PCM_FLAGS_EMPTY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEGetFileHandle()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the file handle of a stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEGetFileHandle (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_FILE_HANDLE *pFileHandle)
+{
+ *pFileHandle = pState->fileHandle;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateParams()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch and volume parameters for a PCM stream
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ * gainLeft - linear gain multipler in 1.15 fraction format
+ * gainRight - linear gain multipler in 1.15 fraction format
+ * pitch - pitch shift in cents
+ * initial - initial settings, set current gain
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes
+ * In mono mode, leftGain controls the output gain and rightGain is ignored
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+/*lint -esym(715, gainRight) used only in 2-channel version */
+EAS_RESULT EAS_PEUpdateParams (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch, EAS_I16 gainLeft, EAS_I16 gainRight)
+{
+
+ pState->gainLeft = gainLeft;
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ pState->gainRight = gainRight;
+#endif
+
+ pState->pitch = pitch;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PELocate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function seeks to the requested place in the file. Accuracy
+ * is dependent on the sample rate and block size.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pState - stream handle
+ * time - media time in milliseconds
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PELocate (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState, EAS_I32 time)
+{
+ if (pState->pDecoder->pfLocate == NULL)
+ return EAS_ERROR_FEATURE_NOT_AVAILABLE;
+
+ return pState->pDecoder->pfLocate(pEASData, pState, time);
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the volume parameters for a PCM stream
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ * gainLeft - linear gain multipler in 1.15 fraction format
+ * gainRight - linear gain multipler in 1.15 fraction format
+ * initial - initial settings, set current gain
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes
+ * In mono mode, leftGain controls the output gain and rightGain is ignored
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdateVolume (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 volume)
+{
+ pState->volume = volume;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdatePitch()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch parameter for a PCM stream
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * pState - pointer to S_PCM_STATE for this stream
+ * pitch - new pitch value in pitch cents
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdatePitch (S_EAS_DATA* pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch)
+{
+ pState->pitch = pitch;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEPause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mute and stop rendering a PCM stream. Sets the gain target to zero and stops the playback
+ * at the end of the next audio frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEPause (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState)
+{
+ /* set state to stopping */
+ pState->state = EAS_STATE_PAUSING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PEResume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume rendering a PCM stream. Sets the gain target back to its
+ * previous setting and restarts playback at the end of the next audio
+ * frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEResume (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState)
+{
+ /* set state to stopping */
+ pState->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+EAS_U32 getDecayScale(EAS_U32 index)
+{
+ EAS_U32 utemp;
+
+ //envelope decay segment
+ switch (index)
+ {
+ case 0: //no decay
+ utemp = 512;//32768;
+ break;
+ case 1: //.0156 dB per update
+ utemp = 511;//32709;
+ break;
+ case 2: //.03125
+ utemp = 510;//32649;
+ break;
+ case 3: //.0625
+ utemp = 508;//32532;
+ break;
+ case 4: //.125
+ utemp = 505;//32298;
+ break;
+ case 5: //.25
+ utemp = 497;//31835;
+ break;
+ case 6: //.5
+ utemp = 483;//30929;
+ break;
+ case 7: //1.0
+ utemp = 456;//29193;
+ break;
+ case 8: //2.0
+ utemp = 406;//26008;
+ break;
+ case 9: //4.0
+ utemp = 323;//20642;
+ break;
+ case 10: //8.0
+ utemp = 203;//13004;
+ break;
+ case 11: //16.0
+ utemp = 81;//5160;
+ break;
+ case 12: //32.0
+ utemp = 13;//813;
+ break;
+ case 13: //64.0
+ utemp = 0;//20;
+ break;
+ case 14: //128.0
+ utemp = 0;
+ break;
+ case 15: //256.0
+ default:
+ utemp = 0;
+ break;
+ }
+ //printf("getdecayscale returned %d\n",utemp);
+ return utemp;
+}
+
+EAS_U32 getAttackIncrement(EAS_U32 index)
+{
+ EAS_U32 utemp;
+
+ //envelope decay segment
+ switch (index)
+ {
+ case 0:
+ utemp = 32;
+ break;
+ case 1:
+ utemp = 64;
+ break;
+ case 2:
+ utemp = 128;
+ break;
+ case 3:
+ utemp = 256;
+ break;
+ case 4:
+ utemp = 512;
+ break;
+ case 5:
+ utemp = 1024;
+ break;
+ case 6:
+ utemp = 2048;
+ break;
+ case 7:
+ utemp = 4096;
+ break;
+ case 8:
+ utemp = 8192;
+ break;
+ case 9:
+ utemp = 16384;
+ break;
+ case 10:
+ utemp = 32768;
+ break;
+ case 11:
+ utemp = 65536;
+ break;
+ case 12:
+ utemp = 65536;
+ break;
+ case 13:
+ utemp = 65536;
+ break;
+ case 14:
+ utemp = 65535;
+ break;
+ case 15:
+ default:
+ utemp = 0;
+ break;
+ }
+ //printf("getattackincrement returned %d\n",utemp);
+ return utemp;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PERelease()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Put the PCM stream envelope into release.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PERelease (S_EAS_DATA *pEASData, EAS_PCM_HANDLE pState)
+{
+ EAS_U32 utemp;
+
+ //printf("handling note-off part of envelope\n");
+ /*if the note is not ignore release or sustained*/
+ if (((pState->envData >> 24) & 0x0F)==0)
+ {
+ /* set envelope state to release */
+ pState->envState = PCM_ENV_RELEASE;
+ utemp = ((pState->envData >> 20) & 0x0F);
+ pState->envScale = getDecayScale(utemp); //getReleaseScale(utemp);
+ }
+ else
+ {
+ /*else change envelope state to sustain */
+ pState->envState = PCM_ENV_SUSTAIN;
+ utemp = ((pState->envData >> 28) & 0x0F);
+ pState->envScale = getDecayScale(utemp); //getSustainScale(utemp);
+ }
+ //since we are in release, don't let anything hang around too long
+ //printf("checking env scale, val = %d\n",((S_PCM_STATE*) handle)->envScale);
+ if (pState->envScale > 505)
+ pState->envScale = 505;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * FindSlot()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Locates an empty stream slot and assigns the file handle
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * fileHandle - file handle
+ * pCallbackFunc - function to be called back upon EAS_STATE_STOPPED
+ *
+ * Outputs:
+ * returns handle to slot or NULL if all slots are used
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static S_PCM_STATE *FindSlot (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_PCM_CALLBACK pCallbackFunc, EAS_VOID_PTR cbInstData)
+{
+ EAS_INT i;
+ S_PCM_STATE *pState;
+
+#ifndef NO_PCM_STEAL
+ S_PCM_STATE *foundState = NULL;
+ EAS_INT count = 0;
+ EAS_U32 startOrder = 0xFFFFFFFF;
+ S_PCM_STATE *stealState = NULL;
+ EAS_U32 youngest = 0;
+
+ /* find an empty slot, count total in use, and find oldest in use (lowest start order) */
+ for (i = 0, pState = pEASData->pPCMStreams; i < MAX_PCM_STREAMS; i++, pState++)
+ {
+ /* if this one is available */
+ if (pState->fileHandle == NULL)
+ {
+ foundState = pState;
+ }
+ /* else this one is in use, so see if it is the oldest, and count total in use */
+ /* also find youngest */
+ else
+ {
+ /*one more voice in use*/
+ count++;
+ /* is this the oldest? (lowest start order) */
+ if ((pState->state != EAS_STATE_STOPPING) && (pState->startOrder < startOrder))
+ {
+ /* remember this one */
+ stealState = pState;
+ /* remember the oldest so far */
+ startOrder = pState->startOrder;
+ }
+ /* is this the youngest? (highest start order) */
+ if (pState->startOrder >= youngest)
+ {
+ youngest = pState->startOrder;
+ }
+ }
+ }
+
+ /* if there are too many voices active, stop the oldest one */
+ if (count > PCM_STREAM_THRESHOLD)
+ {
+ //printf("stealing!!!\n");
+ /* make sure we got one, although we should always have one at this point */
+ if (stealState != NULL)
+ {
+ //flag this as stopping, so it will get shut off
+ stealState->state = EAS_STATE_STOPPING;
+ }
+ }
+
+ /* if there are no available open streams (we won't likely see this, due to stealing) */
+ if (foundState == NULL)
+ return NULL;
+
+ /* save info */
+ foundState->startOrder = youngest + 1;
+ foundState->fileHandle = fileHandle;
+ foundState->pCallback = pCallbackFunc;
+ foundState->cbInstData = cbInstData;
+ return foundState;
+#else
+ /* find an empty slot*/
+ for (i = 0; i < MAX_PCM_STREAMS; i++)
+ {
+ pState = &pEASData->pPCMStreams[i];
+ if (pState->fileHandle != NULL)
+ continue;
+
+ pState->fileHandle = fileHandle;
+ pState->pCallback = pCallbackFunc;
+ pState->cbInstData = cbInstData;
+ return pState;
+ }
+ return NULL;
+#endif
+}
+
+#ifdef _LOOKUP_SAMPLE_RATE
+/*----------------------------------------------------------------------------
+ * CalcBaseFreq()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculates the fractional phase increment for the sample rate converter
+ *
+ * Inputs:
+ * sampleRate - sample rate in samples/sec
+ *
+ * Outputs:
+ * Returns fractional sample rate with a 15-bit fraction
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_U32 CalcBaseFreq (EAS_U32 sampleRate)
+{
+ EAS_INT i;
+
+ /* look up the conversion rate */
+ for (i = 0; i < (EAS_INT)(SRC_CONV_RATE_ENTRIES); i ++)
+ {
+ if (srcConvRate[i][0] == sampleRate)
+ return srcConvRate[i][1];
+ }
+
+ /* if not found in table, do it the long way */
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Sample rate %u not in table, calculating by division\n", sampleRate); */ }
+
+ return (SRC_RATE_MULTIPLER * (EAS_U32) sampleRate) >> 15;
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * InitPCMStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Start an ADPCM stream playback. Decodes the header, preps the engine.
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT InitPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState)
+{
+
+ /* initialize the data structure */
+ pState->bytesLeft = pState->byteCount;
+ pState->phase = 0;
+ pState->srcByte = 0;
+ pState->decoderL.acc = 0;
+ pState->decoderL.output = 0;
+ pState->decoderL.x0 = pState->decoderL.x1 = 0;
+ pState->decoderL.step = 0;
+ pState->decoderR.acc = 0;
+ pState->decoderR.output = 0;
+ pState->decoderR.x0 = pState->decoderR.x1 = 0;
+ pState->decoderR.step = 0;
+ pState->hiNibble = EAS_FALSE;
+ pState->pitch = 0;
+ pState->blockCount = 0;
+ pState->gainLeft = PCM_DEFAULT_GAIN_SETTING;
+// pState->currentGainLeft = PCM_DEFAULT_GAIN_SETTING;
+ pState->envValue = 0;
+ pState->envState = PCM_ENV_START;
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ pState->gainRight = PCM_DEFAULT_GAIN_SETTING;
+// pState->currentGainRight = PCM_DEFAULT_GAIN_SETTING;
+#endif
+ pState->state = EAS_STATE_READY;
+
+ /* initialize the decoder */
+ if (pState->pDecoder->pfInit)
+ return (*pState->pDecoder->pfInit)(pEASData, pState);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RenderPCMStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes a buffer of ADPCM data.
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RenderPCMStream (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 numSamples)
+{
+ EAS_RESULT result;
+ EAS_U32 phaseInc;
+ EAS_I32 gainLeft, gainIncLeft;
+ EAS_I32 *pOut;
+ EAS_I32 temp;
+ EAS_U32 utemp;
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_I32 gainRight, gainIncRight;
+#endif
+
+#if 0
+ printf("env data: AR = %d, DR = %d, SL = %d, SR = %d, RR = %d\n",
+ ((pState->envData >> 12) & 0x0F),
+ ((pState->envData >> 16) & 0x0F),
+ ((pState->envData >> 8) & 0x0F),
+ ((pState->envData >> 28) & 0x0F),
+ ((pState->envData >> 20) & 0x0F));
+#endif
+
+ if (pState->envState == PCM_ENV_START)
+ {
+ //printf("env start\n");
+ utemp = ((pState->envData >> 12) & 0x0F);
+ //if fastest rate, attack is already completed
+ //do the same for slowest rate, since that allows zero to be passed for default envelope
+ if (utemp == 0x0F || utemp == 0x00)
+ {
+ //start envelope at full
+ pState->envValue = (32768<<7);
+ //jump right into decay
+ utemp = ((pState->envData >> 16) & 0x0F);
+ pState->envScale = getDecayScale(utemp);
+ pState->envState = PCM_ENV_DECAY;
+ pState->currentGainLeft = (EAS_I16) FMUL_15x15(pState->gainLeft, pState->volume);
+ pState->currentGainRight = (EAS_I16) FMUL_15x15(pState->gainRight, pState->volume);
+ }
+ //else attack has a ramp
+ else
+ {
+ //start the envelope very low
+ pState->envValue = (2<<7);
+ pState->currentGainLeft = 0;
+ pState->currentGainRight = 0;
+ //get envelope attack scaling value
+ pState->envScale = getAttackIncrement(utemp);
+ //go to attack state
+ pState->envState = PCM_ENV_ATTACK;
+ }
+ }
+ if (pState->envState == PCM_ENV_ATTACK)
+ {
+ //printf("env attack, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale);
+ //update envelope value
+ pState->envValue = pState->envValue + (pState->envScale << 7);
+ //check envelope level and update state if needed
+ if (pState->envValue >= (32768<<7))
+ {
+ pState->envValue = (32768<<7);
+ utemp = ((pState->envData >> 16) & 0x0F);
+ pState->envScale = getDecayScale(utemp);
+ pState->envState = PCM_ENV_DECAY;
+ }
+ }
+ else if (pState->envState == PCM_ENV_DECAY)
+ {
+ //printf("env decay, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale);
+ //update envelope value
+ pState->envValue = (pState->envValue * pState->envScale)>>9;
+ //check envelope level against sustain level and update state if needed
+ utemp = ((pState->envData >> 8) & 0x0F);
+ if (utemp == (EAS_U32)0x0F)
+ utemp = (2<<7);
+ else
+ {
+ utemp = ((32769<<7) >> (utemp>>1));
+ }
+ if (pState->envValue <= utemp)
+ {
+ utemp = ((pState->envData >> 28) & 0x0F);
+ pState->envScale = getDecayScale(utemp); //getSustainScale(utemp);
+ pState->envState = PCM_ENV_SUSTAIN;
+ }
+ }
+ else if (pState->envState == PCM_ENV_SUSTAIN)
+ {
+ //printf("env sustain, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale);
+ //update envelope value
+ pState->envValue = (pState->envValue * pState->envScale)>>9;
+ //check envelope level against bottom level and update state if needed
+ if (pState->envValue <= (2<<7))
+ {
+ //no more decay
+ pState->envScale = 512;
+ pState->envState = PCM_ENV_END;
+ }
+ }
+ else if (pState->envState == PCM_ENV_RELEASE)
+ {
+ //printf("env release, env value = %d, env scale = %d\n",pState->envValue>>7,pState->envScale);
+ //update envelope value
+ pState->envValue = (pState->envValue * pState->envScale)>>9;
+ //check envelope level against bottom level and update state if needed
+ if (pState->envValue <= (2<<7))
+ {
+ //no more decay
+ pState->envScale = 512;
+ pState->envState = PCM_ENV_END;
+ }
+ }
+ else if (pState->envState == PCM_ENV_END)
+ {
+ //printf("env end\n");
+ /* set state to stopping, already ramped down */
+ pState->state = EAS_STATE_STOPPING;
+ }
+
+ //pState->gainLeft = (EAS_U16)((pState->gainLeft * (pState->envValue>>7))>>15);
+ //pState->gainRight = (EAS_U16)((pState->gainRight * (pState->envValue>>7))>>15);
+
+ /* gain to 32-bits to increase resolution on anti-zipper filter */
+ /*lint -e{703} use shift for performance */
+ gainLeft = (EAS_I32) pState->currentGainLeft << SYNTH_UPDATE_PERIOD_IN_BITS;
+#if (NUM_OUTPUT_CHANNELS == 2)
+ /*lint -e{703} use shift for performance */
+ gainRight = (EAS_I32) pState->currentGainRight << SYNTH_UPDATE_PERIOD_IN_BITS;
+#endif
+
+ /* calculate a new gain increment, gain target is zero if pausing */
+ if ((pState->state == EAS_STATE_PAUSING) || (pState->state == EAS_STATE_PAUSED))
+ {
+ gainIncLeft = -pState->currentGainLeft;
+#if (NUM_OUTPUT_CHANNELS == 2)
+ gainIncRight= -pState->currentGainRight;
+#endif
+ }
+ else
+ {
+ EAS_I32 gain = FMUL_15x15(pState->envValue >> 7, pState->volume);
+ gainIncLeft = FMUL_15x15(pState->gainLeft, gain) - pState->currentGainLeft;
+#if (NUM_OUTPUT_CHANNELS == 2)
+ gainIncRight = FMUL_15x15(pState->gainRight, gain) - pState->currentGainRight;
+#endif
+ }
+
+ /* calculate phase increment */
+ phaseInc = pState->basefreq;
+
+ /* convert pitch cents to linear multiplier */
+ if (pState->pitch)
+ {
+ temp = EAS_Calculate2toX(pState->pitch);
+ phaseInc = FMUL_15x15(phaseInc, temp);
+ }
+ phaseInc = phaseInc << pState->rateShift;
+
+ /* pointer to mix buffer */
+ pOut = pEASData->pMixBuffer;
+
+ /* render a buffer of samples */
+ while (numSamples--)
+ {
+
+ /* interpolate an output sample */
+ pState->decoderL.output = pState->decoderL.x0 + FMUL_15x15((pState->decoderL.x1 - pState->decoderL.x0), pState->phase & PHASE_FRAC_MASK);
+
+ /* stereo output */
+#if (NUM_OUTPUT_CHANNELS == 2)
+
+ /* stereo stream? */
+ if (pState->flags & PCM_FLAGS_STEREO)
+ pState->decoderR.output = pState->decoderR.x0 + FMUL_15x15((pState->decoderR.x1 - pState->decoderR.x0), pState->phase & PHASE_FRAC_MASK);
+
+ /* gain scale and mix */
+ /*lint -e{704} use shift instead of division */
+ *pOut++ += (pState->decoderL.output * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS;
+ gainLeft += gainIncLeft;
+
+ /*lint -e{704} use shift instead of division */
+ if (pState->flags & PCM_FLAGS_STEREO)
+ *pOut++ += (pState->decoderR.output * (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS;
+ else
+ *pOut++ += (pState->decoderL.output * (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS;
+
+ gainRight += gainIncRight;
+
+ /* mono output */
+#else
+ /* if stereo stream, decode right channel and mix to mono */
+ if (pState->flags & PCM_FLAGS_STEREO)
+ {
+ pState->decoderR.output= pState->decoderR.x0 + FMUL_15x15((pState->decoderR.x1 - pState->decoderR.x0), pState->phase & PHASE_FRAC_MASK);
+
+ /* for mono, sum stereo ADPCM to mono */
+ /*lint -e{704} use shift instead of division */
+ *pOut++ += ((pState->decoderL.output + pState->decoderR.output) * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS;
+ }
+ else
+ /*lint -e{704} use shift instead of division */
+ *pOut++ += (pState->decoderL.output * (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS)) >> PCM_MIXER_GUARD_BITS;
+
+ gainLeft += gainIncLeft;
+#endif
+
+ /* advance phase accumulator */
+ pState->phase += phaseInc;
+
+ /* if integer part of phase accumulator is non-zero, advance to next sample */
+ while (pState->phase & ~PHASE_FRAC_MASK)
+ {
+ pState->decoderL.x0 = pState->decoderL.x1;
+ pState->decoderR.x0 = pState->decoderR.x1;
+
+ /* give the source a chance to continue the stream */
+ if (!pState->bytesLeft && pState->pCallback && ((pState->flags & PCM_FLAGS_EMPTY) == 0))
+ {
+ pState->flags |= PCM_FLAGS_EMPTY;
+ (*pState->pCallback)(pEASData, pState->cbInstData, pState, EAS_STATE_EMPTY);
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "RenderPCMStream: After empty callback, bytesLeft = %d\n", pState->bytesLeft); */ }
+ }
+
+ /* decode the next sample */
+ if ((result = (*pState->pDecoder->pfDecodeSample)(pEASData, pState)) != EAS_SUCCESS)
+ return result;
+
+ /* adjust phase by one sample */
+ pState->phase -= (1L << NUM_PHASE_FRAC_BITS);
+ }
+
+ }
+
+ /* save new gain */
+ /*lint -e{704} use shift instead of division */
+ pState->currentGainLeft = (EAS_I16) (gainLeft >> SYNTH_UPDATE_PERIOD_IN_BITS);
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ /*lint -e{704} use shift instead of division */
+ pState->currentGainRight = (EAS_I16) (gainRight >> SYNTH_UPDATE_PERIOD_IN_BITS);
+#endif
+
+ /* if pausing, set new state and notify */
+ if (pState->state == EAS_STATE_PAUSING)
+ {
+ pState->state = EAS_STATE_PAUSED;
+ if (pState->pCallback)
+ (*pState->pCallback)(pEASData, pState->cbInstData, pState, pState->state);
+ }
+
+ /* if out of data, set stopped state and notify */
+ if (pState->bytesLeft == 0 || pState->state == EAS_STATE_STOPPING)
+ {
+ pState->state = EAS_STATE_STOPPED;
+
+ /* do callback unless the file has already been closed */
+ if (pState->pCallback && pState->fileHandle)
+ (*pState->pCallback)(pEASData, pState->cbInstData, pState, pState->state);
+ }
+
+ if (pState->state == EAS_STATE_READY)
+ pState->state = EAS_STATE_PLAY;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * LinearPCMDecode()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Decodes a PCM sample
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT LinearPCMDecode (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState)
+{
+ EAS_RESULT result;
+ EAS_HW_DATA_HANDLE hwInstData;
+
+ hwInstData = ((S_EAS_DATA*) pEASData)->hwInstData;
+
+ /* if out of data, check for loop */
+ if ((pState->bytesLeft == 0) && (pState->loopSamples != 0))
+ {
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, (EAS_I32) (pState->startPos + pState->loopLocation))) != EAS_SUCCESS)
+ return result;
+ pState->bytesLeft = pState->byteCount = (EAS_I32) pState->bytesLeftLoop;
+ pState->flags &= ~PCM_FLAGS_EMPTY;
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "LinearPCMDecode: Rewind file to %d, bytesLeft = %d\n", pState->startPos, pState->bytesLeft); */ }
+ }
+
+ if (pState->bytesLeft)
+ {
+
+ /* check format byte for 8-bit samples */
+ if (pState->flags & PCM_FLAGS_8_BIT)
+ {
+ /* fetch left or mono sample */
+ if ((result = EAS_HWGetByte(hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS)
+ return result;
+
+ /* if unsigned */
+ if (pState->flags & PCM_FLAGS_UNSIGNED)
+ {
+ /*lint -e{734} converting unsigned 8-bit to signed 16-bit */
+ pState->decoderL.x1 = (EAS_PCM)(((EAS_PCM) pState->srcByte << 8) ^ 0x8000);
+ }
+ else
+ {
+ /*lint -e{734} converting signed 8-bit to signed 16-bit */
+ pState->decoderL.x1 = (EAS_PCM)((EAS_PCM) pState->srcByte << 8);
+ }
+ pState->bytesLeft--;
+
+ /* fetch right sample */
+ if(pState->flags & PCM_FLAGS_STEREO)
+ {
+ if ((result = EAS_HWGetByte(hwInstData, pState->fileHandle, &pState->srcByte)) != EAS_SUCCESS)
+ return result;
+
+ /* if unsigned */
+ if (pState->flags & PCM_FLAGS_UNSIGNED)
+ {
+ /*lint -e{734} converting unsigned 8-bit to signed 16-bit */
+ pState->decoderR.x1 = (EAS_PCM)(((EAS_PCM) pState->srcByte << 8) ^ 0x8000);
+ }
+ else
+ {
+ /*lint -e{734} converting signed 8-bit to signed 16-bit */
+ pState->decoderR.x1 = (EAS_PCM)((EAS_PCM) pState->srcByte << 8);
+ }
+ pState->bytesLeft--;
+ }
+ }
+
+ /* must be 16-bit samples */
+ else
+ {
+ //unsigned 16 bit currently not supported
+ if (pState->flags & PCM_FLAGS_UNSIGNED)
+ {
+ return EAS_ERROR_INVALID_PCM_TYPE;
+ }
+
+ /* fetch left or mono sample */
+ if ((result = EAS_HWGetWord(hwInstData, pState->fileHandle, &pState->decoderL.x1, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+ pState->bytesLeft -= 2;
+
+ /* fetch right sample */
+ if(pState->flags & PCM_FLAGS_STEREO)
+ {
+ if ((result = EAS_HWGetWord(hwInstData, pState->fileHandle, &pState->decoderR.x1, EAS_FALSE)) != EAS_SUCCESS)
+ return result;
+ pState->bytesLeft -= 2;
+ }
+ }
+ }
+
+ /* no more data, force zero samples */
+ else
+ pState->decoderL.x1 = pState->decoderR.x1 = 0;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * LinearPCMLocate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Locate in a linear PCM stream
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT LinearPCMLocate (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time)
+{
+ EAS_RESULT result;
+ EAS_I32 temp;
+ EAS_I32 secs, msecs;
+ EAS_INT shift;
+
+ /* calculate size of sample frame */
+ if (pState->flags & PCM_FLAGS_8_BIT)
+ shift = 0;
+ else
+ shift = 1;
+ if (pState->flags & PCM_FLAGS_STEREO)
+ shift++;
+
+ /* break down into secs and msecs */
+ secs = time / 1000;
+ msecs = time - (secs * 1000);
+
+ /* calculate sample number fraction from msecs */
+ temp = (msecs * pState->sampleRate);
+ temp = (temp >> 10) + ((temp * 49) >> 21);
+
+ /* add integer sample count */
+ temp += secs * pState->sampleRate;
+
+ /* calculate the position based on sample frame size */
+ /*lint -e{703} use shift for performance */
+ temp <<= shift;
+
+ /* past end of sample? */
+ if (temp > (EAS_I32) pState->loopStart)
+ {
+ /* if not looped, flag error */
+ if (pState->loopSamples == 0)
+ {
+ pState->bytesLeft = 0;
+ pState->flags |= PCM_FLAGS_EMPTY;
+ return EAS_ERROR_LOCATE_BEYOND_END;
+ }
+
+ /* looped sample - calculate position in loop */
+ while (temp > (EAS_I32) pState->loopStart)
+ temp -= (EAS_I32) pState->loopStart;
+ }
+
+ /* seek to new position */
+ if ((result = EAS_PESeek(pEASData, pState, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* reset state */
+ if ((pState->state != EAS_STATE_PAUSING) && (pState->state != EAS_STATE_PAUSED))
+ pState->state = EAS_STATE_READY;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_PESeek
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Locate to a particular byte in a PCM stream
+ *----------------------------------------------------------------------------
+ * This bit is tricky because the chunks may not be contiguous,
+ * so we have to rely on the parser to position in the file. We
+ * do this by seeking to the end of each chunk and simulating an
+ * empty buffer condition until we get to where we want to go.
+ *
+ * A better solution would be a parser API for re-positioning,
+ * but there isn't time at the moment to re-factor all the
+ * parsers to support a new API.
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PESeek (S_EAS_DATA *pEASData, S_PCM_STATE *pState, EAS_I32 *pLocation)
+{
+ EAS_RESULT result;
+
+ /* seek to start of audio */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pState->fileHandle, pState->startPos)) != EAS_SUCCESS)
+ {
+ pState->state = EAS_STATE_ERROR;
+ return result;
+ }
+ pState->bytesLeft = pState->bytesLeftLoop;
+
+ /* skip through chunks until we find the right chunk */
+ while (*pLocation > (EAS_I32) pState->bytesLeft)
+ {
+ /* seek to end of audio chunk */
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: Seek to offset = %d\n", pState->bytesLeft); */ }
+ if ((result = EAS_HWFileSeekOfs(pEASData->hwInstData, pState->fileHandle, pState->bytesLeft)) != EAS_SUCCESS)
+ {
+ pState->state = EAS_STATE_ERROR;
+ return result;
+ }
+ *pLocation -= pState->bytesLeft;
+ pState->bytesLeft = 0;
+ pState->flags |= PCM_FLAGS_EMPTY;
+
+ /* retrieve more data */
+ if (pState->pCallback)
+ (*pState->pCallback)(pEASData, pState->cbInstData, pState, EAS_STATE_EMPTY);
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: bytesLeft=%d, byte location = %d\n", pState->bytesLeft, *pLocation); */ }
+
+ /* no more samples */
+ if (pState->bytesLeft == 0)
+ return EAS_ERROR_LOCATE_BEYOND_END;
+ }
+
+ /* seek to new offset in current chunk */
+ if (*pLocation > 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "EAS_PESeek: Seek to offset = %d\n", *pLocation); */ }
+ if ((result = EAS_HWFileSeekOfs(pEASData->hwInstData, pState->fileHandle, *pLocation)) != EAS_SUCCESS)
+ {
+ pState->state = EAS_STATE_ERROR;
+ return result;
+ }
+
+ /* if not streamed, calculate number of bytes left */
+ if (pState->flags & PCM_FLAGS_STREAMING)
+ pState->bytesLeft = 0x7fffffff;
+ else
+ pState->bytesLeft -= *pLocation;
+ }
+ return EAS_SUCCESS;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_pcm.h b/arm-wt-22k/lib_src/eas_pcm.h new file mode 100644 index 0000000..c161757 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pcm.h @@ -0,0 +1,359 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcm.h
+ *
+ * Contents and purpose:
+ * External function prototypes for eas_pcm.c module
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 847 $
+ * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PCM_H
+#define _EAS_PCM_H
+
+/* default gain setting - roughly unity gain */
+#define PCM_DEFAULT_GAIN_SETTING 0x6000
+
+typedef struct s_pcm_state_tag *EAS_PCM_HANDLE;
+typedef void (*EAS_PCM_CALLBACK) (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR cbInstData, EAS_PCM_HANDLE pcmHandle, EAS_STATE state);
+
+/* parameters for EAS_PEOpenStream */
+typedef struct s_pcm_open_params_tag
+{
+ EAS_FILE_HANDLE fileHandle;
+ EAS_I32 decoder;
+ EAS_U32 sampleRate;
+ EAS_I32 size;
+ EAS_U32 loopStart;
+ EAS_U32 loopSamples;
+ EAS_I32 blockSize;
+ EAS_U32 flags;
+ EAS_U32 envData;
+ EAS_I16 volume;
+ EAS_PCM_CALLBACK pCallbackFunc;
+ EAS_VOID_PTR cbInstData;
+ } S_PCM_OPEN_PARAMS;
+
+/*----------------------------------------------------------------------------
+ * EAS_PEInit()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEInit (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Shuts down the PCM engine
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEShutdown (EAS_DATA_HANDLE pEASData);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEOpenStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Starts up a PCM playback
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEOpenStream (EAS_DATA_HANDLE pEASData, S_PCM_OPEN_PARAMS *pParams, EAS_PCM_HANDLE *pHandle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEContinueStream()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Continues a PCM stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEContinueStream (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_I32 size);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEGetFileHandle()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the file handle of a stream
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEGetFileHandle (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_FILE_HANDLE *pFileHandle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PERender()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Render a buffer of PCM audio
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PERender (EAS_DATA_HANDLE pEASData, EAS_I32 numSamples);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateParams()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch and volume parameters using MIDI controls
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEUpdateParams (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch, EAS_I16 gainLeft, EAS_I16 gainRight);
+
+/*----------------------------------------------------------------------------
+ * EAS_PELocate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This function seeks to the requested place in the file. Accuracy
+ * is dependent on the sample rate and block size.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pState - stream handle
+ * time - media time in milliseconds
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PELocate (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I32 time);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdateVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the volume parameters for a PCM stream
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ * gainLeft - linear gain multipler in 1.15 fraction format
+ * gainRight - linear gain multipler in 1.15 fraction format
+ * initial - initial settings, set current gain
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes
+ * In mono mode, leftGain controls the output gain and rightGain is ignored
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdateVolume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 volume);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEUpdatePitch()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the pitch parameter for a PCM stream
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * pState - pointer to S_PCM_STATE for this stream
+ * pitch - new pitch value in pitch cents
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+EAS_RESULT EAS_PEUpdatePitch (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE pState, EAS_I16 pitch);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * This interface is also exposed in the internal library for use by the other modules.
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEState (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle, EAS_STATE *pState);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEClose()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEClose (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEReset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEReset (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEPause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mute and pause rendering a PCM stream. Sets the gain target to zero and stops the playback
+ * at the end of the next audio frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEPause (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PEResume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume rendering a PCM stream. Sets the gain target back to its
+ * previous setting and restarts playback at the end of the next audio
+ * frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PEResume (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+/*----------------------------------------------------------------------------
+ * EAS_PERelease()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Put the PCM stream envelope into release.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_PCM_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PERelease (EAS_DATA_HANDLE pEASData, EAS_PCM_HANDLE handle);
+
+#endif /* end _EAS_PCM_H */
+
diff --git a/arm-wt-22k/lib_src/eas_pcmdata.c b/arm-wt-22k/lib_src/eas_pcmdata.c new file mode 100644 index 0000000..5649f07 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pcmdata.c @@ -0,0 +1,35 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcmdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data for the PCM engine.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+
+/* static data allocation */
+S_PCM_STATE eas_PCMData[MAX_PCM_STREAMS];
+
+
diff --git a/arm-wt-22k/lib_src/eas_pcmdata.h b/arm-wt-22k/lib_src/eas_pcmdata.h new file mode 100644 index 0000000..be2f8e5 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_pcmdata.h @@ -0,0 +1,157 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_pcmdata.h
+ *
+ * Contents and purpose:
+ * Data declarations for the PCM engine
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 847 $
+ * $Date: 2007-08-27 21:30:08 -0700 (Mon, 27 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_PCMDATA_H
+#define _EAS_PCMDATA_H
+
+/* sets the maximum number of simultaneous PCM streams */
+#ifndef MAX_PCM_STREAMS
+#define MAX_PCM_STREAMS 16
+#define PCM_STREAM_THRESHOLD (MAX_PCM_STREAMS - 4)
+#endif
+
+/* coefficents for high-pass filter in ADPCM */
+#define INTEGRATOR_COEFFICIENT 100 /* coefficient for leaky integrator */
+
+/* additional flags in S_PCM_STATE.flags used internal to module */
+#define PCM_FLAGS_EMPTY 0x01000000 /* unsigned format */
+
+/*----------------------------------------------------------------------------
+ * S_PCM_STATE
+ *
+ * Retains state information for PCM streams.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_decoder_state_tag
+{
+ EAS_I32 output; /* last output for DC offset filter */
+ EAS_I32 acc; /* accumulator for DC offset filter */
+ EAS_I32 step; /* current ADPCM step size */
+ EAS_PCM x1; /* current generated sample */
+ EAS_PCM x0; /* previous generated sample */
+} S_DECODER_STATE;
+
+typedef enum
+{
+ PCM_ENV_START = 0,
+ PCM_ENV_ATTACK,
+ PCM_ENV_DECAY,
+ PCM_ENV_SUSTAIN,
+ PCM_ENV_RELEASE,
+ PCM_ENV_END
+} E_PCM_ENV_STATE;
+
+typedef struct s_pcm_state_tag
+{
+#ifdef _CHECKED_BUILD
+ EAS_U32 handleCheck; /* signature check for checked build */
+#endif
+ EAS_FILE_HANDLE fileHandle; /* pointer to input file */
+ EAS_PCM_CALLBACK pCallback; /* pointer to callback function */
+ EAS_VOID_PTR cbInstData; /* instance data for callback function */
+ struct s_decoder_interface_tag EAS_CONST * pDecoder; /* pointer to decoder interface */
+ EAS_STATE state; /* stream state */
+ EAS_I32 time; /* media time */
+ EAS_I32 startPos; /* start of PCM stream */
+ EAS_I32 loopLocation; /* file location where loop starts */
+ EAS_I32 byteCount; /* size of file */
+ EAS_U32 loopStart; /* loop start, offset in samples from startPos */
+ /* NOTE: For CMF, we use this to store total sample size */
+ EAS_U32 loopSamples; /* total loop length, in samples, 0 means no loop */
+ /* NOTE: For CMF, non-zero means looped */
+ EAS_U32 samplesInLoop; /* samples left in the loop to play back */
+ EAS_I32 samplesTilLoop; /* samples left to play until top of loop */
+ EAS_I32 bytesLeft; /* count of bytes left in stream */
+ EAS_I32 bytesLeftLoop; /* count of bytes left in stream, value at start of loop */
+ EAS_U32 phase; /* current phase for interpolator */
+ EAS_U32 basefreq; /* frequency multiplier */
+ EAS_U32 flags; /* stream flags */
+ EAS_U32 envData; /* envelope data (and LFO data) */
+ EAS_U32 envValue; /* current envelope value */
+ EAS_U32 envScale; /* current envelope scale */
+ EAS_U32 startOrder; /* start order index, first is 0, next is 1, etc. */
+ S_DECODER_STATE decoderL; /* left (mono) ADPCM state */
+ S_DECODER_STATE decoderR; /* right ADPCM state */
+ S_DECODER_STATE decoderLLoop; /* left (mono) ADPCM state, value at start of loop */
+ S_DECODER_STATE decoderRLoop; /* right ADPCM state, value at start of loop */
+ E_PCM_ENV_STATE envState; /* current envelope state */
+ EAS_I16 volume; /* volume for stream */
+ EAS_I16 pitch; /* relative pitch in cents - zero is unity playback */
+ EAS_I16 gainLeft; /* requested gain */
+ EAS_I16 gainRight; /* requested gain */
+ EAS_I16 currentGainLeft; /* current gain for anti-zipper filter */
+ EAS_I16 currentGainRight; /* current gain for anti-zipper filter */
+ EAS_U16 blockSize; /* block size for ADPCM decoder */
+ EAS_U16 blockCount; /* block counter for ADPCM decoder */
+ EAS_U16 sampleRate; /* input sample rate */
+ EAS_U8 srcByte; /* source byte */
+ EAS_U8 msBitCount; /* count keeps track of MS bits */
+ EAS_U8 msBitMask; /* mask keeps track of MS bits */
+ EAS_U8 msBitValue; /* value keeps track of MS bits */
+ EAS_U8 msBitCountLoop; /* count keeps track of MS bits, value at loop start */
+ EAS_U8 msBitMaskLoop; /* mask keeps track of MS bits, value at loop start */
+ EAS_U8 msBitValueLoop; /* value keeps track of MS bits, value at loop start */
+ EAS_BOOL8 hiNibble; /* indicates high/low nibble is next */
+ EAS_BOOL8 hiNibbleLoop; /* indicates high/low nibble is next, value loop start */
+ EAS_U8 rateShift; /* for playback rate greater than 1.0 */
+} S_PCM_STATE;
+
+/*----------------------------------------------------------------------------
+ * S_DECODER_INTERFACE
+ *
+ * Generic interface for audio decoders
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_decoder_interface_tag
+{
+ EAS_RESULT (* EAS_CONST pfInit)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+ EAS_RESULT (* EAS_CONST pfDecodeSample)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState);
+ EAS_RESULT (* EAS_CONST pfLocate)(EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 time);
+} S_DECODER_INTERFACE;
+
+
+/* header chunk for SMAF ADPCM */
+#define TAG_YAMAHA_ADPCM 0x4d776100
+#define TAG_MASK 0xffffff00
+#define TAG_RIFF_FILE 0x52494646
+#define TAG_WAVE_CHUNK 0x57415645
+#define TAG_FMT_CHUNK 0x666d7420
+
+/*----------------------------------------------------------------------------
+ * EAS_PESeek
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Locate to a particular byte in a PCM stream
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_PESeek (EAS_DATA_HANDLE pEASData, S_PCM_STATE *pState, EAS_I32 *pLocation);
+
+#endif /* _EAS_PCMDATA_H */
+
diff --git a/arm-wt-22k/lib_src/eas_public.c b/arm-wt-22k/lib_src/eas_public.c new file mode 100644 index 0000000..d67ef7e --- /dev/null +++ b/arm-wt-22k/lib_src/eas_public.c @@ -0,0 +1,2601 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_public.c + * + * Contents and purpose: + * Contains EAS library public interface + * + * Copyright Sonic Network Inc. 2004 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 842 $ + * $Date: 2007-08-23 14:32:31 -0700 (Thu, 23 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +#include "eas_synthcfg.h" +#include "eas.h" +#include "eas_config.h" +#include "eas_host.h" +#include "eas_report.h" +#include "eas_data.h" +#include "eas_parser.h" +#include "eas_pcm.h" +#include "eas_midi.h" +#include "eas_mixer.h" +#include "eas_build.h" +#include "eas_vm_protos.h" +#include "eas_math.h" + +#ifdef JET_INTERFACE +#include "jet_data.h" +#endif + +#ifdef DLS_SYNTHESIZER +#include "eas_mdls.h" +#endif + +/* number of events to parse before calling EAS_HWYield function */ +#define YIELD_EVENT_COUNT 10 + +/*---------------------------------------------------------------------------- + * easLibConfig + * + * This structure is available through the EAS public interface to allow + * the user to check the configuration of the library. + *---------------------------------------------------------------------------- +*/ +static const S_EAS_LIB_CONFIG easLibConfig = +{ + LIB_VERSION, +#ifdef _CHECKED_BUILD + EAS_TRUE, +#else + EAS_FALSE, +#endif + MAX_SYNTH_VOICES, + NUM_OUTPUT_CHANNELS, + _OUTPUT_SAMPLE_RATE, + BUFFER_SIZE_IN_MONO_SAMPLES, +#ifdef _FILTER_ENABLED + EAS_TRUE, +#else + EAS_FALSE, +#endif + _BUILD_TIME_, + _BUILD_VERSION_ +}; + +/* local prototypes */ +static EAS_RESULT EAS_ParseEvents (S_EAS_DATA *pEASData, S_EAS_STREAM *pStream, EAS_U32 endTime, EAS_INT parseMode); + +/*---------------------------------------------------------------------------- + * EAS_SetStreamParameter + *---------------------------------------------------------------------------- + * Sets the specified parameter in the stream. Allows access to + * customizable settings within the individual file parsers. + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * pStream - stream handle + * param - enumerated parameter (see eas_parser.h) + * value - new value + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_SetStreamParameter (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 param, EAS_I32 value) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule->pfSetData) + return (*pParserModule->pfSetData)(pEASData, pStream->handle, param, value); + return EAS_ERROR_FEATURE_NOT_AVAILABLE; +} + +/*---------------------------------------------------------------------------- + * EAS_GetStreamParameter + *---------------------------------------------------------------------------- + * Sets the specified parameter in the stream. Allows access to + * customizable settings within the individual file parsers. + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * pStream - stream handle + * param - enumerated parameter (see eas_parser.h) + * pValue - pointer to variable to receive current setting + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_GetStreamParameter (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 param, EAS_I32 *pValue) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule->pfGetData) + return (*pParserModule->pfGetData)(pEASData, pStream->handle, param, pValue); + return EAS_ERROR_FEATURE_NOT_AVAILABLE; +} + +/*---------------------------------------------------------------------------- + * EAS_StreamReady() + *---------------------------------------------------------------------------- + * This routine sets common parameters like transpose, volume, etc. + * First, it attempts to use the parser EAS_SetStreamParameter interface. If that + * fails, it attempts to get the synth handle from the parser and + * set the parameter directly on the synth. This eliminates duplicate + * code in the parser. + *---------------------------------------------------------------------------- +*/ +EAS_BOOL EAS_StreamReady (S_EAS_DATA *pEASData, EAS_HANDLE pStream) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_STATE state; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule->pfState(pEASData, pStream->handle, &state) != EAS_SUCCESS) + return EAS_FALSE; + return (state < EAS_STATE_OPEN); +} + +/*---------------------------------------------------------------------------- + * EAS_IntSetStrmParam() + *---------------------------------------------------------------------------- + * This routine sets common parameters like transpose, volume, etc. + * First, it attempts to use the parser EAS_SetStreamParameter interface. If that + * fails, it attempts to get the synth handle from the parser and + * set the parameter directly on the synth. This eliminates duplicate + * code in the parser. + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_IntSetStrmParam (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_INT param, EAS_I32 value) +{ + S_SYNTH *pSynth; + + /* try to set the parameter using stream interface */ + if (EAS_SetStreamParameter(pEASData, pStream, param, value) == EAS_SUCCESS) + return EAS_SUCCESS; + + /* get a pointer to the synth object and set it directly */ + /*lint -e{740} we are cheating by passing a pointer through this interface */ + if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) + return EAS_ERROR_INVALID_PARAMETER; + + if (pSynth == NULL) + return EAS_ERROR_INVALID_PARAMETER; + + switch (param) + { + +#ifdef DLS_SYNTHESIZER + case PARSER_DATA_DLS_COLLECTION: + { + EAS_RESULT result = VMSetDLSLib(pSynth, (EAS_DLSLIB_HANDLE) value); + if (result == EAS_SUCCESS) + { + DLSAddRef((S_DLS*) value); + VMInitializeAllChannels(pEASData->pVoiceMgr, pSynth); + } + return result; + } +#endif + + case PARSER_DATA_EAS_LIBRARY: + return VMSetEASLib(pSynth, (EAS_SNDLIB_HANDLE) value); + + case PARSER_DATA_POLYPHONY: + return VMSetPolyphony(pEASData->pVoiceMgr, pSynth, value); + + case PARSER_DATA_PRIORITY: + return VMSetPriority(pEASData->pVoiceMgr, pSynth, value); + + case PARSER_DATA_TRANSPOSITION: + VMSetTranposition(pSynth, value); + break; + + case PARSER_DATA_VOLUME: + VMSetVolume(pSynth, (EAS_U16) value); + break; + + default: + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Invalid paramter %d in call to EAS_IntSetStrmParam", param); */ } + return EAS_ERROR_INVALID_PARAMETER; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_IntGetStrmParam() + *---------------------------------------------------------------------------- + * This routine gets common parameters like transpose, volume, etc. + * First, it attempts to use the parser EAS_GetStreamParameter interface. If that + * fails, it attempts to get the synth handle from the parser and + * get the parameter directly on the synth. + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_IntGetStrmParam (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_INT param, EAS_I32 *pValue) +{ + S_SYNTH *pSynth; + + /* try to set the parameter */ + if (EAS_GetStreamParameter(pEASData, pStream, param, pValue) == EAS_SUCCESS) + return EAS_SUCCESS; + + /* get a pointer to the synth object and retrieve data directly */ + /*lint -e{740} we are cheating by passing a pointer through this interface */ + if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) + return EAS_ERROR_INVALID_PARAMETER; + + if (pSynth == NULL) + return EAS_ERROR_INVALID_PARAMETER; + + switch (param) + { + case PARSER_DATA_POLYPHONY: + return VMGetPolyphony(pEASData->pVoiceMgr, pSynth, pValue); + + case PARSER_DATA_PRIORITY: + return VMGetPriority(pEASData->pVoiceMgr, pSynth, pValue); + + case PARSER_DATA_TRANSPOSITION: + VMGetTranposition(pSynth, pValue); + break; + + case PARSER_DATA_NOTE_COUNT: + *pValue = VMGetNoteCount(pSynth); + break; + + default: + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Invalid paramter %d in call to EAS_IntSetStrmParam", param); */ } + return EAS_ERROR_INVALID_PARAMETER; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_AllocateStream() + *---------------------------------------------------------------------------- + * Purpose: + * Allocates a stream handle + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +static EAS_INT EAS_AllocateStream (EAS_DATA_HANDLE pEASData) +{ + EAS_INT streamNum; + + /* check for static allocation, only one stream allowed */ + if (pEASData->staticMemoryModel) + { + if (pEASData->streams[0].handle != NULL) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Attempt to open multiple streams in static model\n"); */ } + return -1; + } + return 0; + } + + /* dynamic model */ + for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) + if (pEASData->streams[streamNum].handle == NULL) + break; + if (streamNum == MAX_NUMBER_STREAMS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Exceeded maximum number of open streams\n"); */ } + return -1; + } + return streamNum; +} + +/*---------------------------------------------------------------------------- + * EAS_InitStream() + *---------------------------------------------------------------------------- + * Purpose: + * Initialize a stream + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +static void EAS_InitStream (S_EAS_STREAM *pStream, EAS_VOID_PTR pParserModule, EAS_VOID_PTR streamHandle) +{ + pStream->pParserModule = pParserModule; + pStream->handle = streamHandle; + pStream->time = 0; + pStream->frameLength = AUDIO_FRAME_LENGTH; + pStream->repeatCount = 0; + pStream->volume = DEFAULT_STREAM_VOLUME; + pStream->streamFlags = 0; +} + +/*---------------------------------------------------------------------------- + * EAS_Config() + *---------------------------------------------------------------------------- + * Purpose: + * Returns a pointer to a structure containing the configuration options + * in this library build. + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC const S_EAS_LIB_CONFIG *EAS_Config (void) +{ + return &easLibConfig; +} + +/*---------------------------------------------------------------------------- + * EAS_Init() + *---------------------------------------------------------------------------- + * Purpose: + * Initialize the synthesizer library + * + * Inputs: + * ppEASData - pointer to data handle variable for this instance + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Init (EAS_DATA_HANDLE *ppEASData) +{ + EAS_HW_DATA_HANDLE pHWInstData; + EAS_RESULT result; + S_EAS_DATA *pEASData; + EAS_INT module; + EAS_BOOL staticMemoryModel; + + /* get the memory model */ + staticMemoryModel = EAS_CMStaticMemoryModel(); + + /* initialize the host wrapper interface */ + *ppEASData = NULL; + if ((result = EAS_HWInit(&pHWInstData)) != EAS_SUCCESS) + return result; + + /* check Configuration Module for S_EAS_DATA allocation */ + if (staticMemoryModel) + pEASData = EAS_CMEnumData(EAS_CM_EAS_DATA); + else + pEASData = EAS_HWMalloc(pHWInstData, sizeof(S_EAS_DATA)); + if (!pEASData) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate EAS library memory\n"); */ } + return EAS_ERROR_MALLOC_FAILED; + } + + /* initialize some data */ + EAS_HWMemSet(pEASData, 0, sizeof(S_EAS_DATA)); + pEASData->staticMemoryModel = (EAS_BOOL8) staticMemoryModel; + pEASData->hwInstData = pHWInstData; + pEASData->renderTime = 0; + + /* set header search flag */ +#ifdef FILE_HEADER_SEARCH + pEASData->searchHeaderFlag = EAS_TRUE; +#endif + + /* initalize parameters */ + EAS_SetVolume(pEASData, NULL, DEFAULT_VOLUME); + +#ifdef _METRICS_ENABLED + /* initalize the metrics module */ + pEASData->pMetricsModule = EAS_CMEnumOptModules(EAS_MODULE_METRICS); + if (pEASData->pMetricsModule != NULL) + { + if ((result = (*pEASData->pMetricsModule->pfInit)(pEASData, &pEASData->pMetricsData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld initializing metrics module\n", result); */ } + return result; + } + } +#endif + + /* initailize the voice manager & synthesizer */ + if ((result = VMInitialize(pEASData)) != EAS_SUCCESS) + return result; + + /* initialize mix engine */ + if ((result = EAS_MixEngineInit(pEASData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld starting up mix engine\n", result); */ } + return result; + } + + /* initialize effects modules */ + for (module = 0; module < NUM_EFFECTS_MODULES; module++) + { + pEASData->effectsModules[module].effect = EAS_CMEnumFXModules(module); + if (pEASData->effectsModules[module].effect != NULL) + { + if ((result = (*pEASData->effectsModules[module].effect->pfInit)(pEASData, &pEASData->effectsModules[module].effectData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Initialization of effects module %d returned %d\n", module, result); */ } + return result; + } + } + } + + /* initialize PCM engine */ + if ((result = EAS_PEInit(pEASData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "EAS_PEInit failed with error code %ld\n", result); */ } + return result; + } + + /* return instance data pointer to host */ + *ppEASData = pEASData; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_Shutdown() + *---------------------------------------------------------------------------- + * Purpose: + * Shuts down the library. Deallocates any memory associated with the + * synthesizer (dynamic memory model only) + * + * Inputs: + * pEASData - handle to data for this instance + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Shutdown (EAS_DATA_HANDLE pEASData) +{ + EAS_HW_DATA_HANDLE hwInstData; + EAS_RESULT result, reportResult; + EAS_INT i; + + /* establish pointers */ + hwInstData = pEASData->hwInstData; + + /* check for NULL handle */ + if (!pEASData) + return EAS_ERROR_HANDLE_INTEGRITY; + + /* if there are streams open, close them */ + reportResult = EAS_SUCCESS; + for (i = 0; i < MAX_NUMBER_STREAMS; i++) + { + if (pEASData->streams[i].pParserModule && pEASData->streams[i].handle) + { + if ((result = (*((S_FILE_PARSER_INTERFACE*)(pEASData->streams[i].pParserModule))->pfClose)(pEASData, pEASData->streams[i].handle)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down parser module\n", result); */ } + reportResult = result; + } + } + } + + /* shutdown PCM engine */ + if ((result = EAS_PEShutdown(pEASData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down PCM engine\n", result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + + /* shutdown mix engine */ + if ((result = EAS_MixEngineShutdown(pEASData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down mix engine\n", result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + + /* shutdown effects modules */ + for (i = 0; i < NUM_EFFECTS_MODULES; i++) + { + if (pEASData->effectsModules[i].effect) + { + if ((result = (*pEASData->effectsModules[i].effect->pfShutdown)(pEASData, pEASData->effectsModules[i].effectData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Shutdown of effects module %d returned %d\n", i, result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + } + } + + /* shutdown the voice manager & synthesizer */ + VMShutdown(pEASData); + +#ifdef _METRICS_ENABLED + /* shutdown the metrics module */ + if (pEASData->pMetricsModule != NULL) + { + if ((result = (*pEASData->pMetricsModule->pfShutdown)(pEASData, pEASData->pMetricsData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down metrics module\n", result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + } +#endif + + /* release allocated memory */ + if (!pEASData->staticMemoryModel) + EAS_HWFree(hwInstData, pEASData); + + /* shutdown host wrappers */ + if (hwInstData) + { + if ((result = EAS_HWShutdown(hwInstData)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Error %ld shutting down host wrappers\n", result); */ } + if (reportResult == EAS_SUCCESS) + reportResult = result; + } + } + + return reportResult; +} + +#ifdef JET_INTERFACE +/*---------------------------------------------------------------------------- + * EAS_OpenJETStream() + *---------------------------------------------------------------------------- + * Private interface for JET to open an SMF stream with an offset + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_OpenJETStream (EAS_DATA_HANDLE pEASData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, EAS_HANDLE *ppStream) +{ + EAS_RESULT result; + EAS_VOID_PTR streamHandle; + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_INT streamNum; + + /* allocate a stream */ + if ((streamNum = EAS_AllocateStream(pEASData)) < 0) + return EAS_ERROR_MAX_STREAMS_OPEN; + + /* check Configuration Module for SMF parser */ + *ppStream = NULL; + streamHandle = NULL; + pParserModule = (S_FILE_PARSER_INTERFACE *) EAS_CMEnumModules(0); + if (pParserModule == NULL) + return EAS_ERROR_UNRECOGNIZED_FORMAT; + + /* see if SMF parser recognizes the file */ + if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, offset)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } + return result; + } + + /* parser recognized the file, return the handle */ + if (streamHandle) + { + EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); + *ppStream = &pEASData->streams[streamNum]; + return EAS_SUCCESS; + } + + return EAS_ERROR_UNRECOGNIZED_FORMAT; +} +#endif + +/*---------------------------------------------------------------------------- + * EAS_OpenFile() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a file for audio playback. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * pHandle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_OpenFile (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *ppStream) +{ + EAS_RESULT result; + EAS_FILE_HANDLE fileHandle; + EAS_VOID_PTR streamHandle; + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_INT streamNum; + EAS_INT moduleNum; + + /* open the file */ + if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) + return result; + + /* allocate a stream */ + if ((streamNum = EAS_AllocateStream(pEASData)) < 0) + return EAS_ERROR_MAX_STREAMS_OPEN; + + /* check Configuration Module for file parsers */ + pParserModule = NULL; + *ppStream = NULL; + streamHandle = NULL; + for (moduleNum = 0; ; moduleNum++) + { + pParserModule = (S_FILE_PARSER_INTERFACE *) EAS_CMEnumModules(moduleNum); + if (pParserModule == NULL) + break; + + /* see if this parser recognizes it */ + if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, 0L)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } + return result; + } + + /* parser recognized the file, return the handle */ + if (streamHandle) + { + + /* save the parser pointer and file handle */ + EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); + *ppStream = &pEASData->streams[streamNum]; + return EAS_SUCCESS; + } + + /* rewind the file for the next parser */ + if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, 0L)) != EAS_SUCCESS) + return result; + } + + /* no parser was able to recognize the file, close it and return an error */ + EAS_HWCloseFile(pEASData->hwInstData, fileHandle); + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "No parser recognized the requested file\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; +} + +#ifdef MMAPI_SUPPORT +/*---------------------------------------------------------------------------- + * EAS_MMAPIToneControl() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a ToneControl file for audio playback. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * pHandle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MMAPIToneControl (EAS_DATA_HANDLE pEASData, EAS_FILE_LOCATOR locator, EAS_HANDLE *ppStream) +{ + EAS_RESULT result; + EAS_FILE_HANDLE fileHandle; + EAS_VOID_PTR streamHandle; + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_INT streamNum; + + /* check if the tone control parser is available */ + *ppStream = NULL; + streamHandle = NULL; + pParserModule = EAS_CMEnumOptModules(EAS_MODULE_MMAPI_TONE_CONTROL); + if (pParserModule == NULL) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_MMAPIToneControl: ToneControl parser not available\n"); */ } + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + } + + /* open the file */ + if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) + return result; + + /* allocate a stream */ + if ((streamNum = EAS_AllocateStream(pEASData)) < 0) + return EAS_ERROR_MAX_STREAMS_OPEN; + + /* see if ToneControl parser recognizes it */ + if ((result = (*pParserModule->pfCheckFileType)(pEASData, fileHandle, &streamHandle, 0L)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "CheckFileType returned error %ld\n", result); */ } + return result; + } + + /* parser accepted the file, return the handle */ + if (streamHandle) + { + + /* save the parser pointer and file handle */ + EAS_InitStream(&pEASData->streams[streamNum], pParserModule, streamHandle); + *ppStream = &pEASData->streams[streamNum]; + return EAS_SUCCESS; + } + + /* parser did not recognize the file, close it and return an error */ + EAS_HWCloseFile(pEASData->hwInstData, fileHandle); + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "No parser recognized the requested file\n"); */ } + return EAS_ERROR_UNRECOGNIZED_FORMAT; +} + +/*---------------------------------------------------------------------------- + * EAS_GetWaveFmtChunk + *---------------------------------------------------------------------------- + * Helper function to retrieve WAVE file fmt chunk for MMAPI + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * pStream - stream handle + * pFmtChunk - pointer to variable to receive current setting + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetWaveFmtChunk (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_VOID_PTR *ppFmtChunk) +{ + EAS_RESULT result; + EAS_I32 value; + + if ((result = EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_FORMAT, &value)) != EAS_SUCCESS) + return result; + *ppFmtChunk = (EAS_VOID_PTR) value; + return EAS_SUCCESS; +} +#endif + +/*---------------------------------------------------------------------------- + * EAS_GetFileType + *---------------------------------------------------------------------------- + * Returns the file type (see eas_types.h for enumerations) + *---------------------------------------------------------------------------- + * pEASData - pointer to EAS persistent data object + * pStream - stream handle + * pFileType - pointer to variable to receive file type + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetFileType (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_I32 *pFileType) +{ + if (!EAS_StreamReady (pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_FILE_TYPE, pFileType); +} + +/*---------------------------------------------------------------------------- + * EAS_Prepare() + *---------------------------------------------------------------------------- + * Purpose: + * Prepares the synthesizer to play the file or stream. Parses the first + * frame of data from the file and arms the synthesizer. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Prepare (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_STATE state; + EAS_RESULT result; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + /* check for valid state */ + result = pParserModule->pfState(pEASData, pStream->handle, &state); + if (result == EAS_SUCCESS) + { + /* prepare the stream */ + if (state == EAS_STATE_OPEN) + { + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + result = (*pParserModule->pfPrepare)(pEASData, pStream->handle); + + /* set volume */ + if (result == EAS_SUCCESS) + result = EAS_SetVolume(pEASData, pStream, pStream->volume); + } + else + result = EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + } + + return result; +} + +/*---------------------------------------------------------------------------- + * EAS_Render() + *---------------------------------------------------------------------------- + * Purpose: + * Parse the Midi data and render PCM audio data. + * + * Inputs: + * pEASData - buffer for internal EAS data + * pOut - output buffer pointer + * nNumRequested - requested num samples to generate + * pnNumGenerated - actual number of samples generated + * + * Outputs: + * EAS_SUCCESS if PCM data was successfully rendered + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Render (EAS_DATA_HANDLE pEASData, EAS_PCM *pOut, EAS_I32 numRequested, EAS_I32 *pNumGenerated) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + EAS_I32 voicesRendered; + EAS_STATE parserState; + EAS_INT streamNum; + + /* assume no samples generated and reset workload */ + *pNumGenerated = 0; + VMInitWorkload(pEASData->pVoiceMgr); + + /* no support for other buffer sizes yet */ + if (numRequested != BUFFER_SIZE_IN_MONO_SAMPLES) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "This library supports only %ld samples in buffer, host requested %ld samples\n", + (EAS_I32) BUFFER_SIZE_IN_MONO_SAMPLES, numRequested); */ } + return EAS_BUFFER_SIZE_MISMATCH; + } + +#ifdef _METRICS_ENABLED + /* start performance counter */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); +#endif + + /* prep the frame buffer, do mix engine prep only if TRUE */ +#ifdef _SPLIT_ARCHITECTURE + if (VMStartFrame(pEASData)) + EAS_MixEnginePrep(pEASData, numRequested); +#else + /* prep the mix engine */ + EAS_MixEnginePrep(pEASData, numRequested); +#endif + + /* save the output buffer pointer */ + pEASData->pOutputAudioBuffer = pOut; + + +#ifdef _METRICS_ENABLED + /* start performance counter */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_PARSE_TIME); +#endif + + /* if we haven't finished parsing from last time, do it now */ + /* need to parse another frame of events before we render again */ + for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) + { + /* clear the locate flag */ + pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_LOCATE; + + if (pEASData->streams[streamNum].pParserModule) + { + + /* establish pointer to parser module */ + pParserModule = pEASData->streams[streamNum].pParserModule; + + /* handle pause */ + if (pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_PAUSE) + { + if (pParserModule->pfPause) + result = pParserModule->pfPause(pEASData, pEASData->streams[streamNum].handle); + pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_PAUSE; + } + + /* get current state */ + if ((result = (*pParserModule->pfState)(pEASData, pEASData->streams[streamNum].handle, &parserState)) != EAS_SUCCESS) + return result; + + /* handle resume */ + if (parserState == EAS_STATE_PAUSED) + { + if (pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_RESUME) + { + if (pParserModule->pfResume) + result = pParserModule->pfResume(pEASData, pEASData->streams[streamNum].handle); + pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_RESUME; + } + } + + /* if necessary, parse stream */ + if ((pEASData->streams[streamNum].streamFlags & STREAM_FLAGS_PARSED) == 0) + if ((result = EAS_ParseEvents(pEASData, &pEASData->streams[streamNum], pEASData->streams[streamNum].time + pEASData->streams[streamNum].frameLength, eParserModePlay)) != EAS_SUCCESS) + return result; + + /* check for an early abort */ + if ((pEASData->streams[streamNum].streamFlags) == 0) + { + +#ifdef _METRICS_ENABLED + /* stop performance counter */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); +#endif + + return EAS_SUCCESS; + } + + /* check for repeat */ + if (pEASData->streams[streamNum].repeatCount) + { + + /* check for stopped state */ + if ((result = (*pParserModule->pfState)(pEASData, pEASData->streams[streamNum].handle, &parserState)) != EAS_SUCCESS) + return result; + if (parserState == EAS_STATE_STOPPED) + { + + /* decrement repeat count, unless it is negative */ + if (pEASData->streams[streamNum].repeatCount > 0) + pEASData->streams[streamNum].repeatCount--; + + /* reset the parser */ + if ((result = (*pParserModule->pfReset)(pEASData, pEASData->streams[streamNum].handle)) != EAS_SUCCESS) + return result; + pEASData->streams[streamNum].time = 0; + } + } + } + } + +#ifdef _METRICS_ENABLED + /* stop performance counter */ + if (pEASData->pMetricsData) + (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_PARSE_TIME); +#endif + +#ifdef _METRICS_ENABLED + /* start the render timer */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_RENDER_TIME); +#endif + + /* render audio */ + if ((result = VMRender(pEASData->pVoiceMgr, BUFFER_SIZE_IN_MONO_SAMPLES, pEASData->pMixBuffer, &voicesRendered)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "pfRender function returned error %ld\n", result); */ } + return result; + } + +#ifdef _METRICS_ENABLED + /* stop the render timer */ + if (pEASData->pMetricsData) { + (*pEASData->pMetricsModule->pfIncrementCounter)(pEASData->pMetricsData, EAS_PM_FRAME_COUNT, 1); + (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_RENDER_TIME); + (*pEASData->pMetricsModule->pfIncrementCounter)(pEASData->pMetricsData, EAS_PM_TOTAL_VOICE_COUNT, (EAS_U32) voicesRendered); + (void)(*pEASData->pMetricsModule->pfRecordMaxValue)(pEASData->pMetricsData, EAS_PM_MAX_VOICES, (EAS_U32) voicesRendered); + } +#endif + + //2 Do we really need frameParsed? + /* need to parse another frame of events before we render again */ + for (streamNum = 0; streamNum < MAX_NUMBER_STREAMS; streamNum++) + if (pEASData->streams[streamNum].pParserModule != NULL) + pEASData->streams[streamNum].streamFlags &= ~STREAM_FLAGS_PARSED; + +#ifdef _METRICS_ENABLED + /* start performance counter */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_STREAM_TIME); +#endif + + /* render PCM audio */ + if ((result = EAS_PERender(pEASData, numRequested)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "EAS_PERender returned error %ld\n", result); */ } + return result; + } + +#ifdef _METRICS_ENABLED + /* stop the stream timer */ + if (pEASData->pMetricsData) + (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_STREAM_TIME); +#endif + +#ifdef _METRICS_ENABLED + /* start the post timer */ + if (pEASData->pMetricsData) + (*pEASData->pMetricsModule->pfStartTimer)(pEASData->pMetricsData, EAS_PM_POST_TIME); +#endif + + /* for split architecture, send DSP vectors. Do post only if return is TRUE */ +#ifdef _SPLIT_ARCHITECTURE + if (VMEndFrame(pEASData)) + { + /* now do post-processing */ + EAS_MixEnginePost(pEASData, numRequested); + *pNumGenerated = numRequested; + } +#else + /* now do post-processing */ + EAS_MixEnginePost(pEASData, numRequested); + *pNumGenerated = numRequested; +#endif + +#ifdef _METRICS_ENABLED + /* stop the post timer */ + if (pEASData->pMetricsData) + (void)(*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_POST_TIME); +#endif + + /* advance render time */ + pEASData->renderTime += AUDIO_FRAME_LENGTH; + +#if 0 + /* dump workload for debug */ + if (pEASData->pVoiceMgr->workload) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Workload = %d\n", pEASData->pVoiceMgr->workload); */ } +#endif + +#ifdef _METRICS_ENABLED + /* stop performance counter */ + if (pEASData->pMetricsData) + { + PERF_TIMER temp; + temp = (*pEASData->pMetricsModule->pfStopTimer)(pEASData->pMetricsData, EAS_PM_TOTAL_TIME); + + /* if max render time, record the number of voices and time */ + if ((*pEASData->pMetricsModule->pfRecordMaxValue) + (pEASData->pMetricsData, EAS_PM_MAX_CYCLES, (EAS_U32) temp)) + { + (*pEASData->pMetricsModule->pfRecordValue)(pEASData->pMetricsData, EAS_PM_MAX_CYCLES_VOICES, (EAS_U32) voicesRendered); + (*pEASData->pMetricsModule->pfRecordValue)(pEASData->pMetricsData, EAS_PM_MAX_CYCLES_TIME, (EAS_I32) (pEASData->renderTime >> 8)); + } + } +#endif + +#ifdef JET_INTERFACE + /* let JET to do its thing */ + if (pEASData->jetHandle != NULL) + { + result = JET_Process(pEASData); + if (result != EAS_SUCCESS) + return result; + } +#endif + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetRepeat() + *---------------------------------------------------------------------------- + * Purpose: + * Set the selected stream to repeat. + * + * Inputs: + * pEASData - handle to data for this instance + * handle - handle to stream + * repeatCount - repeat count + * + * Outputs: + * + * Side Effects: + * + * Notes: + * 0 = no repeat + * 1 = repeat once, i.e. play through twice + * -1 = repeat forever + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_PUBLIC EAS_RESULT EAS_SetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 repeatCount) +{ + pStream->repeatCount = repeatCount; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_GetRepeat() + *---------------------------------------------------------------------------- + * Purpose: + * Gets the current repeat count for the selected stream. + * + * Inputs: + * pEASData - handle to data for this instance + * handle - handle to stream + * pRrepeatCount - pointer to variable to hold repeat count + * + * Outputs: + * + * Side Effects: + * + * Notes: + * 0 = no repeat + * 1 = repeat once, i.e. play through twice + * -1 = repeat forever + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_PUBLIC EAS_RESULT EAS_GetRepeat (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pRepeatCount) +{ + *pRepeatCount = pStream->repeatCount; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetPlaybackRate() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the playback rate. + * + * Inputs: + * pEASData - handle to data for this instance + * handle - handle to stream + * rate - rate (28-bit fractional amount) + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_PUBLIC EAS_RESULT EAS_SetPlaybackRate (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U32 rate) +{ + + /* check range */ + if ((rate < (1 << 27)) || (rate > (1 << 29))) + return EAS_ERROR_INVALID_PARAMETER; + + /* calculate new frame length + * + * NOTE: The maximum frame length we can accomodate based on a + * maximum rate of 2.0 (2^28) is 2047 (2^13-1). To accomodate a + * longer frame length or a higher maximum rate, the fixed point + * divide below will need to be adjusted + */ + pStream->frameLength = (AUDIO_FRAME_LENGTH * (rate >> 8)) >> 20; + + /* notify stream of new playback rate */ + EAS_SetStreamParameter(pEASData, pStream, PARSER_DATA_PLAYBACK_RATE, (EAS_I32) rate); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetTransposition) + *---------------------------------------------------------------------------- + * Purpose: + * Sets the key tranposition for the synthesizer. Transposes all + * melodic instruments by the specified amount. Range is limited + * to +/-12 semitones. + * + * Inputs: + * pEASData - handle to data for this instance + * handle - handle to stream + * transposition - +/-12 semitones + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetTransposition (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 transposition) +{ + + /* check range */ + if ((transposition < -12) || (transposition > 12)) + return EAS_ERROR_INVALID_PARAMETER; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_TRANSPOSITION, transposition); +} + +/*---------------------------------------------------------------------------- + * EAS_ParseEvents() + *---------------------------------------------------------------------------- + * Purpose: + * Parse events in the current streams until the desired time is reached. + * + * Inputs: + * pEASData - buffer for internal EAS data + * endTime - stop parsing if this time is reached + * parseMode - play, locate, or metadata + * + * Outputs: + * EAS_SUCCESS if PCM data was successfully rendered + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT EAS_ParseEvents (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_U32 endTime, EAS_INT parseMode) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + EAS_I32 parserState; + EAS_BOOL done; + EAS_INT yieldCount = YIELD_EVENT_COUNT; + EAS_U32 time = 0; + + /* does this parser have a time function? */ + pParserModule = pStream->pParserModule; + if (pParserModule->pfTime == NULL) + { + /* check state */ + if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &parserState)) != EAS_SUCCESS) + return result; + /* if play state, advance time */ + if ((parserState >= EAS_STATE_READY) && (parserState <= EAS_STATE_PAUSING)) + pStream->time += pStream->frameLength; + done = EAS_TRUE; + } + + /* assume we're not done, in case we abort out */ + else + { + pStream->streamFlags &= ~STREAM_FLAGS_PARSED; + done = EAS_FALSE; + } + + while (!done) + { + + /* check for stopped state */ + if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &parserState)) != EAS_SUCCESS) + return result; + if (parserState > EAS_STATE_PLAY) + { + /* save current time if we're not in play mode */ + if (parseMode != eParserModePlay) + pStream->time = time << 8; + done = EAS_TRUE; + break; + } + + /* get the next event time */ + if (pParserModule->pfTime) + { + if ((result = (*pParserModule->pfTime)(pEASData, pStream->handle, &time)) != EAS_SUCCESS) + return result; + + /* if next event is within this frame, parse it */ + if (time < (endTime >> 8)) + { + + /* parse the next event */ + if (pParserModule->pfEvent) + if ((result = (*pParserModule->pfEvent)(pEASData, pStream->handle, parseMode)) != EAS_SUCCESS) + return result; + } + + /* no more events in this frame, advance time */ + else + { + pStream->time = endTime; + done = EAS_TRUE; + } + } + + /* check for max workload exceeded */ + if (VMCheckWorkload(pEASData->pVoiceMgr)) + { + /* stop even though we may not have parsed + * all the events in this frame. The parser will try to + * catch up on the next frame. + */ + break; + } + + /* give host a chance for an early abort */ + if (--yieldCount == 0) + { + if (EAS_HWYield(pEASData->hwInstData)) + break; + yieldCount = YIELD_EVENT_COUNT; + } + } + + /* if no early abort, parsing is complete for this frame */ + if (done) + pStream->streamFlags |= STREAM_FLAGS_PARSED; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_ParseMetaData() + *---------------------------------------------------------------------------- + * Purpose: + * + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * playLength - pointer to variable to store the play length (in msecs) + * + * Outputs: + * + * + * Side Effects: + * - resets the parser to the start of the file + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_ParseMetaData (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *playLength) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + EAS_STATE state; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + /* check parser state */ + if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &state)) != EAS_SUCCESS) + return result; + if (state >= EAS_STATE_OPEN) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* if parser has metadata function, use that */ + if (pParserModule->pfGetMetaData != NULL) + return pParserModule->pfGetMetaData(pEASData, pStream->handle, playLength); + + /* reset the parser to the beginning */ + if ((result = (*pParserModule->pfReset)(pEASData, pStream->handle)) != EAS_SUCCESS) + return result; + + /* parse the file to end */ + pStream->time = 0; + VMInitWorkload(pEASData->pVoiceMgr); + if ((result = EAS_ParseEvents(pEASData, pStream, 0x7fffffff, eParserModeMetaData)) != EAS_SUCCESS) + return result; + + /* get the parser time */ + if ((result = EAS_GetLocation(pEASData, pStream, playLength)) != EAS_SUCCESS) + return result; + + /* reset the parser to the beginning */ + pStream->time = 0; + return (*pParserModule->pfReset)(pEASData, pStream->handle); +} + +/*---------------------------------------------------------------------------- + * EAS_RegisterMetaDataCallback() + *---------------------------------------------------------------------------- + * Purpose: + * Registers a metadata callback function for parsed metadata. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * cbFunc - pointer to host callback function + * metaDataBuffer - pointer to metadata buffer + * metaDataBufSize - maximum size of the metadata buffer + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_RegisterMetaDataCallback ( + EAS_DATA_HANDLE pEASData, + EAS_HANDLE pStream, + EAS_METADATA_CBFUNC cbFunc, + char *metaDataBuffer, + EAS_I32 metaDataBufSize, + EAS_VOID_PTR pUserData) +{ + S_METADATA_CB metadata; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* register callback function */ + metadata.callback = cbFunc; + metadata.buffer = metaDataBuffer; + metadata.bufferSize = metaDataBufSize; + metadata.pUserData = pUserData; + return EAS_SetStreamParameter(pEASData, pStream, PARSER_DATA_METADATA_CB, (EAS_I32) &metadata); +} + +/*---------------------------------------------------------------------------- + * EAS_GetNoteCount () + *---------------------------------------------------------------------------- + * Returns the total number of notes played in this stream + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetNoteCount (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pNoteCount) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_NOTE_COUNT, pNoteCount); +} + +/*---------------------------------------------------------------------------- + * EAS_CloseFile() + *---------------------------------------------------------------------------- + * Purpose: + * Closes an audio file or stream. Playback should have either paused or + * completed (EAS_State returns EAS_PAUSED or EAS_STOPPED). + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_CloseFile (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + + /* call the close function */ + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + result = (*pParserModule->pfClose)(pEASData, pStream->handle); + + /* clear the handle and parser interface pointer */ + pStream->handle = NULL; + pStream->pParserModule = NULL; + return result; +} + +/*---------------------------------------------------------------------------- + * EAS_OpenMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Opens a raw MIDI stream allowing the host to route MIDI cable data directly to the synthesizer + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * pHandle - pointer to variable to hold file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_OpenMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE *ppStream, EAS_HANDLE streamHandle) +{ + EAS_RESULT result; + S_INTERACTIVE_MIDI *pMIDIStream; + EAS_INT streamNum; + + /* initialize some pointers */ + *ppStream = NULL; + + /* allocate a stream */ + if ((streamNum = EAS_AllocateStream(pEASData)) < 0) + return EAS_ERROR_MAX_STREAMS_OPEN; + + /* check Configuration Module for S_EAS_DATA allocation */ + if (pEASData->staticMemoryModel) + pMIDIStream = EAS_CMEnumData(EAS_CM_MIDI_STREAM_DATA); + else + pMIDIStream = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_INTERACTIVE_MIDI)); + + /* allocate dynamic memory */ + if (!pMIDIStream) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate MIDI stream data\n"); */ } + return EAS_ERROR_MALLOC_FAILED; + } + + /* zero the memory to insure complete initialization */ + EAS_HWMemSet(pMIDIStream, 0, sizeof(S_INTERACTIVE_MIDI)); + EAS_InitStream(&pEASData->streams[streamNum], NULL, pMIDIStream); + + /* instantiate a new synthesizer */ + if (streamHandle == NULL) + { + result = VMInitMIDI(pEASData, &pMIDIStream->pSynth); + } + + /* use an existing synthesizer */ + else + { + EAS_I32 value; + result = EAS_GetStreamParameter(pEASData, streamHandle, PARSER_DATA_SYNTH_HANDLE, &value); + pMIDIStream->pSynth = (S_SYNTH*) value; + VMIncRefCount(pMIDIStream->pSynth); + } + if (result != EAS_SUCCESS) + { + EAS_CloseMIDIStream(pEASData, &pEASData->streams[streamNum]); + return result; + } + + /* initialize the MIDI stream data */ + EAS_InitMIDIStream(&pMIDIStream->stream); + + *ppStream = (EAS_HANDLE) &pEASData->streams[streamNum]; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_WriteMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Send data to the MIDI stream device + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - stream handle + * pBuffer - pointer to buffer + * count - number of bytes to write + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_WriteMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U8 *pBuffer, EAS_I32 count) +{ + S_INTERACTIVE_MIDI *pMIDIStream; + EAS_RESULT result; + + pMIDIStream = (S_INTERACTIVE_MIDI*) pStream->handle; + + if (count <= 0) + return EAS_ERROR_PARAMETER_RANGE; + + /* send the entire buffer */ + while (count--) + { + if ((result = EAS_ParseMIDIStream(pEASData, pMIDIStream->pSynth, &pMIDIStream->stream, *pBuffer++, eParserModePlay)) != EAS_SUCCESS) + return result; + } + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_CloseMIDIStream() + *---------------------------------------------------------------------------- + * Purpose: + * Closes a raw MIDI stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_CloseMIDIStream (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + S_INTERACTIVE_MIDI *pMIDIStream; + + pMIDIStream = (S_INTERACTIVE_MIDI*) pStream->handle; + + /* close synth */ + if (pMIDIStream->pSynth != NULL) + { + VMMIDIShutdown(pEASData, pMIDIStream->pSynth); + pMIDIStream->pSynth = NULL; + } + + /* release allocated memory */ + if (!pEASData->staticMemoryModel) + EAS_HWFree(((S_EAS_DATA*) pEASData)->hwInstData, pMIDIStream); + + pStream->handle = NULL; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_State() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the state of an audio file or stream. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_State (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_STATE *pState) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + + /* call the parser to return state */ + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, pState)) != EAS_SUCCESS) + return result; + + /* if repeat count is set for this parser, mask the stopped state from the application */ + if (pStream->repeatCount && (*pState == EAS_STATE_STOPPED)) + *pState = EAS_STATE_PLAY; + + /* if we're not ready or playing, we don't need to hide state from host */ + if (*pState > EAS_STATE_PLAY) + return EAS_SUCCESS; + + /* if stream is about to be paused, report it as paused */ + if (pStream->streamFlags & STREAM_FLAGS_PAUSE) + { + if (pStream->streamFlags & STREAM_FLAGS_LOCATE) + *pState = EAS_STATE_PAUSED; + else + *pState = EAS_STATE_PAUSING; + } + + /* if stream is about to resume, report it as playing */ + if (pStream->streamFlags & STREAM_FLAGS_RESUME) + *pState = EAS_STATE_PLAY; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Set the polyphony of the stream. A value of 0 allows the stream + * to use all voices (set by EAS_SetSynthPolyphony). + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 polyphonyCount) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_POLYPHONY, polyphonyCount); +} + +/*---------------------------------------------------------------------------- + * EAS_GetPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current polyphony setting of the stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * pPolyphonyCount - pointer to variable to receive polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetPolyphony (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pPolyphonyCount) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_POLYPHONY, pPolyphonyCount); +} + +/*---------------------------------------------------------------------------- + * EAS_SetSynthPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Set the polyphony of the synth . Value must be >= 1 and <= the + * maximum number of voices. This function will pin the polyphony + * at those limits + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * synthNum - synthesizer number (0 = onboard, 1 = DSP) + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 polyphonyCount) +{ + return VMSetSynthPolyphony(pEASData->pVoiceMgr, synthNum, polyphonyCount); +} + +/*---------------------------------------------------------------------------- + * EAS_GetSynthPolyphony() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current polyphony setting of the synth + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * synthNum - synthesizer number (0 = onboard, 1 = DSP) + * pPolyphonyCount - pointer to variable to receive polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetSynthPolyphony (EAS_DATA_HANDLE pEASData, EAS_I32 synthNum, EAS_I32 *pPolyphonyCount) +{ + return VMGetSynthPolyphony(pEASData->pVoiceMgr, synthNum, pPolyphonyCount); +} + +/*---------------------------------------------------------------------------- + * EAS_SetPriority() + *---------------------------------------------------------------------------- + * Purpose: + * Set the priority of the stream. Determines which stream's voices + * are stolen when there are insufficient voices for all notes. + * Value must be in the range of 1-15, lower values are higher + * priority. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * polyphonyCount - the desired polyphony count + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 priority) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_PRIORITY, priority); +} + +/*---------------------------------------------------------------------------- + * EAS_GetPriority() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current priority setting of the stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * pPriority - pointer to variable to receive priority + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetPriority (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pPriority) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntGetStrmParam(pEASData, pStream, PARSER_DATA_PRIORITY, pPriority); +} + +/*---------------------------------------------------------------------------- + * EAS_SetVolume() + *---------------------------------------------------------------------------- + * Purpose: + * Set the master gain for the mix engine in 1dB increments + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * volume - the desired master gain (100 is max) + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * overrides any previously set master volume from sysex + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 volume) +{ + EAS_I16 gain; + + /* check range */ + if ((volume < 0) || (volume > EAS_MAX_VOLUME)) + return EAS_ERROR_PARAMETER_RANGE; + + /* stream volume */ + if (pStream != NULL) + { + EAS_I32 gainOffset; + EAS_RESULT result; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* get gain offset */ + pStream->volume = (EAS_U8) volume; + result = EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_GAIN_OFFSET, &gainOffset); + if (result == EAS_SUCCESS) + volume += gainOffset; + + /* set stream volume */ + gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); + + /* convert to linear scalar */ + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_VOLUME, gain); + } + + /* master volume */ + pEASData->masterVolume = (EAS_U8) volume; +#if (NUM_OUTPUT_CHANNELS == 1) + /* leave 3dB headroom for mono output */ + volume -= 3; +#endif + + gain = EAS_VolumeToGain(volume - STREAM_VOLUME_HEADROOM); + pEASData->masterGain = gain; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_GetVolume() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the master volume for the synthesizer. The default volume setting is + * 50. The volume range is 0 to 100; + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * volume - the desired master volume + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * overrides any previously set master volume from sysex + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_I32 EAS_GetVolume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + if (pStream == NULL) + return pEASData->masterVolume; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return pStream->volume; +} + +/*---------------------------------------------------------------------------- + * EAS_SetMaxLoad() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the maximum workload the parsers will do in a single call to + * EAS_Render. The units are currently arbitrary, but should correlate + * well to the actual CPU cycles consumed. The primary effect is to + * reduce the occasional peaks in CPU cycles consumed when parsing + * dense parts of a MIDI score. + * + * Inputs: + * pEASData - handle to data for this instance + * maxLoad - the desired maximum workload + * + * Outputs: + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetMaxLoad (EAS_DATA_HANDLE pEASData, EAS_I32 maxLoad) +{ + VMSetWorkload(pEASData->pVoiceMgr, maxLoad); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetMaxPCMStreams() + *---------------------------------------------------------------------------- + * Sets the maximum number of PCM streams allowed in parsers that + * use PCM streaming. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * streamHandle - handle returned by EAS_OpenFile + * maxNumStreams - maximum number of PCM streams + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetMaxPCMStreams (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 maxNumStreams) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_MAX_PCM_STREAMS, maxNumStreams); +} + +/*---------------------------------------------------------------------------- + * EAS_Locate() + *---------------------------------------------------------------------------- + * Purpose: + * Locate into the file associated with the handle. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file handle + * milliseconds - playback offset from start of file in milliseconds + * + * Outputs: + * + * + * Side Effects: + * the actual offset will be quantized to the closest update period, typically + * a resolution of 5.9ms. Notes that are started prior to this time will not + * sound. Any notes currently playing will be shut off. + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Locate (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 milliseconds, EAS_BOOL offset) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_RESULT result; + EAS_U32 requestedTime; + EAS_STATE state; + + /* get pointer to parser function table */ + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + if ((result = (*pParserModule->pfState)(pEASData, pStream->handle, &state)) != EAS_SUCCESS) + return result; + if (state >= EAS_STATE_OPEN) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* handle offset and limit to start of file */ + /*lint -e{704} use shift for performance*/ + if (offset) + milliseconds += (EAS_I32) pStream->time >> 8; + if (milliseconds < 0) + milliseconds = 0; + + /* check to see if the request is different from the current time */ + requestedTime = (EAS_U32) milliseconds; + if (requestedTime == (pStream->time >> 8)) + return EAS_SUCCESS; + + /* set the locate flag */ + pStream->streamFlags |= STREAM_FLAGS_LOCATE; + + /* use the parser locate function, if available */ + if (pParserModule->pfLocate != NULL) + { + EAS_BOOL parserLocate = EAS_FALSE; + result = pParserModule->pfLocate(pEASData, pStream->handle, (EAS_I32) requestedTime, &parserLocate); + if (!parserLocate) + { + if (result == EAS_SUCCESS) + pStream->time = requestedTime << 8; + return result; + } + } + + /* if we were paused and not going to resume, set pause request flag */ + if (((state == EAS_STATE_PAUSING) || (state == EAS_STATE_PAUSED)) && ((pStream->streamFlags & STREAM_FLAGS_RESUME) == 0)) + pStream->streamFlags |= STREAM_FLAGS_PAUSE; + + /* reset the synth and parser */ + if ((result = (*pParserModule->pfReset)(pEASData, pStream->handle)) != EAS_SUCCESS) + return result; + pStream->time = 0; + + /* locating forward, clear parsed flag and parse data until we get to the requested location */ + if ((result = EAS_ParseEvents(pEASData, pStream, requestedTime << 8, eParserModeLocate)) != EAS_SUCCESS) + return result; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_GetLocation() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current playback offset + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file handle + * + * Outputs: + * The offset in milliseconds from the start of the current sequence, quantized + * to the nearest update period. Actual resolution is typically 5.9 ms. + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_PUBLIC EAS_RESULT EAS_GetLocation (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 *pTime) +{ + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + *pTime = pStream->time >> 8; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_GetRenderTime() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current playback offset + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * Gets the render time clock in msecs. + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetRenderTime (EAS_DATA_HANDLE pEASData, EAS_I32 *pTime) +{ + *pTime = pEASData->renderTime >> 8; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_Pause() + *---------------------------------------------------------------------------- + * Purpose: + * Pauses the playback of the data associated with this handle. The audio + * is gracefully ramped down to prevent clicks and pops. It may take several + * buffers of audio before the audio is muted. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Pause (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_STATE state; + EAS_RESULT result; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + /* check for valid state */ + result = pParserModule->pfState(pEASData, pStream->handle, &state); + if (result == EAS_SUCCESS) + { + if ((state != EAS_STATE_PLAY) && (state != EAS_STATE_READY) && ((pStream->streamFlags & STREAM_FLAGS_RESUME) == 0)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* make sure parser implements pause */ + if (pParserModule->pfPause == NULL) + result = EAS_ERROR_NOT_IMPLEMENTED; + + /* clear resume flag */ + pStream->streamFlags &= ~STREAM_FLAGS_RESUME; + + /* set pause flag */ + pStream->streamFlags |= STREAM_FLAGS_PAUSE; + +#if 0 + /* pause the stream */ + if (pParserModule->pfPause) + result = pParserModule->pfPause(pEASData, pStream->handle); + else + result = EAS_ERROR_NOT_IMPLEMENTED; +#endif + } + + return result; +} + +/*---------------------------------------------------------------------------- + * EAS_Resume() + *---------------------------------------------------------------------------- + * Purpose: + * Resumes the playback of the data associated with this handle. The audio + * is gracefully ramped up to prevent clicks and pops. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * handle - file or stream handle + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_Resume (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream) +{ + S_FILE_PARSER_INTERFACE *pParserModule; + EAS_STATE state; + EAS_RESULT result; + + pParserModule = (S_FILE_PARSER_INTERFACE*) pStream->pParserModule; + if (pParserModule == NULL) + return EAS_ERROR_FEATURE_NOT_AVAILABLE; + + /* check for valid state */ + result = pParserModule->pfState(pEASData, pStream->handle, &state); + if (result == EAS_SUCCESS) + { + if ((state != EAS_STATE_PAUSED) && (state != EAS_STATE_PAUSING) && ((pStream->streamFlags & STREAM_FLAGS_PAUSE) == 0)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* make sure parser implements this function */ + if (pParserModule->pfResume == NULL) + result = EAS_ERROR_NOT_IMPLEMENTED; + + /* clear pause flag */ + pStream->streamFlags &= ~STREAM_FLAGS_PAUSE; + + /* set resume flag */ + pStream->streamFlags |= STREAM_FLAGS_RESUME; + +#if 0 + /* resume the stream */ + if (pParserModule->pfResume) + result = pParserModule->pfResume(pEASData, pStream->handle); + else + result = EAS_ERROR_NOT_IMPLEMENTED; +#endif + } + + return result; +} + +/*---------------------------------------------------------------------------- + * EAS_GetParameter() + *---------------------------------------------------------------------------- + * Purpose: + * Set the parameter of a module. See E_MODULES for a list of modules + * and the header files of the modules for a list of parameters. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * handle - file or stream handle + * module - enumerated module number + * param - enumerated parameter number + * pValue - pointer to variable to receive parameter value + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 *pValue) +{ + + if (module >= NUM_EFFECTS_MODULES) + return EAS_ERROR_INVALID_MODULE; + + if (pEASData->effectsModules[module].effectData == NULL) + return EAS_ERROR_INVALID_MODULE; + + return (*pEASData->effectsModules[module].effect->pFGetParam) + (pEASData->effectsModules[module].effectData, param, pValue); +} + +/*---------------------------------------------------------------------------- + * EAS_SetParameter() + *---------------------------------------------------------------------------- + * Purpose: + * Set the parameter of a module. See E_MODULES for a list of modules + * and the header files of the modules for a list of parameters. + * + * Inputs: + * psEASData - pointer to overall EAS data structure + * handle - file or stream handle + * module - enumerated module number + * param - enumerated parameter number + * value - new parameter value + * + * Outputs: + * + * + * Side Effects: + * + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetParameter (EAS_DATA_HANDLE pEASData, EAS_I32 module, EAS_I32 param, EAS_I32 value) +{ + + if (module >= NUM_EFFECTS_MODULES) + return EAS_ERROR_INVALID_MODULE; + + if (pEASData->effectsModules[module].effectData == NULL) + return EAS_ERROR_INVALID_MODULE; + + return (*pEASData->effectsModules[module].effect->pFSetParam) + (pEASData->effectsModules[module].effectData, param, value); +} + +#ifdef _METRICS_ENABLED +/*---------------------------------------------------------------------------- + * EAS_MetricsReport() + *---------------------------------------------------------------------------- + * Purpose: + * Displays the current metrics through the metrics interface. + * + * Inputs: + * p - instance data handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MetricsReport (EAS_DATA_HANDLE pEASData) +{ + if (!pEASData->pMetricsModule) + return EAS_ERROR_INVALID_MODULE; + + return (*pEASData->pMetricsModule->pfReport)(pEASData->pMetricsData); +} + +/*---------------------------------------------------------------------------- + * EAS_MetricsReset() + *---------------------------------------------------------------------------- + * Purpose: + * Resets the metrics. + * + * Inputs: + * p - instance data handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_MetricsReset (EAS_DATA_HANDLE pEASData) +{ + + if (!pEASData->pMetricsModule) + return EAS_ERROR_INVALID_MODULE; + + return (*pEASData->pMetricsModule->pfReset)(pEASData->pMetricsData); +} +#endif + +/*---------------------------------------------------------------------------- + * EAS_SetSoundLibrary() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the location of the sound library. + * + * Inputs: + * pEASData - instance data handle + * pSoundLib - pointer to sound library + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetSoundLibrary (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_SNDLIB_HANDLE pSndLib) +{ + if (pStream) + { + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_EAS_LIBRARY, (EAS_I32) pSndLib); + } + + return VMSetGlobalEASLib(pEASData->pVoiceMgr, pSndLib); +} + +/*---------------------------------------------------------------------------- + * EAS_SetHeaderSearchFlag() + *---------------------------------------------------------------------------- + * By default, when EAS_OpenFile is called, the parsers check the + * first few bytes of the file looking for a specific header. Some + * mobile devices may add a header to the start of a file, which + * will prevent the parser from recognizing the file. If the + * searchFlag is set to EAS_TRUE, the parser will search the entire + * file looking for the header. This may enable EAS to recognize + * some files that it would ordinarily reject. The negative is that + * it make take slightly longer to process the EAS_OpenFile request. + * + * Inputs: + * pEASData - instance data handle + * searchFlag - search flag (EAS_TRUE or EAS_FALSE) + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetHeaderSearchFlag (EAS_DATA_HANDLE pEASData, EAS_BOOL searchFlag) +{ + pEASData->searchHeaderFlag = (EAS_BOOL8) searchFlag; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_SetPlayMode() + *---------------------------------------------------------------------------- + * Some file formats support special play modes, such as iMode partial + * play mode. This call can be used to change the play mode. The + * default play mode (usually straight playback) is always zero. + * + * Inputs: + * pEASData - instance data handle + * handle - file or stream handle + * playMode - play mode (see file parser for specifics) + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetPlayMode (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_I32 playMode) +{ + return EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_PLAY_MODE, playMode); +} + +#ifdef DLS_SYNTHESIZER +/*---------------------------------------------------------------------------- + * EAS_LoadDLSCollection() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the location of the sound library. + * + * Inputs: + * pEASData - instance data handle + * pSoundLib - pointer to sound library + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_LoadDLSCollection (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_FILE_LOCATOR locator) +{ + EAS_FILE_HANDLE fileHandle; + EAS_RESULT result; + EAS_DLSLIB_HANDLE pDLS; + + if (pStream != NULL) + { + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + } + + /* open the file */ + if ((result = EAS_HWOpenFile(pEASData->hwInstData, locator, &fileHandle, EAS_FILE_READ)) != EAS_SUCCESS) + return result; + + /* parse the file */ + result = DLSParser(pEASData->hwInstData, fileHandle, 0, &pDLS); + EAS_HWCloseFile(pEASData->hwInstData, fileHandle); + + if (result == EAS_SUCCESS) + { + + /* if a stream pStream is specified, point it to the DLS collection */ + if (pStream) + result = EAS_IntSetStrmParam(pEASData, pStream, PARSER_DATA_DLS_COLLECTION, (EAS_I32) pDLS); + + /* global DLS load */ + else + result = VMSetGlobalDLSLib(pEASData, pDLS); + } + + return result; +} +#endif + +#ifdef EXTERNAL_AUDIO +/*---------------------------------------------------------------------------- + * EAS_RegExtAudioCallback() + *---------------------------------------------------------------------------- + * Purpose: + * Registers callback functions for audio events. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * cbProgChgFunc - pointer to host callback function for program change + * cbEventFunc - pointer to host callback functio for note events + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_RegExtAudioCallback (EAS_DATA_HANDLE pEASData, + EAS_HANDLE pStream, + EAS_VOID_PTR pInstData, + EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, + EAS_EXT_EVENT_FUNC cbEventFunc) +{ + S_SYNTH *pSynth; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) + return EAS_ERROR_INVALID_PARAMETER; + + if (pSynth == NULL) + return EAS_ERROR_INVALID_PARAMETER; + + VMRegExtAudioCallback(pSynth, pInstData, cbProgChgFunc, cbEventFunc); + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * EAS_GetMIDIControllers() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current state of MIDI controllers on the requested channel. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - file or stream handle + * pControl - pointer to structure to receive data + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_GetMIDIControllers (EAS_DATA_HANDLE pEASData, EAS_HANDLE pStream, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl) +{ + S_SYNTH *pSynth; + + if (!EAS_StreamReady(pEASData, pStream)) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + if (EAS_GetStreamParameter(pEASData, pStream, PARSER_DATA_SYNTH_HANDLE, (EAS_I32*) &pSynth) != EAS_SUCCESS) + return EAS_ERROR_INVALID_PARAMETER; + + if (pSynth == NULL) + return EAS_ERROR_INVALID_PARAMETER; + + VMGetMIDIControllers(pSynth, channel, pControl); + return EAS_SUCCESS; +} +#endif + +#ifdef _SPLIT_ARCHITECTURE +/*---------------------------------------------------------------------------- + * EAS_SetFrameBuffer() + *---------------------------------------------------------------------------- + * Purpose: + * Sets the frame buffer pointer passed to the IPC communications functions + * + * Inputs: + * pEASData - instance data handle + * locator - file locator + * + * Outputs: + * + * + * Side Effects: + * May overlay instruments in the GM sound set + * + *---------------------------------------------------------------------------- +*/ +EAS_PUBLIC EAS_RESULT EAS_SetFrameBuffer (EAS_DATA_HANDLE pEASData, EAS_FRAME_BUFFER_HANDLE pFrameBuffer) +{ + if (pEASData->pVoiceMgr) + pEASData->pVoiceMgr->pFrameBuffer = pFrameBuffer; + return EAS_SUCCESS; +} +#endif + +/*---------------------------------------------------------------------------- + * EAS_SearchFile + *---------------------------------------------------------------------------- + * Search file for specific sequence starting at current file + * position. Returns offset to start of sequence. + * + * Inputs: + * pEASData - pointer to EAS persistent data object + * fileHandle - file handle + * searchString - pointer to search sequence + * len - length of search sequence + * pOffset - pointer to variable to store offset to sequence + * + * Returns EAS_EOF if end-of-file is reached + *---------------------------------------------------------------------------- +*/ +EAS_RESULT EAS_SearchFile (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, const EAS_U8 *searchString, EAS_I32 len, EAS_I32 *pOffset) +{ + EAS_RESULT result; + EAS_INT index; + EAS_U8 c; + + *pOffset = -1; + index = 0; + for (;;) + { + result = EAS_HWGetByte(pEASData->hwInstData, fileHandle, &c); + if (result != EAS_SUCCESS) + return result; + if (c == searchString[index]) + { + index++; + if (index == 4) + { + result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, pOffset); + if (result != EAS_SUCCESS) + return result; + *pOffset -= len; + break; + } + } + else + index = 0; + } + return EAS_SUCCESS; +} + + diff --git a/arm-wt-22k/lib_src/eas_reverb.c b/arm-wt-22k/lib_src/eas_reverb.c new file mode 100644 index 0000000..6d99862 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_reverb.c @@ -0,0 +1,1154 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverb.c
+ *
+ * Contents and purpose:
+ * Contains the implementation of the Reverb effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 510 $
+ * $Date: 2006-12-19 01:47:33 -0800 (Tue, 19 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+/*------------------------------------
+ * includes
+ *------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_effects.h"
+#include "eas_math.h"
+#include "eas_reverbdata.h"
+#include "eas_reverb.h"
+#include "eas_config.h"
+#include "eas_host.h"
+#include "eas_report.h"
+
+/* prototypes for effects interface */
+static EAS_RESULT ReverbInit (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData);
+static void ReverbProcess (EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples);
+static EAS_RESULT ReverbShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT ReverbGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT ReverbSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+
+/* common effects interface for configuration module */
+const S_EFFECTS_INTERFACE EAS_Reverb =
+{
+ ReverbInit,
+ ReverbProcess,
+ ReverbShutdown,
+ ReverbGetParam,
+ ReverbSetParam
+};
+
+
+
+/*----------------------------------------------------------------------------
+ * InitializeReverb()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbInit(EAS_DATA_HANDLE pEASData, EAS_VOID_PTR *pInstData)
+{
+ EAS_I32 i;
+ EAS_U16 nOffset;
+ EAS_INT temp;
+
+ S_REVERB_OBJECT *pReverbData;
+ S_REVERB_PRESET *pPreset;
+
+ /* check Configuration Module for data allocation */
+ if (pEASData->staticMemoryModel)
+ pReverbData = EAS_CMEnumFXData(EAS_MODULE_REVERB);
+
+ /* allocate dynamic memory */
+ else
+ pReverbData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_REVERB_OBJECT));
+
+ if (pReverbData == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL, "Failed to allocate Reverb memory\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+
+ /* clear the structure */
+ EAS_HWMemSet(pReverbData, 0, sizeof(S_REVERB_OBJECT));
+
+ ReverbReadInPresets(pReverbData);
+
+ pReverbData->m_nMinSamplesToAdd = REVERB_UPDATE_PERIOD_IN_SAMPLES;
+
+ pReverbData->m_nRevOutFbkR = 0;
+ pReverbData->m_nRevOutFbkL = 0;
+
+ pReverbData->m_sAp0.m_zApIn = AP0_IN;
+ pReverbData->m_sAp0.m_zApOut = AP0_IN + DEFAULT_AP0_LENGTH;
+ pReverbData->m_sAp0.m_nApGain = DEFAULT_AP0_GAIN;
+
+ pReverbData->m_zD0In = DELAY0_IN;
+
+ pReverbData->m_sAp1.m_zApIn = AP1_IN;
+ pReverbData->m_sAp1.m_zApOut = AP1_IN + DEFAULT_AP1_LENGTH;
+ pReverbData->m_sAp1.m_nApGain = DEFAULT_AP1_GAIN;
+
+ pReverbData->m_zD1In = DELAY1_IN;
+
+ pReverbData->m_zLpf0 = 0;
+ pReverbData->m_zLpf1 = 0;
+ pReverbData->m_nLpfFwd = 8837;
+ pReverbData->m_nLpfFbk = 6494;
+
+ pReverbData->m_nSin = 0;
+ pReverbData->m_nCos = 0;
+ pReverbData->m_nSinIncrement = 0;
+ pReverbData->m_nCosIncrement = 0;
+
+ // set xfade parameters
+ pReverbData->m_nXfadeInterval = (EAS_U16)REVERB_XFADE_PERIOD_IN_SAMPLES;
+ pReverbData->m_nXfadeCounter = pReverbData->m_nXfadeInterval + 1; // force update on first iteration
+ pReverbData->m_nPhase = -32768;
+ pReverbData->m_nPhaseIncrement = REVERB_XFADE_PHASE_INCREMENT;
+
+ pReverbData->m_nNoise = (EAS_I16)0xABCD;
+
+ pReverbData->m_nMaxExcursion = 0x007F;
+
+ // set delay tap lengths
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion,
+ &pReverbData->m_nNoise );
+
+ pReverbData->m_zD1Cross =
+ DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset;
+
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion,
+ &pReverbData->m_nNoise );
+
+ pReverbData->m_zD0Cross =
+ DELAY1_OUT - pReverbData->m_nMaxExcursion - nOffset;
+
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion,
+ &pReverbData->m_nNoise );
+
+ pReverbData->m_zD0Self =
+ DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset;
+
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion,
+ &pReverbData->m_nNoise );
+
+ pReverbData->m_zD1Self =
+ DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset;
+
+ // for debugging purposes, allow noise generator
+ pReverbData->m_bUseNoise = EAS_FALSE;
+
+ // for debugging purposes, allow bypass
+ pReverbData->m_bBypass = EAS_TRUE; //EAS_FALSE;
+
+ pReverbData->m_nNextRoom = 1;
+
+ pReverbData->m_nCurrentRoom = pReverbData->m_nNextRoom + 1; // force update on first iteration
+
+ pReverbData->m_nWet = REVERB_DEFAULT_WET;
+
+ pReverbData->m_nDry = REVERB_DEFAULT_DRY;
+
+ // set base index into circular buffer
+ pReverbData->m_nBaseIndex = 0;
+
+ // set the early reflections, L
+ pReverbData->m_sEarlyL.m_nLpfFbk = 4915;
+ pReverbData->m_sEarlyL.m_nLpfFwd = 27852;
+ pReverbData->m_sEarlyL.m_zLpf = 0;
+
+ for (i=0; i < REVERB_MAX_NUM_REFLECTIONS; i++)
+ {
+ pReverbData->m_sEarlyL.m_nGain[i] = 0;
+ pReverbData->m_sEarlyL.m_zDelay[i] = 0;
+ }
+
+ // set the early reflections, R
+ pReverbData->m_sEarlyR.m_nLpfFbk = 4915;
+ pReverbData->m_sEarlyR.m_nLpfFwd = 27852;
+ pReverbData->m_sEarlyR.m_zLpf = 0;
+
+ for (i=0; i < REVERB_MAX_NUM_REFLECTIONS; i++)
+ {
+ pReverbData->m_sEarlyR.m_nGain[i] = 0;
+ pReverbData->m_sEarlyR.m_zDelay[i] = 0;
+ }
+
+ // clear the reverb delay line
+ for (i=0; i < REVERB_BUFFER_SIZE_IN_SAMPLES; i++)
+ {
+ pReverbData->m_nDelayLine[i] = 0;
+ }
+
+ ////////////////////////////////
+ ///code from the EAS DEMO Reverb
+ //now copy from the new preset into the reverb
+ pPreset = &pReverbData->m_sPreset.m_sPreset[pReverbData->m_nNextRoom];
+
+ pReverbData->m_nLpfFbk = pPreset->m_nLpfFbk;
+ pReverbData->m_nLpfFwd = pPreset->m_nLpfFwd;
+
+ pReverbData->m_nEarly = pPreset->m_nEarly;
+ pReverbData->m_nWet = pPreset->m_nWet;
+ pReverbData->m_nDry = pPreset->m_nDry;
+
+ pReverbData->m_nMaxExcursion = pPreset->m_nMaxExcursion;
+ //stored as time based, convert to sample based
+ temp = pPreset->m_nXfadeInterval;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_nXfadeInterval = (EAS_U16) temp;
+ //gsReverbObject.m_nXfadeInterval = pPreset->m_nXfadeInterval;
+
+ pReverbData->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain;
+ //stored as time based, convert to absolute sample value
+ temp = pPreset->m_nAp0_ApOut;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_sAp0.m_zApOut = (EAS_U16) (pReverbData->m_sAp0.m_zApIn + temp);
+ //gsReverbObject.m_sAp0.m_zApOut = pPreset->m_nAp0_ApOut;
+
+ pReverbData->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain;
+ //stored as time based, convert to absolute sample value
+ temp = pPreset->m_nAp1_ApOut;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_sAp1.m_zApOut = (EAS_U16) (pReverbData->m_sAp1.m_zApIn + temp);
+ //gsReverbObject.m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut;
+ ///code from the EAS DEMO Reverb
+ ////////////////////////////////
+
+ *pInstData = pReverbData;
+
+ return EAS_SUCCESS;
+
+} /* end InitializeReverb */
+
+
+
+/*----------------------------------------------------------------------------
+ * ReverbProcess()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reverberate the requested number of samples (block based processing)
+ *
+ * Inputs:
+ * pInputBuffer - src buffer
+ * pOutputBuffer - dst buffer
+ * nNumSamplesToAdd - number of samples to write to buffer
+ *
+ * Outputs:
+ * number of samples actually written to buffer
+ *
+ * Side Effects:
+ * - samples are added to the presently free buffer
+ *
+ *----------------------------------------------------------------------------
+*/
+static void ReverbProcess(EAS_VOID_PTR pInstData, EAS_PCM *pSrc, EAS_PCM *pDst, EAS_I32 numSamples)
+{
+ S_REVERB_OBJECT *pReverbData;
+
+ pReverbData = (S_REVERB_OBJECT*) pInstData;
+
+ //if bypassed or the preset forces the signal to be completely dry
+ if (pReverbData->m_bBypass ||
+ (pReverbData->m_nWet == 0 && pReverbData->m_nDry == 32767))
+ {
+ if (pSrc != pDst)
+ EAS_HWMemCpy(pSrc, pDst, numSamples * NUM_OUTPUT_CHANNELS * (EAS_I32) sizeof(EAS_PCM));
+ return;
+ }
+
+ if (pReverbData->m_nNextRoom != pReverbData->m_nCurrentRoom)
+ {
+ ReverbUpdateRoom(pReverbData);
+ }
+
+ ReverbUpdateXfade(pReverbData, numSamples);
+
+ Reverb(pReverbData, numSamples, pDst, pSrc);
+
+ /* check if update counter needs to be reset */
+ if (pReverbData->m_nUpdateCounter >= REVERB_MODULO_UPDATE_PERIOD_IN_SAMPLES)
+ {
+ /* update interval has elapsed, so reset counter */
+ pReverbData->m_nUpdateCounter = 0;
+ } /* end if m_nUpdateCounter >= update interval */
+
+ /* increment update counter */
+ pReverbData->m_nUpdateCounter += (EAS_I16)numSamples;
+
+} /* end ComputeReverb */
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateXfade
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the xfade parameters as required
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to write to buffer
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - xfade parameters will be changed
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbUpdateXfade(S_REVERB_OBJECT *pReverbData, EAS_INT nNumSamplesToAdd)
+{
+ EAS_U16 nOffset;
+ EAS_I16 tempCos;
+ EAS_I16 tempSin;
+
+ if (pReverbData->m_nXfadeCounter >= pReverbData->m_nXfadeInterval)
+ {
+ /* update interval has elapsed, so reset counter */
+ pReverbData->m_nXfadeCounter = 0;
+
+ // Pin the sin,cos values to min / max values to ensure that the
+ // modulated taps' coefs are zero (thus no clicks)
+ if (pReverbData->m_nPhaseIncrement > 0)
+ {
+ // if phase increment > 0, then sin -> 1, cos -> 0
+ pReverbData->m_nSin = 32767;
+ pReverbData->m_nCos = 0;
+
+ // reset the phase to match the sin, cos values
+ pReverbData->m_nPhase = 32767;
+
+ // modulate the cross taps because their tap coefs are zero
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise );
+
+ pReverbData->m_zD1Cross =
+ DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset;
+
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise );
+
+ pReverbData->m_zD0Cross =
+ DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset;
+ }
+ else
+ {
+ // if phase increment < 0, then sin -> 0, cos -> 1
+ pReverbData->m_nSin = 0;
+ pReverbData->m_nCos = 32767;
+
+ // reset the phase to match the sin, cos values
+ pReverbData->m_nPhase = -32768;
+
+ // modulate the self taps because their tap coefs are zero
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise );
+
+ pReverbData->m_zD0Self =
+ DELAY0_OUT - pReverbData->m_nMaxExcursion - nOffset;
+
+ nOffset = ReverbCalculateNoise( pReverbData->m_nMaxExcursion, &pReverbData->m_nNoise );
+
+ pReverbData->m_zD1Self =
+ DELAY1_OUT - pReverbData->m_nMaxExcursion + nOffset;
+
+ } // end if-else (pReverbData->m_nPhaseIncrement > 0)
+
+ // Reverse the direction of the sin,cos so that the
+ // tap whose coef was previously increasing now decreases
+ // and vice versa
+ pReverbData->m_nPhaseIncrement = -pReverbData->m_nPhaseIncrement;
+
+ } // end if counter >= update interval
+
+ //compute what phase will be next time
+ pReverbData->m_nPhase += pReverbData->m_nPhaseIncrement;
+
+ //calculate what the new sin and cos need to reach by the next update
+ ReverbCalculateSinCos(pReverbData->m_nPhase, &tempSin, &tempCos);
+
+ //calculate the per-sample increment required to get there by the next update
+ /*lint -e{702} shift for performance */
+ pReverbData->m_nSinIncrement =
+ (tempSin - pReverbData->m_nSin) >> REVERB_UPDATE_PERIOD_IN_BITS;
+
+ /*lint -e{702} shift for performance */
+ pReverbData->m_nCosIncrement =
+ (tempCos - pReverbData->m_nCos) >> REVERB_UPDATE_PERIOD_IN_BITS;
+
+
+ /* increment update counter */
+ pReverbData->m_nXfadeCounter += (EAS_U16) nNumSamplesToAdd;
+
+ return EAS_SUCCESS;
+
+} /* end ReverbUpdateXfade */
+
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateNoise
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a noise sample and limit its value
+ *
+ * Inputs:
+ * nMaxExcursion - noise value is limited to this value
+ * pnNoise - return new noise sample in this (not limited)
+ *
+ * Outputs:
+ * new limited noise value
+ *
+ * Side Effects:
+ * - *pnNoise noise value is updated
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_U16 ReverbCalculateNoise(EAS_U16 nMaxExcursion, EAS_I16 *pnNoise)
+{
+ // calculate new noise value
+ *pnNoise = (EAS_I16) (*pnNoise * 5 + 1);
+
+#if 0 // 1xxx, test
+ *pnNoise = 0;
+#endif // 1xxx, test
+
+ // return the limited noise value
+ return (nMaxExcursion & (*pnNoise));
+
+} /* end ReverbCalculateNoise */
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateSinCos
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a new sin and cosine value based on the given phase
+ *
+ * Inputs:
+ * nPhase - phase angle
+ * pnSin - input old value, output new value
+ * pnCos - input old value, output new value
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - *pnSin, *pnCos are updated
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbCalculateSinCos(EAS_I16 nPhase, EAS_I16 *pnSin, EAS_I16 *pnCos)
+{
+ EAS_I32 nTemp;
+ EAS_I32 nNetAngle;
+
+ // -1 <= nPhase < 1
+ // However, for the calculation, we need a value
+ // that ranges from -1/2 to +1/2, so divide the phase by 2
+ /*lint -e{702} shift for performance */
+ nNetAngle = nPhase >> 1;
+
+ /*
+ Implement the following
+ sin(x) = (2-4*c)*x^2 + c + x
+ cos(x) = (2-4*c)*x^2 + c - x
+
+ where c = 1/sqrt(2)
+ using the a0 + x*(a1 + x*a2) approach
+ */
+
+ /* limit the input "angle" to be between -0.5 and +0.5 */
+ if (nNetAngle > EG1_HALF)
+ {
+ nNetAngle = EG1_HALF;
+ }
+ else if (nNetAngle < EG1_MINUS_HALF)
+ {
+ nNetAngle = EG1_MINUS_HALF;
+ }
+
+ /* calculate sin */
+ nTemp = EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle);
+ nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle);
+ *pnSin = (EAS_I16) SATURATE_EG1(nTemp);
+
+ /* calculate cos */
+ nTemp = -EG1_ONE + MULT_EG1_EG1(REVERB_PAN_G2, nNetAngle);
+ nTemp = REVERB_PAN_G0 + MULT_EG1_EG1(nTemp, nNetAngle);
+ *pnCos = (EAS_I16) SATURATE_EG1(nTemp);
+
+ return EAS_SUCCESS;
+} /* end ReverbCalculateSinCos */
+
+/*----------------------------------------------------------------------------
+ * Reverb
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * apply reverb to the given signal
+ *
+ * Inputs:
+ * nNu
+ * pnSin - input old value, output new value
+ * pnCos - input old value, output new value
+ *
+ * Outputs:
+ * number of samples actually reverberated
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT Reverb(S_REVERB_OBJECT *pReverbData, EAS_INT nNumSamplesToAdd, EAS_PCM *pOutputBuffer, EAS_PCM *pInputBuffer)
+{
+ EAS_I32 i;
+ EAS_I32 nDelayOut;
+ EAS_U16 nBase;
+
+ EAS_U32 nAddr;
+ EAS_I32 nTemp1;
+ EAS_I32 nTemp2;
+ EAS_I32 nApIn;
+ EAS_I32 nApOut;
+
+ EAS_I32 j;
+ EAS_I32 nEarlyOut;
+
+ EAS_I32 tempValue;
+
+
+ // get the base address
+ nBase = pReverbData->m_nBaseIndex;
+
+ for (i=0; i < nNumSamplesToAdd; i++)
+ {
+ // ********** Left Allpass - start
+ // left input = (left dry/4) + right feedback from previous period
+ /*lint -e{702} use shift for performance */
+ nApIn = ((*pInputBuffer++)>>2) + pReverbData->m_nRevOutFbkR;
+// nApIn = *pInputBuffer++; // 1xxx test and debug ap
+
+ // fetch allpass delay line out
+ //nAddr = CIRCULAR(nBase, psAp0->m_zApOut, REVERB_BUFFER_MASK);
+ nAddr = CIRCULAR(nBase, pReverbData->m_sAp0.m_zApOut, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate allpass feedforward; subtract the feedforward result
+ nTemp1 = MULT_EG1_EG1(nApIn, pReverbData->m_sAp0.m_nApGain);
+ nApOut = SATURATE(nDelayOut - nTemp1); // allpass output
+
+ // calculate allpass feedback; add the feedback result
+ nTemp1 = MULT_EG1_EG1(nApOut, pReverbData->m_sAp0.m_nApGain);
+ nTemp1 = SATURATE(nApIn + nTemp1);
+
+ // inject into allpass delay
+ nAddr = CIRCULAR(nBase, pReverbData->m_sAp0.m_zApIn, REVERB_BUFFER_MASK);
+ pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nTemp1;
+
+ // inject allpass output into delay line
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD0In, REVERB_BUFFER_MASK);
+ pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nApOut;
+
+ // ********** Left Allpass - end
+
+ // ********** Right Allpass - start
+ // right input = (right dry/4) + left feedback from previous period
+ /*lint -e{702} use shift for performance */
+ nApIn = ((*pInputBuffer++)>>2) + pReverbData->m_nRevOutFbkL;
+// nApIn = *pInputBuffer++; // 1xxx test and debug ap
+
+ // fetch allpass delay line out
+ nAddr = CIRCULAR(nBase, pReverbData->m_sAp1.m_zApOut, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate allpass feedforward; subtract the feedforward result
+ nTemp1 = MULT_EG1_EG1(nApIn, pReverbData->m_sAp1.m_nApGain);
+ nApOut = SATURATE(nDelayOut - nTemp1); // allpass output
+
+ // calculate allpass feedback; add the feedback result
+ nTemp1 = MULT_EG1_EG1(nApOut, pReverbData->m_sAp1.m_nApGain);
+ nTemp1 = SATURATE(nApIn + nTemp1);
+
+ // inject into allpass delay
+ nAddr = CIRCULAR(nBase, pReverbData->m_sAp1.m_zApIn, REVERB_BUFFER_MASK);
+ pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nTemp1;
+
+ // inject allpass output into delay line
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD1In, REVERB_BUFFER_MASK);
+ pReverbData->m_nDelayLine[nAddr] = (EAS_PCM) nApOut;
+
+ // ********** Right Allpass - end
+
+ // ********** D0 output - start
+ // fetch delay line self out
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD0Self, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate delay line self out
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nSin);
+
+ // fetch delay line cross out
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD1Cross, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate delay line self out
+ nTemp2 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nCos);
+
+ // calculate unfiltered delay out
+ nDelayOut = SATURATE(nTemp1 + nTemp2);
+
+ // calculate lowpass filter (mixer scale factor included in LPF feedforward)
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nLpfFwd);
+
+ nTemp2 = MULT_EG1_EG1(pReverbData->m_zLpf0, pReverbData->m_nLpfFbk);
+
+ // calculate filtered delay out and simultaneously update LPF state variable
+ // filtered delay output is stored in m_zLpf0
+ pReverbData->m_zLpf0 = (EAS_PCM) SATURATE(nTemp1 + nTemp2);
+
+ // ********** D0 output - end
+
+ // ********** D1 output - start
+ // fetch delay line self out
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD1Self, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate delay line self out
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nSin);
+
+ // fetch delay line cross out
+ nAddr = CIRCULAR(nBase, pReverbData->m_zD0Cross, REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate delay line self out
+ nTemp2 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nCos);
+
+ // calculate unfiltered delay out
+ nDelayOut = SATURATE(nTemp1 + nTemp2);
+
+ // calculate lowpass filter (mixer scale factor included in LPF feedforward)
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_nLpfFwd);
+
+ nTemp2 = MULT_EG1_EG1(pReverbData->m_zLpf1, pReverbData->m_nLpfFbk);
+
+ // calculate filtered delay out and simultaneously update LPF state variable
+ // filtered delay output is stored in m_zLpf1
+ pReverbData->m_zLpf1 = (EAS_PCM)SATURATE(nTemp1 + nTemp2);
+
+ // ********** D1 output - end
+
+ // ********** mixer and feedback - start
+ // sum is fedback to right input (R + L)
+ pReverbData->m_nRevOutFbkL =
+ (EAS_PCM)SATURATE((EAS_I32)pReverbData->m_zLpf1 + (EAS_I32)pReverbData->m_zLpf0);
+
+ // difference is feedback to left input (R - L)
+ /*lint -e{685} lint complains that it can't saturate negative */
+ pReverbData->m_nRevOutFbkR =
+ (EAS_PCM)SATURATE((EAS_I32)pReverbData->m_zLpf1 - (EAS_I32)pReverbData->m_zLpf0);
+
+ // ********** mixer and feedback - end
+
+ // ********** start early reflection generator, left
+ //psEarly = &(pReverbData->m_sEarlyL);
+
+ nEarlyOut = 0;
+
+ for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++)
+ {
+ // fetch delay line out
+ //nAddr = CIRCULAR(nBase, psEarly->m_zDelay[j], REVERB_BUFFER_MASK);
+ nAddr = CIRCULAR(nBase, pReverbData->m_sEarlyL.m_zDelay[j], REVERB_BUFFER_MASK);
+
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate reflection
+ //nTemp1 = MULT_EG1_EG1(nDelayOut, psEarly->m_nGain[j]);
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_sEarlyL.m_nGain[j]);
+
+ nEarlyOut = SATURATE(nEarlyOut + nTemp1);
+
+ } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++)
+
+ // apply lowpass to early reflections
+ //nTemp1 = MULT_EG1_EG1(nEarlyOut, psEarly->m_nLpfFwd);
+ nTemp1 = MULT_EG1_EG1(nEarlyOut, pReverbData->m_sEarlyL.m_nLpfFwd);
+
+ //nTemp2 = MULT_EG1_EG1(psEarly->m_zLpf, psEarly->m_nLpfFbk);
+ nTemp2 = MULT_EG1_EG1(pReverbData->m_sEarlyL.m_zLpf, pReverbData->m_sEarlyL.m_nLpfFbk);
+
+
+ // calculate filtered out and simultaneously update LPF state variable
+ // filtered output is stored in m_zLpf1
+ //psEarly->m_zLpf = SATURATE(nTemp1 + nTemp2);
+ pReverbData->m_sEarlyL.m_zLpf = (EAS_PCM) SATURATE(nTemp1 + nTemp2);
+
+ // combine filtered early and late reflections for output
+ //*pOutputBuffer++ = inL;
+ //tempValue = SATURATE(psEarly->m_zLpf + pReverbData->m_nRevOutFbkL);
+ tempValue = SATURATE((EAS_I32)pReverbData->m_sEarlyL.m_zLpf + (EAS_I32)pReverbData->m_nRevOutFbkL);
+ //scale reverb output by wet level
+ /*lint -e{701} use shift for performance */
+ tempValue = MULT_EG1_EG1(tempValue, (pReverbData->m_nWet<<1));
+ //sum with output buffer
+ tempValue += *pOutputBuffer;
+ *pOutputBuffer++ = (EAS_PCM)SATURATE(tempValue);
+
+ // ********** end early reflection generator, left
+
+ // ********** start early reflection generator, right
+ //psEarly = &(pReverbData->m_sEarlyR);
+
+ nEarlyOut = 0;
+
+ for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++)
+ {
+ // fetch delay line out
+ nAddr = CIRCULAR(nBase, pReverbData->m_sEarlyR.m_zDelay[j], REVERB_BUFFER_MASK);
+ nDelayOut = pReverbData->m_nDelayLine[nAddr];
+
+ // calculate reflection
+ nTemp1 = MULT_EG1_EG1(nDelayOut, pReverbData->m_sEarlyR.m_nGain[j]);
+
+ nEarlyOut = SATURATE(nEarlyOut + nTemp1);
+
+ } // end for (j=0; j < REVERB_MAX_NUM_REFLECTIONS; j++)
+
+ // apply lowpass to early reflections
+ nTemp1 = MULT_EG1_EG1(nEarlyOut, pReverbData->m_sEarlyR.m_nLpfFwd);
+
+ nTemp2 = MULT_EG1_EG1(pReverbData->m_sEarlyR.m_zLpf, pReverbData->m_sEarlyR.m_nLpfFbk);
+
+ // calculate filtered out and simultaneously update LPF state variable
+ // filtered output is stored in m_zLpf1
+ pReverbData->m_sEarlyR.m_zLpf = (EAS_PCM)SATURATE(nTemp1 + nTemp2);
+
+ // combine filtered early and late reflections for output
+ //*pOutputBuffer++ = inR;
+ tempValue = SATURATE((EAS_I32)pReverbData->m_sEarlyR.m_zLpf + (EAS_I32)pReverbData->m_nRevOutFbkR);
+ //scale reverb output by wet level
+ /*lint -e{701} use shift for performance */
+ tempValue = MULT_EG1_EG1(tempValue, (pReverbData->m_nWet << 1));
+ //sum with output buffer
+ tempValue = tempValue + *pOutputBuffer;
+ *pOutputBuffer++ = (EAS_PCM)SATURATE(tempValue);
+
+ // ********** end early reflection generator, right
+
+ // decrement base addr for next sample period
+ nBase--;
+
+ pReverbData->m_nSin += pReverbData->m_nSinIncrement;
+ pReverbData->m_nCos += pReverbData->m_nCosIncrement;
+
+ } // end for (i=0; i < nNumSamplesToAdd; i++)
+
+ // store the most up to date version
+ pReverbData->m_nBaseIndex = nBase;
+
+ return EAS_SUCCESS;
+} /* end Reverb */
+
+
+
+/*----------------------------------------------------------------------------
+ * ReverbShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initializes the Reverb effect.
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbShutdown (EAS_DATA_HANDLE pEASData, EAS_VOID_PTR pInstData)
+{
+ /* check Configuration Module for static memory allocation */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pInstData);
+ return EAS_SUCCESS;
+} /* end ReverbShutdown */
+
+/*----------------------------------------------------------------------------
+ * ReverbGetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get a Reverb parameter
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ * param - parameter index
+ * *pValue - pointer to variable to hold retrieved value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbGetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_REVERB_OBJECT *p;
+
+ p = (S_REVERB_OBJECT*) pInstData;
+
+ switch (param)
+ {
+ case EAS_PARAM_REVERB_BYPASS:
+ *pValue = (EAS_I32) p->m_bBypass;
+ break;
+ case EAS_PARAM_REVERB_PRESET:
+ *pValue = (EAS_I8) p->m_nCurrentRoom;
+ break;
+ case EAS_PARAM_REVERB_WET:
+ *pValue = p->m_nWet;
+ break;
+ case EAS_PARAM_REVERB_DRY:
+ *pValue = p->m_nDry;
+ break;
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+} /* end ReverbGetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ReverbSetParam()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set a Reverb parameter
+ *
+ * Inputs:
+ * pInstData - handle to instance data
+ * param - parameter index
+ * *pValue - new paramter value
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbSetParam (EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_REVERB_OBJECT *p;
+
+ p = (S_REVERB_OBJECT*) pInstData;
+
+ switch (param)
+ {
+ case EAS_PARAM_REVERB_BYPASS:
+ p->m_bBypass = (EAS_BOOL) value;
+ break;
+ case EAS_PARAM_REVERB_PRESET:
+ if(value!=EAS_PARAM_REVERB_LARGE_HALL && value!=EAS_PARAM_REVERB_HALL &&
+ value!=EAS_PARAM_REVERB_CHAMBER && value!=EAS_PARAM_REVERB_ROOM)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nNextRoom = (EAS_I16)value;
+ break;
+ case EAS_PARAM_REVERB_WET:
+ if(value>EAS_REVERB_WET_MAX || value<EAS_REVERB_WET_MIN)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nWet = (EAS_I16)value;
+ break;
+ case EAS_PARAM_REVERB_DRY:
+ if(value>EAS_REVERB_DRY_MAX || value<EAS_REVERB_DRY_MIN)
+ return EAS_ERROR_INVALID_PARAMETER;
+ p->m_nDry = (EAS_I16)value;
+ break;
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+} /* end ReverbSetParam */
+
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateRoom
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the room's preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - reverb paramters (fbk, fwd, etc) will be changed
+ * - m_nCurrentRoom := m_nNextRoom
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbUpdateRoom(S_REVERB_OBJECT *pReverbData)
+{
+ EAS_INT temp;
+
+ S_REVERB_PRESET *pPreset = &pReverbData->m_sPreset.m_sPreset[pReverbData->m_nNextRoom];
+
+ pReverbData->m_nLpfFwd = pPreset->m_nLpfFwd;
+ pReverbData->m_nLpfFbk = pPreset->m_nLpfFbk;
+
+ pReverbData->m_nEarly = pPreset->m_nEarly;
+ pReverbData->m_nWet = pPreset->m_nWet;
+ pReverbData->m_nDry = pPreset->m_nDry;
+
+
+ pReverbData->m_nMaxExcursion = pPreset->m_nMaxExcursion;
+ //stored as time based, convert to sample based
+ temp = pPreset->m_nXfadeInterval;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_nXfadeInterval = (EAS_U16) temp;
+ //gpsReverbObject->m_nXfadeInterval = pPreset->m_nXfadeInterval;
+ pReverbData->m_sAp0.m_nApGain = pPreset->m_nAp0_ApGain;
+ //stored as time based, convert to absolute sample value
+ temp = pPreset->m_nAp0_ApOut;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_sAp0.m_zApOut = (EAS_U16) (pReverbData->m_sAp0.m_zApIn + temp);
+ //gpsReverbObject->m_sAp0.m_zApOut = pPreset->m_nAp0_ApOut;
+ pReverbData->m_sAp1.m_nApGain = pPreset->m_nAp1_ApGain;
+ //stored as time based, convert to absolute sample value
+ temp = pPreset->m_nAp1_ApOut;
+ /*lint -e{702} shift for performance */
+ temp = (temp * _OUTPUT_SAMPLE_RATE) >> 16;
+ pReverbData->m_sAp1.m_zApOut = (EAS_U16) (pReverbData->m_sAp1.m_zApIn + temp);
+ //gpsReverbObject->m_sAp1.m_zApOut = pPreset->m_nAp1_ApOut;
+
+ pReverbData->m_nCurrentRoom = pReverbData->m_nNextRoom;
+
+ return EAS_SUCCESS;
+
+} /* end ReverbUpdateRoom */
+
+
+/*----------------------------------------------------------------------------
+ * ReverbReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global reverb preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbReadInPresets(S_REVERB_OBJECT *pReverbData)
+{
+
+ int preset = 0;
+ int defaultPreset = 0;
+
+ //now init any remaining presets to defaults
+ for (defaultPreset = preset; defaultPreset < REVERB_MAX_ROOM_TYPE; defaultPreset++)
+ {
+ S_REVERB_PRESET *pPreset = &pReverbData->m_sPreset.m_sPreset[defaultPreset];
+ if (defaultPreset == 0 || defaultPreset > REVERB_MAX_ROOM_TYPE-1)
+ {
+ pPreset->m_nLpfFbk = 8307;
+ pPreset->m_nLpfFwd = 14768;
+ pPreset->m_nEarly = 0;
+ pPreset->m_nWet = 27690;
+ pPreset->m_nDry = 32767;
+ pPreset->m_nEarlyL_LpfFbk = 3692;
+ pPreset->m_nEarlyL_LpfFwd = 29075;
+ pPreset->m_nEarlyL_Delay0 = 922;
+ pPreset->m_nEarlyL_Gain0 = 22152;
+ pPreset->m_nEarlyL_Delay1 = 1462;
+ pPreset->m_nEarlyL_Gain1 = 17537;
+ pPreset->m_nEarlyL_Delay2 = 0;
+ pPreset->m_nEarlyL_Gain2 = 14768;
+ pPreset->m_nEarlyL_Delay3 = 1221;
+ pPreset->m_nEarlyL_Gain3 = 14307;
+ pPreset->m_nEarlyL_Delay4 = 0;
+ pPreset->m_nEarlyL_Gain4 = 13384;
+ pPreset->m_nEarlyR_Delay0 = 502;
+ pPreset->m_nEarlyR_Gain0 = 20306;
+ pPreset->m_nEarlyR_Delay1 = 1762;
+ pPreset->m_nEarlyR_Gain1 = 17537;
+ pPreset->m_nEarlyR_Delay2 = 0;
+ pPreset->m_nEarlyR_Gain2 = 14768;
+ pPreset->m_nEarlyR_Delay3 = 0;
+ pPreset->m_nEarlyR_Gain3 = 16153;
+ pPreset->m_nEarlyR_Delay4 = 0;
+ pPreset->m_nEarlyR_Gain4 = 13384;
+ pPreset->m_nMaxExcursion = 127;
+ pPreset->m_nXfadeInterval = 6388;
+ pPreset->m_nAp0_ApGain = 15691;
+ pPreset->m_nAp0_ApOut = 711;
+ pPreset->m_nAp1_ApGain = 17999;
+ pPreset->m_nAp1_ApOut = 1113;
+ pPreset->m_rfu4 = 0;
+ pPreset->m_rfu5 = 0;
+ pPreset->m_rfu6 = 0;
+ pPreset->m_rfu7 = 0;
+ pPreset->m_rfu8 = 0;
+ pPreset->m_rfu9 = 0;
+ pPreset->m_rfu10 = 0;
+ }
+ else if (defaultPreset == 1)
+ {
+ pPreset->m_nLpfFbk = 6461;
+ pPreset->m_nLpfFwd = 14307;
+ pPreset->m_nEarly = 0;
+ pPreset->m_nWet = 27690;
+ pPreset->m_nDry = 32767;
+ pPreset->m_nEarlyL_LpfFbk = 3692;
+ pPreset->m_nEarlyL_LpfFwd = 29075;
+ pPreset->m_nEarlyL_Delay0 = 922;
+ pPreset->m_nEarlyL_Gain0 = 22152;
+ pPreset->m_nEarlyL_Delay1 = 1462;
+ pPreset->m_nEarlyL_Gain1 = 17537;
+ pPreset->m_nEarlyL_Delay2 = 0;
+ pPreset->m_nEarlyL_Gain2 = 14768;
+ pPreset->m_nEarlyL_Delay3 = 1221;
+ pPreset->m_nEarlyL_Gain3 = 14307;
+ pPreset->m_nEarlyL_Delay4 = 0;
+ pPreset->m_nEarlyL_Gain4 = 13384;
+ pPreset->m_nEarlyR_Delay0 = 502;
+ pPreset->m_nEarlyR_Gain0 = 20306;
+ pPreset->m_nEarlyR_Delay1 = 1762;
+ pPreset->m_nEarlyR_Gain1 = 17537;
+ pPreset->m_nEarlyR_Delay2 = 0;
+ pPreset->m_nEarlyR_Gain2 = 14768;
+ pPreset->m_nEarlyR_Delay3 = 0;
+ pPreset->m_nEarlyR_Gain3 = 16153;
+ pPreset->m_nEarlyR_Delay4 = 0;
+ pPreset->m_nEarlyR_Gain4 = 13384;
+ pPreset->m_nMaxExcursion = 127;
+ pPreset->m_nXfadeInterval = 6391;
+ pPreset->m_nAp0_ApGain = 15230;
+ pPreset->m_nAp0_ApOut = 708;
+ pPreset->m_nAp1_ApGain = 9692;
+ pPreset->m_nAp1_ApOut = 1113;
+ pPreset->m_rfu4 = 0;
+ pPreset->m_rfu5 = 0;
+ pPreset->m_rfu6 = 0;
+ pPreset->m_rfu7 = 0;
+ pPreset->m_rfu8 = 0;
+ pPreset->m_rfu9 = 0;
+ pPreset->m_rfu10 = 0;
+ }
+ else if (defaultPreset == 2)
+ {
+ pPreset->m_nLpfFbk = 5077;
+ pPreset->m_nLpfFwd = 12922;
+ pPreset->m_nEarly = 0;
+ pPreset->m_nWet = 24460;
+ pPreset->m_nDry = 32767;
+ pPreset->m_nEarlyL_LpfFbk = 3692;
+ pPreset->m_nEarlyL_LpfFwd = 29075;
+ pPreset->m_nEarlyL_Delay0 = 922;
+ pPreset->m_nEarlyL_Gain0 = 22152;
+ pPreset->m_nEarlyL_Delay1 = 1462;
+ pPreset->m_nEarlyL_Gain1 = 17537;
+ pPreset->m_nEarlyL_Delay2 = 0;
+ pPreset->m_nEarlyL_Gain2 = 14768;
+ pPreset->m_nEarlyL_Delay3 = 1221;
+ pPreset->m_nEarlyL_Gain3 = 14307;
+ pPreset->m_nEarlyL_Delay4 = 0;
+ pPreset->m_nEarlyL_Gain4 = 13384;
+ pPreset->m_nEarlyR_Delay0 = 502;
+ pPreset->m_nEarlyR_Gain0 = 20306;
+ pPreset->m_nEarlyR_Delay1 = 1762;
+ pPreset->m_nEarlyR_Gain1 = 17537;
+ pPreset->m_nEarlyR_Delay2 = 0;
+ pPreset->m_nEarlyR_Gain2 = 14768;
+ pPreset->m_nEarlyR_Delay3 = 0;
+ pPreset->m_nEarlyR_Gain3 = 16153;
+ pPreset->m_nEarlyR_Delay4 = 0;
+ pPreset->m_nEarlyR_Gain4 = 13384;
+ pPreset->m_nMaxExcursion = 127;
+ pPreset->m_nXfadeInterval = 6449;
+ pPreset->m_nAp0_ApGain = 15691;
+ pPreset->m_nAp0_ApOut = 774;
+ pPreset->m_nAp1_ApGain = 15691;
+ pPreset->m_nAp1_ApOut = 1113;
+ pPreset->m_rfu4 = 0;
+ pPreset->m_rfu5 = 0;
+ pPreset->m_rfu6 = 0;
+ pPreset->m_rfu7 = 0;
+ pPreset->m_rfu8 = 0;
+ pPreset->m_rfu9 = 0;
+ pPreset->m_rfu10 = 0;
+ }
+ else if (defaultPreset == 3)
+ {
+ pPreset->m_nLpfFbk = 5077;
+ pPreset->m_nLpfFwd = 11076;
+ pPreset->m_nEarly = 0;
+ pPreset->m_nWet = 23075;
+ pPreset->m_nDry = 32767;
+ pPreset->m_nEarlyL_LpfFbk = 3692;
+ pPreset->m_nEarlyL_LpfFwd = 29075;
+ pPreset->m_nEarlyL_Delay0 = 922;
+ pPreset->m_nEarlyL_Gain0 = 22152;
+ pPreset->m_nEarlyL_Delay1 = 1462;
+ pPreset->m_nEarlyL_Gain1 = 17537;
+ pPreset->m_nEarlyL_Delay2 = 0;
+ pPreset->m_nEarlyL_Gain2 = 14768;
+ pPreset->m_nEarlyL_Delay3 = 1221;
+ pPreset->m_nEarlyL_Gain3 = 14307;
+ pPreset->m_nEarlyL_Delay4 = 0;
+ pPreset->m_nEarlyL_Gain4 = 13384;
+ pPreset->m_nEarlyR_Delay0 = 502;
+ pPreset->m_nEarlyR_Gain0 = 20306;
+ pPreset->m_nEarlyR_Delay1 = 1762;
+ pPreset->m_nEarlyR_Gain1 = 17537;
+ pPreset->m_nEarlyR_Delay2 = 0;
+ pPreset->m_nEarlyR_Gain2 = 14768;
+ pPreset->m_nEarlyR_Delay3 = 0;
+ pPreset->m_nEarlyR_Gain3 = 16153;
+ pPreset->m_nEarlyR_Delay4 = 0;
+ pPreset->m_nEarlyR_Gain4 = 13384;
+ pPreset->m_nMaxExcursion = 127;
+ pPreset->m_nXfadeInterval = 6470; //6483;
+ pPreset->m_nAp0_ApGain = 14768;
+ pPreset->m_nAp0_ApOut = 792;
+ pPreset->m_nAp1_ApGain = 15783;
+ pPreset->m_nAp1_ApOut = 1113;
+ pPreset->m_rfu4 = 0;
+ pPreset->m_rfu5 = 0;
+ pPreset->m_rfu6 = 0;
+ pPreset->m_rfu7 = 0;
+ pPreset->m_rfu8 = 0;
+ pPreset->m_rfu9 = 0;
+ pPreset->m_rfu10 = 0;
+
+ }
+ }
+
+ return EAS_SUCCESS;
+}
diff --git a/arm-wt-22k/lib_src/eas_reverbdata.c b/arm-wt-22k/lib_src/eas_reverbdata.c new file mode 100644 index 0000000..5d48c1b --- /dev/null +++ b/arm-wt-22k/lib_src/eas_reverbdata.c @@ -0,0 +1,34 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverbdata.c
+ *
+ * Contents and purpose:
+ * Contains the static data allocation for the Reverb effect
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 550 $
+ * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_reverbdata.h"
+
+S_REVERB_OBJECT eas_ReverbData;
+
diff --git a/arm-wt-22k/lib_src/eas_reverbdata.h b/arm-wt-22k/lib_src/eas_reverbdata.h new file mode 100644 index 0000000..ef424da --- /dev/null +++ b/arm-wt-22k/lib_src/eas_reverbdata.h @@ -0,0 +1,486 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_reverbdata.h
+ *
+ * Contents and purpose:
+ * Contains the prototypes for the Reverb effect.
+ *
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 499 $
+ * $Date: 2006-12-11 16:07:20 -0800 (Mon, 11 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_REVERBDATA_H
+#define _EAS_REVERBDATA_H
+
+#include "eas_types.h"
+#include "eas_audioconst.h"
+
+/*------------------------------------
+ * defines
+ *------------------------------------
+*/
+
+/*
+CIRCULAR() calculates the array index using modulo arithmetic.
+The "trick" is that modulo arithmetic is simplified by masking
+the effective address where the mask is (2^n)-1. This only works
+if the buffer size is a power of two.
+*/
+#define CIRCULAR(base,offset,size) (EAS_U32)( \
+ ( \
+ ((EAS_I32)(base)) + ((EAS_I32)(offset)) \
+ ) \
+ & size \
+ )
+
+/* reverb parameters are updated every 2^(REVERB_UPDATE_PERIOD_IN_BITS) samples */
+#if defined (_SAMPLE_RATE_8000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 5
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 2048
+
+#elif defined (_SAMPLE_RATE_16000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 6
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 4096
+
+#elif defined (_SAMPLE_RATE_22050)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 7
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 4096
+
+#elif defined (_SAMPLE_RATE_32000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 7
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192
+
+#elif defined (_SAMPLE_RATE_44100)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 8
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192
+
+#elif defined (_SAMPLE_RATE_48000)
+
+#define REVERB_UPDATE_PERIOD_IN_BITS 8
+#define REVERB_BUFFER_SIZE_IN_SAMPLES 8192
+
+#endif
+
+// Define a mask for circular addressing, so that array index
+// can wraparound and stay in array boundary of 0, 1, ..., (buffer size -1)
+// The buffer size MUST be a power of two
+#define REVERB_BUFFER_MASK (REVERB_BUFFER_SIZE_IN_SAMPLES -1)
+
+#define REVERB_MAX_ROOM_TYPE 4 // any room numbers larger than this are invalid
+#define REVERB_MAX_NUM_REFLECTIONS 5 // max num reflections per channel
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */
+#define REVERB_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(0x1L << REVERB_UPDATE_PERIOD_IN_BITS)
+
+/*
+calculate the update counter by bitwise ANDING with this value to
+generate a 2^n modulo value
+*/
+#define REVERB_MODULO_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(REVERB_UPDATE_PERIOD_IN_SAMPLES -1)
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SECONDS seconds */
+#define REVERB_UPDATE_PERIOD_IN_SECONDS (REVERB_UPDATE_PERIOD_IN_SAMPLES / _OUTPUT_SAMPLE_RATE)
+
+// xfade parameters
+#define REVERB_XFADE_PERIOD_IN_SECONDS (100.0 / 1000.0) // xfade once every this many seconds
+
+#define REVERB_XFADE_PERIOD_IN_SAMPLES (REVERB_XFADE_PERIOD_IN_SECONDS * _OUTPUT_SAMPLE_RATE)
+
+#define REVERB_XFADE_PHASE_INCREMENT (EAS_I16)(65536 / ((EAS_I16)REVERB_XFADE_PERIOD_IN_SAMPLES/(EAS_I16)REVERB_UPDATE_PERIOD_IN_SAMPLES))
+
+/**********/
+/* the entire synth uses various flags in a bit field */
+
+/* if flag is set, synth reset has been requested */
+#define REVERB_FLAG_RESET_IS_REQUESTED 0x01 /* bit 0 */
+#define MASK_REVERB_RESET_IS_REQUESTED 0x01
+#define MASK_REVERB_RESET_IS_NOT_REQUESTED (EAS_U32)(~MASK_REVERB_RESET_IS_REQUESTED)
+
+/*
+by default, we always want to update ALL channel parameters
+when we reset the synth (e.g., during GM ON)
+*/
+#define DEFAULT_REVERB_FLAGS 0x0
+
+/* coefficients for generating sin, cos */
+#define REVERB_PAN_G2 4294940151 /* -0.82842712474619 = 2 - 4/sqrt(2) */
+/*
+EAS_I32 nPanG1 = +1.0 for sin
+EAS_I32 nPanG1 = -1.0 for cos
+*/
+#define REVERB_PAN_G0 23170 /* 0.707106781186547 = 1/sqrt(2) */
+
+/*************************************************************/
+// define the input injection points
+#define GUARD 5 // safety guard of this many samples
+
+#define MAX_AP_TIME (double) (20.0/1000.0) // delay time in milliseconds
+#define MAX_DELAY_TIME (double) (65.0/1000.0) // delay time in milliseconds
+
+#define MAX_AP_SAMPLES (int)(((double) MAX_AP_TIME) * ((double) _OUTPUT_SAMPLE_RATE))
+#define MAX_DELAY_SAMPLES (int)(((double) MAX_DELAY_TIME) * ((double) _OUTPUT_SAMPLE_RATE))
+
+#define AP0_IN 0
+#define AP1_IN (AP0_IN + MAX_AP_SAMPLES + GUARD)
+#define DELAY0_IN (AP1_IN + MAX_AP_SAMPLES + GUARD)
+#define DELAY1_IN (DELAY0_IN + MAX_DELAY_SAMPLES + GUARD)
+
+// Define the max offsets for the end points of each section
+// i.e., we don't expect a given section's taps to go beyond
+// the following limits
+#define AP0_OUT (AP0_IN + MAX_AP_SAMPLES -1)
+#define AP1_OUT (AP1_IN + MAX_AP_SAMPLES -1)
+#define DELAY0_OUT (DELAY0_IN + MAX_DELAY_SAMPLES -1)
+#define DELAY1_OUT (DELAY1_IN + MAX_DELAY_SAMPLES -1)
+
+#define REVERB_DEFAULT_ROOM_NUMBER 1 // default preset number
+#define DEFAULT_AP0_LENGTH (int)(((double) (17.0/1000.0)) * ((double) _OUTPUT_SAMPLE_RATE))
+#define DEFAULT_AP0_GAIN 19400
+#define DEFAULT_AP1_LENGTH (int)(((double) (16.5/1000.0)) * ((double) _OUTPUT_SAMPLE_RATE))
+#define DEFAULT_AP1_GAIN -19400
+
+#define REVERB_DEFAULT_WET 32767
+#define REVERB_DEFAULT_DRY 0
+
+#define EAS_REVERB_WET_MAX 32767
+#define EAS_REVERB_WET_MIN 0
+#define EAS_REVERB_DRY_MAX 32767
+#define EAS_REVERB_DRY_MIN 0
+
+/* parameters for each allpass */
+typedef struct
+{
+ EAS_U16 m_zApOut; // delay offset for ap out
+
+ EAS_I16 m_nApGain; // gain for ap
+
+ EAS_U16 m_zApIn; // delay offset for ap in
+
+} S_ALLPASS_OBJECT;
+
+
+/* parameters for each allpass */
+typedef struct
+{
+ EAS_PCM m_zLpf; // actual state variable, not a length
+
+ EAS_I16 m_nLpfFwd; // lpf forward gain
+
+ EAS_I16 m_nLpfFbk; // lpf feedback gain
+
+ EAS_U16 m_zDelay[REVERB_MAX_NUM_REFLECTIONS]; // delay offset for ap out
+
+ EAS_I16 m_nGain[REVERB_MAX_NUM_REFLECTIONS]; // gain for ap
+
+} S_EARLY_REFLECTION_OBJECT;
+
+//demo
+typedef struct
+{
+ EAS_I16 m_nLpfFbk;
+ EAS_I16 m_nLpfFwd;
+
+ EAS_I16 m_nEarly;
+ EAS_I16 m_nWet;
+ EAS_I16 m_nDry;
+
+ EAS_I16 m_nEarlyL_LpfFbk;
+ EAS_I16 m_nEarlyL_LpfFwd;
+
+ EAS_I16 m_nEarlyL_Delay0; //8
+ EAS_I16 m_nEarlyL_Gain0;
+ EAS_I16 m_nEarlyL_Delay1;
+ EAS_I16 m_nEarlyL_Gain1;
+ EAS_I16 m_nEarlyL_Delay2;
+ EAS_I16 m_nEarlyL_Gain2;
+ EAS_I16 m_nEarlyL_Delay3;
+ EAS_I16 m_nEarlyL_Gain3;
+ EAS_I16 m_nEarlyL_Delay4;
+ EAS_I16 m_nEarlyL_Gain4;
+
+ EAS_I16 m_nEarlyR_Delay0; //18
+ EAS_I16 m_nEarlyR_Gain0;
+ EAS_I16 m_nEarlyR_Delay1;
+ EAS_I16 m_nEarlyR_Gain1;
+ EAS_I16 m_nEarlyR_Delay2;
+ EAS_I16 m_nEarlyR_Gain2;
+ EAS_I16 m_nEarlyR_Delay3;
+ EAS_I16 m_nEarlyR_Gain3;
+ EAS_I16 m_nEarlyR_Delay4;
+ EAS_I16 m_nEarlyR_Gain4;
+
+ EAS_U16 m_nMaxExcursion; //28
+ EAS_I16 m_nXfadeInterval;
+
+ EAS_I16 m_nAp0_ApGain; //30
+ EAS_I16 m_nAp0_ApOut;
+ EAS_I16 m_nAp1_ApGain;
+ EAS_I16 m_nAp1_ApOut;
+
+ EAS_I16 m_rfu4;
+ EAS_I16 m_rfu5;
+ EAS_I16 m_rfu6;
+ EAS_I16 m_rfu7;
+ EAS_I16 m_rfu8;
+ EAS_I16 m_rfu9;
+ EAS_I16 m_rfu10; //43
+
+} S_REVERB_PRESET;
+
+typedef struct
+{
+ S_REVERB_PRESET m_sPreset[REVERB_MAX_ROOM_TYPE]; //array of presets
+
+} S_REVERB_PRESET_BANK;
+
+/* parameters for each reverb */
+typedef struct
+{
+ /* controls entire reverb playback volume */
+ /* to conserve memory, use the MSB and ignore the LSB */
+ EAS_U8 m_nMasterVolume;
+
+ /* update counter keeps track of when synth params need updating */
+ /* only needs to be as large as REVERB_UPDATE_PERIOD_IN_SAMPLES */
+ EAS_I16 m_nUpdateCounter;
+
+ EAS_U16 m_nMinSamplesToAdd; /* ComputeReverb() generates this many samples */
+
+ EAS_U8 m_nFlags; /* misc flags/bit fields */
+
+ EAS_PCM *m_pOutputBuffer;
+ EAS_PCM *m_pInputBuffer;
+
+ EAS_U16 m_nNumSamplesInOutputBuffer;
+ EAS_U16 m_nNumSamplesInInputBuffer;
+
+ EAS_U16 m_nNumInputSamplesRead; // if m_nNumInputSamplesRead >= NumSamplesInInputBuffer
+ // then get a new input buffer
+ EAS_PCM *m_pNextInputSample;
+
+ EAS_U16 m_nBaseIndex; // base index for circular buffer
+
+ // reverb delay line offsets, allpass parameters, etc:
+
+ EAS_PCM m_nRevOutFbkR; // combine feedback reverb right out with dry left in
+
+ S_ALLPASS_OBJECT m_sAp0; // allpass 0 (left channel)
+
+ EAS_U16 m_zD0In; // delay offset for delay line D0 in
+
+ EAS_PCM m_nRevOutFbkL; // combine feedback reverb left out with dry right in
+
+ S_ALLPASS_OBJECT m_sAp1; // allpass 1 (right channel)
+
+ EAS_U16 m_zD1In; // delay offset for delay line D1 in
+
+ // delay output taps, notice criss cross order
+ EAS_U16 m_zD0Self; // self feeds forward d0 --> d0
+
+ EAS_U16 m_zD1Cross; // cross feeds across d1 --> d0
+
+ EAS_PCM m_zLpf0; // actual state variable, not a length
+
+ EAS_U16 m_zD1Self; // self feeds forward d1 --> d1
+
+ EAS_U16 m_zD0Cross; // cross feeds across d0 --> d1
+
+ EAS_PCM m_zLpf1; // actual state variable, not a length
+
+ EAS_I16 m_nSin; // gain for self taps
+
+ EAS_I16 m_nCos; // gain for cross taps
+
+ EAS_I16 m_nSinIncrement; // increment for gain
+
+ EAS_I16 m_nCosIncrement; // increment for gain
+
+ EAS_I16 m_nLpfFwd; // lpf forward gain (includes scaling for mixer)
+
+ EAS_I16 m_nLpfFbk; // lpf feedback gain
+
+ EAS_U16 m_nXfadeInterval; // update/xfade after this many samples
+
+ EAS_U16 m_nXfadeCounter; // keep track of when to xfade
+
+ EAS_I16 m_nPhase; // -1 <= m_nPhase < 1
+ // but during sin,cos calculations
+ // use m_nPhase/2
+
+ EAS_I16 m_nPhaseIncrement; // add this to m_nPhase each frame
+
+ EAS_I16 m_nNoise; // random noise sample
+
+ EAS_U16 m_nMaxExcursion; // the taps can excurse +/- this amount
+
+ EAS_BOOL m_bUseNoise; // if EAS_TRUE, use noise as input signal
+
+ EAS_BOOL m_bBypass; // if EAS_TRUE, then bypass reverb and copy input to output
+
+ EAS_I16 m_nCurrentRoom; // preset number for current room
+
+ EAS_I16 m_nNextRoom; // preset number for next room
+
+ EAS_I16 m_nWet; // gain for wet (processed) signal
+
+ EAS_I16 m_nDry; // gain for dry (unprocessed) signal
+
+ EAS_I16 m_nEarly; // gain for early (widen) signal
+
+ S_EARLY_REFLECTION_OBJECT m_sEarlyL; // left channel early reflections
+ S_EARLY_REFLECTION_OBJECT m_sEarlyR; // right channel early reflections
+
+ EAS_PCM m_nDelayLine[REVERB_BUFFER_SIZE_IN_SAMPLES]; // one large delay line for all reverb elements
+
+ S_REVERB_PRESET pPreset;
+
+ S_REVERB_PRESET_BANK m_sPreset;
+
+ //EAS_I8 preset;
+
+} S_REVERB_OBJECT;
+
+
+/*------------------------------------
+ * prototypes
+ *------------------------------------
+*/
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateXfade
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the xfade parameters as required
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to write to buffer
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - xfade parameters will be changed
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbUpdateXfade(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateNoise
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a noise sample and limit its value
+ *
+ * Inputs:
+ * nMaxExcursion - noise value is limited to this value
+ * pnNoise - return new noise sample in this (not limited)
+ *
+ * Outputs:
+ * new limited noise value
+ *
+ * Side Effects:
+ * - *pnNoise noise value is updated
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_U16 ReverbCalculateNoise(EAS_U16 nMaxExcursion, EAS_I16 *pnNoise);
+
+/*----------------------------------------------------------------------------
+ * ReverbCalculateSinCos
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate a new sin and cosine value based on the given phase
+ *
+ * Inputs:
+ * nPhase - phase angle
+ * pnSin - input old value, output new value
+ * pnCos - input old value, output new value
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - *pnSin, *pnCos are updated
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbCalculateSinCos(EAS_I16 nPhase, EAS_I16 *pnSin, EAS_I16 *pnCos);
+
+/*----------------------------------------------------------------------------
+ * Reverb
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * apply reverb to the given signal
+ *
+ * Inputs:
+ * nNu
+ * pnSin - input old value, output new value
+ * pnCos - input old value, output new value
+ *
+ * Outputs:
+ * number of samples actually reverberated
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT Reverb(S_REVERB_OBJECT* pReverbData, EAS_INT nNumSamplesToAdd, EAS_PCM *pOutputBuffer, EAS_PCM *pInputBuffer);
+
+/*----------------------------------------------------------------------------
+ * ReverbReadInPresets()
+ *----------------------------------------------------------------------------
+ * Purpose: sets global reverb preset bank to defaults
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbReadInPresets(S_REVERB_OBJECT* pReverbData);
+
+
+/*----------------------------------------------------------------------------
+ * ReverbUpdateRoom
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the room's preset parameters as required
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * - reverb paramters (fbk, fwd, etc) will be changed
+ * - m_nCurrentRoom := m_nNextRoom
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT ReverbUpdateRoom(S_REVERB_OBJECT* pReverbData);
+
+#endif /* #ifndef _EAS_REVERBDATA_H */
+
+
diff --git a/arm-wt-22k/lib_src/eas_rtttl.c b/arm-wt-22k/lib_src/eas_rtttl.c new file mode 100644 index 0000000..486ad60 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_rtttl.c @@ -0,0 +1,1197 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_rtttl.c
+ *
+ * Contents and purpose:
+ * RTTTL parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 795 $
+ * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_miditypes.h"
+#include "eas_parser.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_midi.h"
+#include "eas_config.h"
+#include "eas_vm_protos.h"
+#include "eas_rtttldata.h"
+#include "eas_ctype.h"
+
+/* increase gain for mono ringtones */
+#define RTTTL_GAIN_OFFSET 8
+
+/* maximum title length including colon separator */
+#define RTTTL_MAX_TITLE_LEN 32
+#define RTTTL_INFINITE_LOOP 15
+
+/* length of 32nd note in 1/256ths of a msec for 63 BPM tempo */
+#define DEFAULT_TICK_CONV 30476
+#define TICK_CONVERT 1920000
+
+/* default channel and program for RTTTL playback */
+#define RTTTL_CHANNEL 0
+#define RTTTL_PROGRAM 80
+#define RTTTL_VELOCITY 127
+
+/* note used for rest */
+#define RTTTL_REST 1
+
+/* multiplier for fixed point triplet conversion */
+#define TRIPLET_MULTIPLIER 683
+#define TRIPLET_SHIFT 10
+
+/* local prototypes */
+static EAS_RESULT RTTTL_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+static EAS_RESULT RTTTL_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT RTTTL_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+static EAS_RESULT RTTTL_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT RTTTL_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT RTTTL_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT RTTTL_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT RTTTL_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT RTTTL_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT RTTTL_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT RTTTL_GetStyle (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData);
+static EAS_RESULT RTTTL_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pDuration);
+static EAS_RESULT RTTTL_GetOctave (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_U8 *pOctave);
+static EAS_RESULT RTTTL_GetTempo (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData);
+static EAS_RESULT RTTTL_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I32 *pValue);
+static EAS_RESULT RTTTL_ParseHeader (S_EAS_DATA *pEASData, S_RTTTL_DATA* pData, EAS_BOOL metaData);
+static EAS_RESULT RTTTL_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue);
+static EAS_RESULT RTTTL_PeekNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue);
+
+/* inline functions */
+EAS_INLINE void RTTTL_PutBackChar (S_RTTTL_DATA *pData, EAS_I8 value) { pData->dataByte = value; }
+
+
+/* lookup table for note values */
+static const EAS_U8 noteTable[] = { 21, 23, 12, 14, 16, 17, 19, 23 };
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_RTTTL_Parser
+ *
+ * This structure contains the functional interface for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_RTTTL_Parser =
+{
+ RTTTL_CheckFileType,
+ RTTTL_Prepare,
+ RTTTL_Time,
+ RTTTL_Event,
+ RTTTL_State,
+ RTTTL_Close,
+ RTTTL_Reset,
+ RTTTL_Pause,
+ RTTTL_Resume,
+ NULL,
+ RTTTL_SetData,
+ RTTTL_GetData,
+ NULL
+};
+
+/*----------------------------------------------------------------------------
+ * RTTTL_CheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset)
+{
+ S_RTTTL_DATA data;
+ S_RTTTL_DATA *pData;
+
+ /* see if we can parse the header */
+ data.fileHandle = fileHandle;
+ data.fileOffset = offset;
+ *ppHandle= NULL;
+ if (RTTTL_ParseHeader (pEASData, &data, EAS_FALSE) == EAS_SUCCESS)
+ {
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pData = EAS_CMEnumData(EAS_CM_RTTTL_DATA);
+ else
+ pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_RTTTL_DATA));
+ if (!pData)
+ return EAS_ERROR_MALLOC_FAILED;
+ EAS_HWMemSet(pData, 0, sizeof(S_RTTTL_DATA));
+
+ /* return a pointer to the instance data */
+ pData->fileHandle = fileHandle;
+ pData->fileOffset = offset;
+ pData->state = EAS_STATE_OPEN;
+ *ppHandle = pData;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Prepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_RTTTL_DATA* pData;
+ EAS_RESULT result;
+
+ /* check for valid state */
+ pData = (S_RTTTL_DATA*) pInstData;
+ if (pData->state != EAS_STATE_OPEN)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* instantiate a synthesizer */
+ if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ }
+ return result;
+ }
+
+ pData->state = EAS_STATE_ERROR;
+ if ((result = RTTTL_ParseHeader (pEASData, pData, (EAS_BOOL) (pData->metadata.callback != NULL))) != EAS_SUCCESS)
+ {
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pData);
+ return result;
+ }
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Time()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the time of the next event in msecs
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pTime - pointer to variable to hold time of next event (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT RTTTL_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime)
+{
+ S_RTTTL_DATA *pData;
+
+ pData = (S_RTTTL_DATA*) pInstData;
+
+ /* return time in milliseconds */
+ /*lint -e{704} use shift instead of division */
+ *pTime = pData->time >> 8;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Event()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the next event in the file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode)
+{
+ S_RTTTL_DATA* pData;
+ EAS_RESULT result;
+ EAS_I32 ticks;
+ EAS_I32 temp;
+ EAS_I8 c;
+ EAS_U8 note;
+ EAS_U8 octave;
+
+ pData = (S_RTTTL_DATA*) pInstData;
+ if (pData->state >= EAS_STATE_OPEN)
+ return EAS_SUCCESS;
+
+ /* initialize MIDI channel when the track starts playing */
+ if (pData->time == 0)
+ {
+ /* set program to square lead */
+ VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, RTTTL_PROGRAM);
+
+ /* set channel volume to max */
+ VMControlChange(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, 7, 127);
+ }
+
+ /* check for end of note */
+ if (pData->note)
+ {
+ /* stop the note */
+ VMStopNote(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, pData->note, 0);
+ pData->note = 0;
+
+ /* check for rest between notes */
+ if (pData->restTicks)
+ {
+ pData->time += pData->restTicks;
+ pData->restTicks = 0;
+ return EAS_SUCCESS;
+ }
+ }
+
+ /* parse the next event */
+ octave = pData->octave;
+ note = 0;
+ ticks = pData->duration * pData->tick;
+ for (;;)
+ {
+
+ /* get next character */
+ if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS)
+ {
+ if (result != EAS_EOF)
+ return result;
+
+ /* end of file, if no notes to process, check for looping */
+ if (!note)
+ {
+ /* if no loop set state to stopping */
+ if (pData->repeatCount == 0)
+ {
+ pData->state = EAS_STATE_STOPPING;
+ VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ return EAS_SUCCESS;
+ }
+
+ /* decrement loop count */
+ if (pData->repeatCount != RTTTL_INFINITE_LOOP)
+ pData->repeatCount--;
+
+ /* if locating, ignore infinite loops */
+ else if (parserMode != eParserModePlay)
+ {
+ pData->state = EAS_STATE_STOPPING;
+ VMReleaseAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ return EAS_SUCCESS;
+ }
+
+ /* loop back to start of notes */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->repeatOffset)) != EAS_SUCCESS)
+ return result;
+ continue;
+ }
+
+ /* still have a note to process */
+ else
+ c = ',';
+ }
+
+ /* bpm */
+ if (c == 'b')
+ {
+ /* peek at next character */
+ if ((result = RTTTL_PeekNextChar(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS)
+ return result;
+
+ /* if a number, must be octave or tempo */
+ if (IsDigit(c))
+ {
+ if ((result = RTTTL_GetNumber(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for octave first */
+ if ((temp >= 4) && (temp <= 7))
+ {
+ octave = (EAS_U8) temp;
+ }
+
+ /* check for tempo */
+ else if ((temp >= 25) && (temp <= 900))
+ {
+ pData->tick = TICK_CONVERT / (EAS_U32) temp;
+ }
+
+ /* don't know what it was */
+ else
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* must be a note */
+ else
+ {
+ note = noteTable[1];
+ }
+ }
+
+ /* octave */
+ else if (c == 'o')
+ {
+ if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &pData->octave)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* style */
+ else if (c == 's')
+ {
+ if ((result = RTTTL_GetStyle(pEASData->hwInstData, pData)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* duration or octave */
+ else if (IsDigit(c))
+ {
+ RTTTL_PutBackChar(pData, c);
+
+ /* duration comes before note */
+ if (!note)
+ {
+ if ((result = RTTTL_GetDuration(pEASData->hwInstData, pData, &c)) != EAS_SUCCESS)
+ return result;
+ ticks = c * pData->tick;
+ }
+
+ /* octave comes after note */
+ else
+ {
+ if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &octave)) != EAS_SUCCESS)
+ return result;
+ }
+ }
+
+ /* note or rest */
+ else if ((c >= 'a') && (c <= 'h'))
+ {
+ note = noteTable[c - 'a'];
+ }
+
+ else if (c == 'p')
+ {
+ note = RTTTL_REST;
+ }
+
+ /* dotted note */
+ else if (c == '.')
+ {
+ /*lint -e{704} shift for performance */
+ ticks += ticks >> 1;
+ }
+
+ /* accidental */
+ else if (c == '#')
+ {
+ if (note)
+ note++;
+ }
+
+ /* end of event */
+ else if ((c == ',') && note)
+ {
+
+ /* handle note events */
+ if (note != RTTTL_REST)
+ {
+
+ /* save note and start it */
+ pData->note = note + octave;
+ if (parserMode == eParserModePlay)
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, RTTTL_CHANNEL, pData->note, RTTTL_VELOCITY);
+
+ /* determine note length */
+ switch (pData->style)
+ {
+ /* natural */
+ case 'n':
+ /*lint -e{704} shift for performance */
+ pData->restTicks = ticks >> 4;
+ break;
+ /* continuous */
+
+ case 'c':
+ pData->restTicks = 0;
+ break;
+
+ /* staccato */
+ case 's':
+ /*lint -e{704} shift for performance */
+ pData->restTicks = ticks >> 1;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "RTTTL_Event: Unexpected style type %c\n", pData->style); */ }
+ break;
+ }
+
+ /* next event is at end of this note */
+ pData->time += ticks - pData->restTicks;
+ }
+
+ /* rest */
+ else
+ pData->time += ticks;
+
+ /* event found, return to caller */
+ break;
+ }
+ }
+
+ pData->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_State()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT RTTTL_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState)
+{
+ S_RTTTL_DATA* pData;
+
+ /* establish pointer to instance data */
+ pData = (S_RTTTL_DATA*) pInstData;
+
+ /* if stopping, check to see if synth voices are active */
+ if (pData->state == EAS_STATE_STOPPING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_STOPPED;
+ }
+
+ if (pData->state == EAS_STATE_PAUSING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_PAUSED;
+ }
+
+ /* return current state */
+ *pState = pData->state;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Close()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_RTTTL_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_RTTTL_DATA*) pInstData;
+
+ /* close the file */
+ if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS)
+ return result;
+
+ /* free the synth */
+ if (pData->pSynth != NULL)
+ VMMIDIShutdown(pEASData, pData->pSynth);
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pData);
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_RTTTL_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_RTTTL_DATA*) pInstData;
+
+ /* reset the synth */
+ VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE);
+
+ /* reset time to zero */
+ pData->time = 0;
+ pData->note = 0;
+
+ /* reset file position and re-parse header */
+ pData->state = EAS_STATE_ERROR;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+ if ((result = RTTTL_ParseHeader (pEASData, pData, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Pause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Pauses the sequencer. Mutes all voices and sets state to pause.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_RTTTL_DATA *pData;
+
+ /* can't pause a stopped stream */
+ pData = (S_RTTTL_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* mute the synthesizer */
+ VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ pData->state = EAS_STATE_PAUSING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_Resume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume playing after a pause, sets state back to playing.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT RTTTL_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_RTTTL_DATA *pData;
+
+ /* can't resume a stopped stream */
+ pData = (S_RTTTL_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* nothing to do but resume playback */
+ pData->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_SetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT RTTTL_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_RTTTL_DATA *pData;
+
+ pData = (S_RTTTL_DATA *) pInstData;
+ switch (param)
+ {
+
+ /* set metadata callback */
+ case PARSER_DATA_METADATA_CB:
+ EAS_HWMemCpy(&pData->metadata, (void*) value, sizeof(S_METADATA_CB));
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT RTTTL_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_RTTTL_DATA *pData;
+
+ pData = (S_RTTTL_DATA *) pInstData;
+ switch (param)
+ {
+ /* return file type as RTTTL */
+ case PARSER_DATA_FILE_TYPE:
+ *pValue = EAS_FILE_RTTTL;
+ break;
+
+#if 0
+ /* set transposition */
+ case PARSER_DATA_TRANSPOSITION:
+ *pValue = pData->transposition;
+ break;
+#endif
+
+ case PARSER_DATA_SYNTH_HANDLE:
+ *pValue = (EAS_I32) pData->pSynth;
+ break;
+
+ case PARSER_DATA_GAIN_OFFSET:
+ *pValue = RTTTL_GAIN_OFFSET;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetStyle()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetStyle (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData)
+{
+ EAS_RESULT result;
+ EAS_I8 style;
+
+ /* get style */
+ if ((result = RTTTL_GetNextChar(hwInstData, pData, &style)) != EAS_SUCCESS)
+ return result;
+
+ if ((style != 's') && (style != 'n') && (style != 'c'))
+ return EAS_ERROR_FILE_FORMAT;
+
+ pData->style = style;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetDuration()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetDuration (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pDuration)
+{
+ EAS_RESULT result;
+ EAS_I32 duration;
+ EAS_I8 temp;
+
+ /* get the duration */
+ if ((result = RTTTL_GetNumber(hwInstData, pData, &duration)) != EAS_SUCCESS)
+ return result;
+
+ if ((duration != 1) && (duration != 2) && (duration != 4) && (duration != 8) && (duration != 16) && (duration != 32))
+ return EAS_ERROR_FILE_FORMAT;
+
+ temp = 64;
+ while (duration)
+ {
+ /*lint -e{704} shift for performance */
+ duration = duration >> 1;
+ /*lint -e{702} use shift for performance */
+ temp = temp >> 1;
+ }
+
+ *pDuration = temp;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetOctave()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetOctave (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_U8 *pOctave)
+{
+ EAS_RESULT result;
+ EAS_I32 octave;
+
+ /* get the tempo */
+ if ((result = RTTTL_GetNumber(hwInstData, pData, &octave)) != EAS_SUCCESS)
+ return result;
+
+ if ((octave < 4) || (octave > 7))
+ return EAS_ERROR_FILE_FORMAT;
+
+ *pOctave = (EAS_U8) (octave * 12);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetTempo()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetTempo (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData)
+{
+ EAS_RESULT result;
+ EAS_I32 tempo;
+
+ /* get the tempo */
+ if ((result = RTTTL_GetNumber(hwInstData, pData, &tempo)) != EAS_SUCCESS)
+ return result;
+
+ if ((tempo < 25) || (tempo > 900))
+ return EAS_ERROR_FILE_FORMAT;
+
+ pData->tick = TICK_CONVERT / (EAS_U32) tempo;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetNumber()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetNumber (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I32 *pValue)
+{
+ EAS_RESULT result;
+ EAS_INT temp;
+ EAS_I8 c;
+
+ *pValue = -1;
+ temp = 0;
+ for (;;)
+ {
+ if ((result = RTTTL_PeekNextChar(hwInstData, pData, &c)) != EAS_SUCCESS)
+ {
+ if ((result == EAS_EOF) && (*pValue != -1))
+ return EAS_SUCCESS;
+ return result;
+ }
+
+ if (IsDigit(c))
+ {
+ pData->dataByte = 0;
+ temp = temp * 10 + c - '0';
+ *pValue = temp;
+ }
+ else
+ return EAS_SUCCESS;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_ParseHeader()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_ParseHeader (S_EAS_DATA *pEASData, S_RTTTL_DATA* pData, EAS_BOOL metaData)
+{
+ EAS_RESULT result;
+ EAS_I32 i;
+ EAS_I8 temp;
+ EAS_I8 control;
+
+ /* initialize some defaults */
+ pData->time = 0;
+ pData->tick = DEFAULT_TICK_CONV;
+ pData->note = 0;
+ pData->duration = 4;
+ pData ->restTicks = 0;
+ pData->octave = 60;
+ pData->repeatOffset = -1;
+ pData->repeatCount = 0;
+ pData->style = 'n';
+ pData->dataByte = 0;
+
+ metaData = metaData && (pData->metadata.buffer != NULL) && (pData->metadata.callback != NULL);
+
+ /* seek to start of data */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* zero the metadata buffer */
+ if (metaData)
+ EAS_HWMemSet(pData->metadata.buffer, 0, pData->metadata.bufferSize);
+
+ /* read the title */
+ for (i = 0; i < RTTTL_MAX_TITLE_LEN; i++)
+ {
+ if ((result = EAS_HWGetByte(pEASData->hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS)
+ return result;
+
+ if (temp == ':')
+ break;
+
+ /* pass along metadata */
+ if (metaData)
+ {
+ if (i < (pData->metadata.bufferSize- 1))
+ pData->metadata.buffer[i] = (char) temp;
+ }
+ }
+
+ /* check for error in title */
+ if (i == RTTTL_MAX_TITLE_LEN)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* pass along metadata */
+ if (metaData)
+ (*pData->metadata.callback)(EAS_METADATA_TITLE, pData->metadata.buffer, pData->metadata.pUserData);
+
+ /* control fields */
+ for (;;)
+ {
+
+ /* get control type */
+ if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &control)) != EAS_SUCCESS)
+ return result;
+
+ /* next char should be equal sign */
+ if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+ if (temp != '=')
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* get the control value */
+ switch (control)
+ {
+
+ /* bpm */
+ case 'b':
+ if ((result = RTTTL_GetTempo(pEASData->hwInstData, pData)) != EAS_SUCCESS)
+ return result;
+ break;
+
+ /* duration */
+ case 'd':
+ if ((result = RTTTL_GetDuration(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+ pData->duration = temp;
+ break;
+
+ /* loop */
+ case 'l':
+ if ((result = RTTTL_GetNumber(pEASData->hwInstData, pData, &i)) != EAS_SUCCESS)
+ return result;
+ if ((i < 0) || (i > 15))
+ return EAS_ERROR_FILE_FORMAT;
+ pData->repeatCount = (EAS_U8) i;
+ break;
+
+ /* octave */
+ case 'o':
+ if ((result = RTTTL_GetOctave(pEASData->hwInstData, pData, &pData->octave)) != EAS_SUCCESS)
+ return result;
+ break;
+
+ /* get style */
+ case 's':
+ if ((result = RTTTL_GetStyle(pEASData->hwInstData, pData)) != EAS_SUCCESS)
+ return result;
+ break;
+
+ /* unrecognized control */
+ default:
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* next character should be comma or colon */
+ if ((result = RTTTL_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for end of control field */
+ if (temp == ':')
+ break;
+
+ /* must be a comma */
+ if (temp != ',')
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* should be at the start of the music block */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, pData->fileHandle, &pData->repeatOffset)) != EAS_SUCCESS)
+ return result;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_GetNextChar()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue)
+{
+ EAS_RESULT result;
+ EAS_I8 temp;
+
+ *pValue = 0;
+ for(;;)
+ {
+
+ /* check for character that has been put back */
+ if (pData->dataByte)
+ {
+ temp = pData->dataByte;
+ pData->dataByte = 0;
+ }
+ else
+ {
+ if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &temp)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* ignore white space */
+ if (!IsSpace(temp))
+ {
+ *pValue = ToLower(temp);
+ return EAS_SUCCESS;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * RTTTL_PeekNextChar()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ *
+ * Inputs:
+ *
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT RTTTL_PeekNextChar (EAS_HW_DATA_HANDLE hwInstData, S_RTTTL_DATA *pData, EAS_I8 *pValue)
+{
+ EAS_RESULT result;
+ EAS_I8 temp;
+
+ *pValue = 0;
+ for(;;)
+ {
+
+ /* read a character from the file, if necessary */
+ if (!pData->dataByte)
+ {
+ if ((result = EAS_HWGetByte(hwInstData, pData->fileHandle, &pData->dataByte)) != EAS_SUCCESS)
+ return result;
+
+ }
+ temp = pData->dataByte;
+
+ /* ignore white space */
+ if (!IsSpace(temp))
+ {
+ *pValue = ToLower(temp);
+ return EAS_SUCCESS;
+ }
+ pData->dataByte = 0;
+ }
+}
+
diff --git a/arm-wt-22k/lib_src/eas_rtttldata.c b/arm-wt-22k/lib_src/eas_rtttldata.c new file mode 100644 index 0000000..7a500bd --- /dev/null +++ b/arm-wt-22k/lib_src/eas_rtttldata.c @@ -0,0 +1,41 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_rtttldata.c
+ *
+ * Contents and purpose:
+ * RTTTL File Parser data module for static memory models
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_rtttldata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_RTTTLData
+ *
+ * Static memory allocation for RTTTL parser
+ *----------------------------------------------------------------------------
+*/
+S_RTTTL_DATA eas_RTTTLData;
+
diff --git a/arm-wt-22k/lib_src/eas_rtttldata.h b/arm-wt-22k/lib_src/eas_rtttldata.h new file mode 100644 index 0000000..bf4c38b --- /dev/null +++ b/arm-wt-22k/lib_src/eas_rtttldata.h @@ -0,0 +1,70 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_rtttldata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data declarations for the RTTTL parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_RTTTLDATA_H
+#define EAS_RTTTLDATA_H
+
+#include "eas_data.h"
+
+
+/* maximum line size as specified in iMelody V1.2 spec */
+#define MAX_LINE_SIZE 75
+
+/*----------------------------------------------------------------------------
+ *
+ * S_RTTTL_DATA
+ *
+ * This structure contains the state data for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+ EAS_FILE_HANDLE fileHandle; /* file handle */
+ S_SYNTH *pSynth; /* synthesizer handle */
+ S_METADATA_CB metadata; /* metadata callback */
+ EAS_I32 fileOffset; /* offset to start of data */
+ EAS_I32 time; /* current time in 256ths of a msec */
+ EAS_I32 tick; /* length of 32nd note in 256th of a msec */
+ EAS_I32 restTicks; /* ticks to rest after current note */
+ EAS_I32 repeatOffset; /* file offset to start of repeat section */
+ EAS_U8 repeatCount; /* repeat counter */
+ EAS_I8 dataByte; /* storage for characters that are "put back" */
+ EAS_U8 state; /* current state EAS_STATE_XXXX */
+ EAS_I8 style; /* from STYLE */
+ EAS_U8 note; /* MIDI note number */
+ EAS_U8 octave; /* decault octave prefix */
+ EAS_I8 duration; /* default note duration */
+} S_RTTTL_DATA;
+
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_smf.c b/arm-wt-22k/lib_src/eas_smf.c new file mode 100644 index 0000000..af02c93 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_smf.c @@ -0,0 +1,1207 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_smf.c + * + * Contents and purpose: + * SMF Type 0 and 1 File Parser + * + * For SMF timebase analysis, see "MIDI Sequencer Analysis.xls". + * + * Copyright Sonic Network Inc. 2005 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 803 $ + * $Date: 2007-08-01 09:57:00 -0700 (Wed, 01 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +#include "eas_data.h" +#include "eas_miditypes.h" +#include "eas_parser.h" +#include "eas_report.h" +#include "eas_host.h" +#include "eas_midi.h" +#include "eas_config.h" +#include "eas_vm_protos.h" +#include "eas_smfdata.h" +#include "eas_smf.h" + +#ifdef JET_INTERFACE +#include "jet_data.h" +#endif + +//3 dls: The timebase for this module is adequate to keep MIDI and +//3 digital audio synchronized for only a few minutes. It should be +//3 sufficient for most mobile applications. If better accuracy is +//3 required, more fractional bits should be added to the timebase. + +static const EAS_U8 smfHeader[] = { 'M', 'T', 'h', 'd' }; + +/* local prototypes */ +static EAS_RESULT SMF_GetVarLenData (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_U32 *pData); +static EAS_RESULT SMF_ParseMetaEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream); +static EAS_RESULT SMF_ParseSysEx (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_U8 f0, EAS_INT parserMode); +static EAS_RESULT SMF_ParseEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_INT parserMode); +static EAS_RESULT SMF_GetDeltaTime (EAS_HW_DATA_HANDLE hwInstData, S_SMF_STREAM *pSMFStream); +static void SMF_UpdateTime (S_SMF_DATA *pSMFData, EAS_U32 ticks); + + +/*---------------------------------------------------------------------------- + * + * SMF_Parser + * + * This structure contains the functional interface for the SMF parser + *---------------------------------------------------------------------------- +*/ +const S_FILE_PARSER_INTERFACE EAS_SMF_Parser = +{ + SMF_CheckFileType, + SMF_Prepare, + SMF_Time, + SMF_Event, + SMF_State, + SMF_Close, + SMF_Reset, + SMF_Pause, + SMF_Resume, + NULL, + SMF_SetData, + SMF_GetData, + NULL +}; + +/*---------------------------------------------------------------------------- + * SMF_CheckFileType() + *---------------------------------------------------------------------------- + * Purpose: + * Check the file type to see if we can parse it + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset) +{ + S_SMF_DATA* pSMFData; + EAS_RESULT result; + + /* seek to starting offset - usually 0 */ + *ppHandle = NULL; + if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset)) != EAS_SUCCESS) + return result; + + /* search through file for header - slow method */ + if (pEASData->searchHeaderFlag) + { + result = EAS_SearchFile(pEASData, fileHandle, smfHeader, sizeof(smfHeader), &offset); + if (result != EAS_SUCCESS) + return (result == EAS_EOF) ? EAS_SUCCESS : result; + } + + /* read the first 4 bytes of the file - quick method */ + else { + EAS_U8 header[4]; + EAS_I32 count; + if ((result = EAS_HWReadFile(pEASData->hwInstData, fileHandle, header, sizeof(header), &count)) != EAS_SUCCESS) + return result; + + /* check for 'MTrk' - return if no match */ + if ((header[0] != 'M') || (header[1] != 'T') || (header[2] != 'h') || (header[3] != 'd')) + return EAS_SUCCESS; + } + + /* check for static memory allocation */ + if (pEASData->staticMemoryModel) + pSMFData = EAS_CMEnumData(EAS_CM_SMF_DATA); + else + { + pSMFData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_SMF_DATA)); + EAS_HWMemSet((void *)pSMFData,0, sizeof(S_SMF_DATA)); + } + if (!pSMFData) + return EAS_ERROR_MALLOC_FAILED; + + /* initialize some critical data */ + pSMFData->fileHandle = fileHandle; + pSMFData->fileOffset = offset; + pSMFData->pSynth = NULL; + pSMFData->time = 0; + pSMFData->state = EAS_STATE_OPEN; + *ppHandle = pSMFData; + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Prepare() + *---------------------------------------------------------------------------- + * Purpose: + * Prepare to parse the file. Allocates instance data (or uses static allocation for + * static memory model). + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) +{ + S_SMF_DATA* pSMFData; + EAS_RESULT result; + + /* check for valid state */ + pSMFData = (S_SMF_DATA *) pInstData; + if (pSMFData->state != EAS_STATE_OPEN) + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + + /* instantiate a synthesizer */ + if ((result = VMInitMIDI(pEASData, &pSMFData->pSynth)) != EAS_SUCCESS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ } + return result; + } + + /* parse the file header and setup the individual stream parsers */ + if ((result = SMF_ParseHeader(pEASData->hwInstData, pSMFData)) != EAS_SUCCESS) + return result; + + /* ready to play */ + pSMFData->state = EAS_STATE_READY; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Time() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the time of the next event in msecs + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * pTime - pointer to variable to hold time of next event (in msecs) + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_RESULT SMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime) +{ + S_SMF_DATA *pSMFData; + + pSMFData = (S_SMF_DATA*) pInstData; + + /* sanity check */ +#ifdef _CHECKED_BUILD + if (pSMFData->state == EAS_STATE_STOPPED) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Can't ask for time on a stopped stream\n"); */ } + } + + if (pSMFData->nextStream == NULL) + { + { /* dpp: EAS_ReportEx( _EAS_SEVERITY_ERROR, "no is NULL\n"); */ } + } +#endif + +#if 0 + /* return time in milliseconds */ + /* if chase mode, lie about time */ + if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) + *pTime = 0; + + else +#endif + + /*lint -e{704} use shift instead of division */ + *pTime = pSMFData->time >> 8; + + *pTime = pSMFData->time >> 8; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Event() + *---------------------------------------------------------------------------- + * Purpose: + * Parse the next event in the file + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode) +{ + S_SMF_DATA* pSMFData; + EAS_RESULT result; + EAS_I32 i; + EAS_U32 ticks; + EAS_U32 temp; + + /* establish pointer to instance data */ + pSMFData = (S_SMF_DATA*) pInstData; + if (pSMFData->state >= EAS_STATE_OPEN) + return EAS_SUCCESS; + + /* get current ticks */ + ticks = pSMFData->nextStream->ticks; + + /* assume that an error occurred */ + pSMFData->state = EAS_STATE_ERROR; + +#ifdef JET_INTERFACE + /* if JET has track muted, set parser mode to mute */ + if (pSMFData->nextStream->midiStream.jetData & MIDI_FLAGS_JET_MUTE) + parserMode = eParserModeMute; +#endif + + /* parse the next event from all the streams */ + if ((result = SMF_ParseEvent(pEASData, pSMFData, pSMFData->nextStream, parserMode)) != EAS_SUCCESS) + { + /* check for unexpected end-of-file */ + if (result != EAS_EOF) + return result; + + /* indicate end of track for this stream */ + pSMFData->nextStream->ticks = SMF_END_OF_TRACK; + } + + /* get next delta time, unless already at end of track */ + else if (pSMFData->nextStream->ticks != SMF_END_OF_TRACK) + { + if ((result = SMF_GetDeltaTime(pEASData->hwInstData, pSMFData->nextStream)) != EAS_SUCCESS) + { + /* check for unexpected end-of-file */ + if (result != EAS_EOF) + return result; + + /* indicate end of track for this stream */ + pSMFData->nextStream->ticks = SMF_END_OF_TRACK; + } + + /* if zero delta to next event, stay with this stream */ + else if (pSMFData->nextStream->ticks == ticks) + { + pSMFData->state = EAS_STATE_PLAY; + return EAS_SUCCESS; + } + } + + /* find next event in all streams */ + temp = 0x7ffffff; + pSMFData->nextStream = NULL; + for (i = 0; i < pSMFData->numStreams; i++) + { + if (pSMFData->streams[i].ticks < temp) + { + temp = pSMFData->streams[i].ticks; + pSMFData->nextStream = &pSMFData->streams[i]; + } + } + + /* are there any more events to parse? */ + if (pSMFData->nextStream) + { + pSMFData->state = EAS_STATE_PLAY; + + /* update the time of the next event */ + SMF_UpdateTime(pSMFData, pSMFData->nextStream->ticks - ticks); + } + else + { + pSMFData->state = EAS_STATE_STOPPING; + VMReleaseAllVoices(pEASData->pVoiceMgr, pSMFData->pSynth); + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_State() + *---------------------------------------------------------------------------- + * Purpose: + * Returns the current state of the stream + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * pState - pointer to variable to store state + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_RESULT SMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState) +{ + S_SMF_DATA* pSMFData; + + /* establish pointer to instance data */ + pSMFData = (S_SMF_DATA*) pInstData; + + /* if stopping, check to see if synth voices are active */ + if (pSMFData->state == EAS_STATE_STOPPING) + { + if (VMActiveVoices(pSMFData->pSynth) == 0) + pSMFData->state = EAS_STATE_STOPPED; + } + + if (pSMFData->state == EAS_STATE_PAUSING) + { + if (VMActiveVoices(pSMFData->pSynth) == 0) + pSMFData->state = EAS_STATE_PAUSED; + } + + /* return current state */ + *pState = pSMFData->state; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Close() + *---------------------------------------------------------------------------- + * Purpose: + * Close the file and clean up + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) +{ + S_SMF_DATA* pSMFData; + EAS_I32 i; + EAS_RESULT result; + + pSMFData = (S_SMF_DATA*) pInstData; + + /* close all the streams */ + for (i = 0; i < pSMFData->numStreams; i++) + { + if (pSMFData->streams[i].fileHandle != NULL) + { + if ((result = EAS_HWCloseFile(pEASData->hwInstData, pSMFData->streams[i].fileHandle)) != EAS_SUCCESS) + return result; + } + } + if (pSMFData->fileHandle != NULL) + if ((result = EAS_HWCloseFile(pEASData->hwInstData, pSMFData->fileHandle)) != EAS_SUCCESS) + return result; + + /* free the synth */ + if (pSMFData->pSynth != NULL) + VMMIDIShutdown(pEASData, pSMFData->pSynth); + + /* if using dynamic memory, free it */ + if (!pEASData->staticMemoryModel) + { + if (pSMFData->streams) + EAS_HWFree(pEASData->hwInstData, pSMFData->streams); + + /* free the instance data */ + EAS_HWFree(pEASData->hwInstData, pSMFData); + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Reset() + *---------------------------------------------------------------------------- + * Purpose: + * Reset the sequencer. Used for locating backwards in the file. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) +{ + S_SMF_DATA* pSMFData; + EAS_I32 i; + EAS_RESULT result; + EAS_U32 ticks; + + pSMFData = (S_SMF_DATA*) pInstData; + + /* reset time to zero */ + pSMFData->time = 0; + + /* reset the synth */ + VMReset(pEASData->pVoiceMgr, pSMFData->pSynth, EAS_TRUE); + + /* find the start of each track */ + ticks = 0x7fffffffL; + pSMFData->nextStream = NULL; + for (i = 0; i < pSMFData->numStreams; i++) + { + + /* reset file position to first byte of data in track */ + if ((result = EAS_HWFileSeek(pEASData->hwInstData, pSMFData->streams[i].fileHandle, pSMFData->streams[i].startFilePos)) != EAS_SUCCESS) + return result; + + /* initalize some data */ + pSMFData->streams[i].ticks = 0; + + /* initalize the MIDI parser data */ + EAS_InitMIDIStream(&pSMFData->streams[i].midiStream); + + /* parse the first delta time in each stream */ + if ((result = SMF_GetDeltaTime(pEASData->hwInstData,&pSMFData->streams[i])) != EAS_SUCCESS) + return result; + if (pSMFData->streams[i].ticks < ticks) + { + ticks = pSMFData->streams[i].ticks; + pSMFData->nextStream = &pSMFData->streams[i]; + } + } + + + pSMFData->state = EAS_STATE_READY; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Pause() + *---------------------------------------------------------------------------- + * Purpose: + * Pauses the sequencer. Mutes all voices and sets state to pause. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +EAS_RESULT SMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) +{ + S_SMF_DATA *pSMFData; + + /* can't pause a stopped stream */ + pSMFData = (S_SMF_DATA*) pInstData; + if (pSMFData->state == EAS_STATE_STOPPED) + return EAS_ERROR_ALREADY_STOPPED; + + /* mute the synthesizer */ + VMMuteAllVoices(pEASData->pVoiceMgr, pSMFData->pSynth); + pSMFData->state = EAS_STATE_PAUSING; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_Resume() + *---------------------------------------------------------------------------- + * Purpose: + * Resume playing after a pause, sets state back to playing. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_RESULT SMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData) +{ + S_SMF_DATA *pSMFData; + + /* can't resume a stopped stream */ + pSMFData = (S_SMF_DATA*) pInstData; + if (pSMFData->state == EAS_STATE_STOPPED) + return EAS_ERROR_ALREADY_STOPPED; + + /* nothing to do but resume playback */ + pSMFData->state = EAS_STATE_PLAY; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_SetData() + *---------------------------------------------------------------------------- + * Purpose: + * Sets parser parameters + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_RESULT SMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value) +{ + S_SMF_DATA *pSMFData; + + pSMFData = (S_SMF_DATA*) pInstData; + switch (param) + { + + /* set metadata callback */ + case PARSER_DATA_METADATA_CB: + EAS_HWMemCpy(&pSMFData->metadata, (void*) value, sizeof(S_METADATA_CB)); + break; + +#ifdef JET_INTERFACE + /* set jet segment and track ID of all tracks for callback function */ + case PARSER_DATA_JET_CB: + { + EAS_U32 i; + EAS_U32 bit = (EAS_U32) value; + bit = (bit << JET_EVENT_SEG_SHIFT) & JET_EVENT_SEG_MASK; + for (i = 0; i < pSMFData->numStreams; i++) + pSMFData->streams[i].midiStream.jetData = + (pSMFData->streams[i].midiStream.jetData & + ~(JET_EVENT_TRACK_MASK | JET_EVENT_SEG_MASK)) | + i << JET_EVENT_TRACK_SHIFT | bit | MIDI_FLAGS_JET_CB; + pSMFData->flags |= SMF_FLAGS_JET_STREAM; + } + break; + + /* set state of all mute flags at once */ + case PARSER_DATA_MUTE_FLAGS: + { + EAS_INT i; + EAS_U32 bit = (EAS_U32) value; + for (i = 0; i < pSMFData->numStreams; i++) + { + if (bit & 1) + pSMFData->streams[i].midiStream.jetData |= MIDI_FLAGS_JET_MUTE; + else + pSMFData->streams[i].midiStream.jetData &= ~MIDI_FLAGS_JET_MUTE; + bit >>= 1; + } + } + break; + + /* set track mute */ + case PARSER_DATA_SET_MUTE: + if (value < pSMFData->numStreams) + pSMFData->streams[value].midiStream.jetData |= MIDI_FLAGS_JET_MUTE; + else + return EAS_ERROR_PARAMETER_RANGE; + break; + + /* clear track mute */ + case PARSER_DATA_CLEAR_MUTE: + if (value < pSMFData->numStreams) + pSMFData->streams[value].midiStream.jetData &= ~MIDI_FLAGS_JET_MUTE; + else + return EAS_ERROR_PARAMETER_RANGE; + break; +#endif + + default: + return EAS_ERROR_INVALID_PARAMETER; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_GetData() + *---------------------------------------------------------------------------- + * Purpose: + * Retrieves parser parameters + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * handle - pointer to file handle + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pEASData) reserved for future use */ +EAS_RESULT SMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue) +{ + S_SMF_DATA *pSMFData; + + pSMFData = (S_SMF_DATA*) pInstData; + switch (param) + { + /* return file type */ + case PARSER_DATA_FILE_TYPE: + if (pSMFData->numStreams == 1) + *pValue = EAS_FILE_SMF0; + else + *pValue = EAS_FILE_SMF1; + break; + +/* now handled in eas_public.c */ +#if 0 + case PARSER_DATA_POLYPHONY: + if (pSMFData->pSynth) + VMGetPolyphony(pEASData->pVoiceMgr, pSMFData->pSynth, pValue); + else + return EAS_ERROR_NOT_VALID_IN_THIS_STATE; + break; + + case PARSER_DATA_PRIORITY: + if (pSMFData->pSynth) + VMGetPriority(pEASData->pVoiceMgr, pSMFData->pSynth, pValue); + break; + + /* set transposition */ + case PARSER_DATA_TRANSPOSITION: + *pValue = pSMFData->transposition; + break; +#endif + + case PARSER_DATA_SYNTH_HANDLE: + *pValue = (EAS_I32) pSMFData->pSynth; + break; + + default: + return EAS_ERROR_INVALID_PARAMETER; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_GetVarLenData() + *---------------------------------------------------------------------------- + * Purpose: + * Reads a varible length quantity from an SMF file + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT SMF_GetVarLenData (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_U32 *pData) +{ + EAS_RESULT result; + EAS_U32 data; + EAS_U8 c; + + /* read until bit 7 is zero */ + data = 0; + do + { + if ((result = EAS_HWGetByte(hwInstData, fileHandle,&c)) != EAS_SUCCESS) + return result; + data = (data << 7) | (c & 0x7f); + } while (c & 0x80); + *pData = data; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_GetDeltaTime() + *---------------------------------------------------------------------------- + * Purpose: + * Reads a varible length quantity from an SMF file + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT SMF_GetDeltaTime (EAS_HW_DATA_HANDLE hwInstData, S_SMF_STREAM *pSMFStream) +{ + EAS_RESULT result; + EAS_U32 ticks; + + if ((result = SMF_GetVarLenData(hwInstData, pSMFStream->fileHandle, &ticks)) != EAS_SUCCESS) + return result; + + pSMFStream->ticks += ticks; + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_ParseMetaEvent() + *---------------------------------------------------------------------------- + * Purpose: + * Reads a varible length quantity from an SMF file + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT SMF_ParseMetaEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream) +{ + EAS_RESULT result; + EAS_U32 len; + EAS_I32 pos; + EAS_U32 temp; + EAS_U8 c; + + /* get the meta-event type */ + if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) + return result; + + /* get the length */ + if ((result = SMF_GetVarLenData(pEASData->hwInstData, pSMFStream->fileHandle, &len)) != EAS_SUCCESS) + return result; + + /* get the current file position so we can skip the event */ + if ((result = EAS_HWFilePos(pEASData->hwInstData, pSMFStream->fileHandle, &pos)) != EAS_SUCCESS) + return result; + pos += (EAS_I32) len; + + /* end of track? */ + if (c == SMF_META_END_OF_TRACK) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Meta-event: end of track\n", c, len); */ } + pSMFStream->ticks = SMF_END_OF_TRACK; + } + + /* tempo event? */ + else if (c == SMF_META_TEMPO) + { + /* read the 3-byte timebase value */ + temp = 0; + while (len--) + { + if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) + return result; + temp = (temp << 8) | c; + } + + pSMFData->tickConv = (EAS_U16) (((temp * 1024) / pSMFData->ppqn + 500) / 1000); + pSMFData->flags |= SMF_FLAGS_HAS_TEMPO; + } + + /* check for time signature - see iMelody spec V1.4 section 4.1.2.2.3.6 */ + else if (c == SMF_META_TIME_SIGNATURE) + { + pSMFData->flags |= SMF_FLAGS_HAS_TIME_SIG; + } + + /* if the host has registered a metadata callback return the metadata */ + else if (pSMFData->metadata.callback) + { + EAS_I32 readLen; + E_EAS_METADATA_TYPE metaType; + + metaType = EAS_METADATA_UNKNOWN; + + /* only process title on the first track */ + if (c == SMF_META_SEQTRK_NAME) + metaType = EAS_METADATA_TITLE; + else if (c == SMF_META_TEXT) + metaType = EAS_METADATA_TEXT; + else if (c == SMF_META_COPYRIGHT) + metaType = EAS_METADATA_COPYRIGHT; + else if (c == SMF_META_LYRIC) + metaType = EAS_METADATA_LYRIC; + + if (metaType != EAS_METADATA_UNKNOWN) + { + readLen = pSMFData->metadata.bufferSize - 1; + if ((EAS_I32) len < readLen) + readLen = (EAS_I32) len; + if ((result = EAS_HWReadFile(pEASData->hwInstData, pSMFStream->fileHandle, pSMFData->metadata.buffer, readLen, &readLen)) != EAS_SUCCESS) + return result; + pSMFData->metadata.buffer[readLen] = 0; + pSMFData->metadata.callback(metaType, pSMFData->metadata.buffer, pSMFData->metadata.pUserData); + } + } + + /* position file to next event - in case we ignored all or part of the meta-event */ + if ((result = EAS_HWFileSeek(pEASData->hwInstData, pSMFStream->fileHandle, pos)) != EAS_SUCCESS) + return result; + + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Meta-event: type=%02x, len=%d\n", c, len); */ } + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_ParseSysEx() + *---------------------------------------------------------------------------- + * Purpose: + * Reads a varible length quantity from an SMF file + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT SMF_ParseSysEx (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_U8 f0, EAS_INT parserMode) +{ + EAS_RESULT result; + EAS_U32 len; + EAS_U8 c; + + /* get the length */ + if ((result = SMF_GetVarLenData(pEASData->hwInstData, pSMFStream->fileHandle, &len)) != EAS_SUCCESS) + return result; + + /* start of SysEx message? */ + if (f0 == 0xf0) + { + if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, f0, parserMode)) != EAS_SUCCESS) + return result; + } + + /* feed the SysEx to the stream parser */ + while (len--) + { + if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) + return result; + if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) + return result; + + /* check for GM system ON */ + if (pSMFStream->midiStream.flags & MIDI_FLAG_GM_ON) + pSMFData->flags |= SMF_FLAGS_HAS_GM_ON; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_ParseEvent() + *---------------------------------------------------------------------------- + * Purpose: + * Reads a varible length quantity from an SMF file + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT SMF_ParseEvent (S_EAS_DATA *pEASData, S_SMF_DATA *pSMFData, S_SMF_STREAM *pSMFStream, EAS_INT parserMode) +{ + EAS_RESULT result; + EAS_U8 c; + + /* get the event type */ + if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) + return result; + + /* parse meta-event */ + if (c == 0xff) + { + if ((result = SMF_ParseMetaEvent(pEASData, pSMFData, pSMFStream)) != EAS_SUCCESS) + return result; + } + + /* parse SysEx */ + else if ((c == 0xf0) || (c == 0xf7)) + { + if ((result = SMF_ParseSysEx(pEASData, pSMFData, pSMFStream, c, parserMode)) != EAS_SUCCESS) + return result; + } + + /* parse MIDI message */ + else + { + if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) + return result; + + /* keep streaming data to the MIDI parser until the message is complete */ + while (pSMFStream->midiStream.pending) + { + if ((result = EAS_HWGetByte(pEASData->hwInstData, pSMFStream->fileHandle, &c)) != EAS_SUCCESS) + return result; + if ((result = EAS_ParseMIDIStream(pEASData, pSMFData->pSynth, &pSMFStream->midiStream, c, parserMode)) != EAS_SUCCESS) + return result; + } + + } + + /* chase mode logic */ + if (pSMFData->time == 0) + { + if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) + { + if (pSMFStream->midiStream.flags & MIDI_FLAG_FIRST_NOTE) + pSMFData->flags &= ~SMF_FLAGS_CHASE_MODE; + } + else if ((pSMFData->flags & SMF_FLAGS_SETUP_BAR) == SMF_FLAGS_SETUP_BAR) + pSMFData->flags = (pSMFData->flags & ~SMF_FLAGS_SETUP_BAR) | SMF_FLAGS_CHASE_MODE; + } + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * SMF_ParseHeader() + *---------------------------------------------------------------------------- + * Purpose: + * Parses the header of an SMF file, allocates memory the stream parsers and initializes the + * stream parsers. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * pSMFData - pointer to parser instance data + * fileHandle - file handle + * fileOffset - offset in the file where the header data starts, usually 0 + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +/*lint -e{801} we know that 'goto' is deprecated - but it's cleaner in this case */ +EAS_RESULT SMF_ParseHeader (EAS_HW_DATA_HANDLE hwInstData, S_SMF_DATA *pSMFData) +{ + EAS_RESULT result; + EAS_I32 i; + EAS_U16 division; + EAS_U32 chunkSize; + EAS_U32 chunkStart; + EAS_U32 temp; + EAS_U32 ticks; + + /* rewind the file and find the end of the header chunk */ + if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, pSMFData->fileOffset + SMF_OFS_HEADER_SIZE)) != EAS_SUCCESS) + goto ReadError; + if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &chunkSize, EAS_TRUE)) != EAS_SUCCESS) + goto ReadError; + + /* determine the number of tracks */ + if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, pSMFData->fileOffset + SMF_OFS_NUM_TRACKS)) != EAS_SUCCESS) + goto ReadError; + if ((result = EAS_HWGetWord(hwInstData, pSMFData->fileHandle, &pSMFData->numStreams, EAS_TRUE)) != EAS_SUCCESS) + goto ReadError; + + /* limit the number of tracks */ + if (pSMFData->numStreams > MAX_SMF_STREAMS) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "SMF file contains %u tracks, playing %d tracks\n", pSMFData->numStreams, MAX_SMF_STREAMS); */ } + pSMFData->numStreams = MAX_SMF_STREAMS; + } else if (pSMFData->numStreams == 0) + { + /* avoid 0 sized allocation */ + return EAS_ERROR_PARAMETER_RANGE; + } + + /* get the time division */ + if ((result = EAS_HWGetWord(hwInstData, pSMFData->fileHandle, &division, EAS_TRUE)) != EAS_SUCCESS) + goto ReadError; + + /* setup default timebase for 120 bpm */ + pSMFData->ppqn = 192; + if (!division || division & 0x8000) + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"No support for SMPTE code timebase\n"); */ } + else + pSMFData->ppqn = (division & 0x7fff); + pSMFData->tickConv = (EAS_U16) (((SMF_DEFAULT_TIMEBASE * 1024) / pSMFData->ppqn + 500) / 1000); + + /* dynamic memory allocation, allocate memory for streams */ + if (pSMFData->streams == NULL) + { + pSMFData->streams = EAS_HWMalloc(hwInstData,sizeof(S_SMF_STREAM) * pSMFData->numStreams); + if (pSMFData->streams == NULL) + return EAS_ERROR_MALLOC_FAILED; + + /* zero the memory to insure complete initialization */ + EAS_HWMemSet((void *)(pSMFData->streams), 0, sizeof(S_SMF_STREAM) * pSMFData->numStreams); + } + + /* find the start of each track */ + chunkStart = (EAS_U32) pSMFData->fileOffset; + ticks = 0x7fffffffL; + pSMFData->nextStream = NULL; + for (i = 0; i < pSMFData->numStreams; i++) + { + + for (;;) + { + + /* calculate start of next chunk - checking for errors */ + temp = chunkStart + SMF_CHUNK_INFO_SIZE + chunkSize; + if (temp <= chunkStart) + { + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"Error in chunk size at offset %d\n", chunkStart); */ } + return EAS_ERROR_FILE_FORMAT; + } + chunkStart = temp; + + /* seek to the start of the next chunk */ + if ((result = EAS_HWFileSeek(hwInstData, pSMFData->fileHandle, (EAS_I32) chunkStart)) != EAS_SUCCESS) + goto ReadError; + + /* read the chunk identifier */ + if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS) + goto ReadError; + + /* read the chunk size */ + if ((result = EAS_HWGetDWord(hwInstData, pSMFData->fileHandle, &chunkSize, EAS_TRUE)) != EAS_SUCCESS) + goto ReadError; + + /* make sure this is an 'MTrk' chunk */ + if (temp == SMF_CHUNK_TYPE_TRACK) + break; + + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING,"Unexpected chunk type: 0x%08x\n", temp); */ } + } + + /* initalize some data */ + pSMFData->streams[i].ticks = 0; + pSMFData->streams[i].fileHandle = pSMFData->fileHandle; + + /* NULL the file handle so we don't try to close it twice */ + pSMFData->fileHandle = NULL; + + /* save this file position as the start of the track */ + pSMFData->streams[i].startFilePos = (EAS_I32) chunkStart + SMF_CHUNK_INFO_SIZE; + + /* initalize the MIDI parser data */ + EAS_InitMIDIStream(&pSMFData->streams[i].midiStream); + + /* parse the first delta time in each stream */ + if ((result = SMF_GetDeltaTime(hwInstData, &pSMFData->streams[i])) != EAS_SUCCESS) + goto ReadError; + + if (pSMFData->streams[i].ticks < ticks) + { + ticks = pSMFData->streams[i].ticks; + pSMFData->nextStream = &pSMFData->streams[i]; + } + + /* more tracks to do, create a duplicate file handle */ + if (i < (pSMFData->numStreams - 1)) + { + if ((result = EAS_HWDupHandle(hwInstData, pSMFData->streams[i].fileHandle, &pSMFData->fileHandle)) != EAS_SUCCESS) + goto ReadError; + } + } + + /* update the time of the next event */ + if (pSMFData->nextStream) + SMF_UpdateTime(pSMFData, pSMFData->nextStream->ticks); + + return EAS_SUCCESS; + + /* ugly goto: but simpler than structured */ + ReadError: + if (result == EAS_EOF) + return EAS_ERROR_FILE_FORMAT; + return result; +} + +/*---------------------------------------------------------------------------- + * SMF_UpdateTime() + *---------------------------------------------------------------------------- + * Purpose: + * Update the millisecond time base by converting the ticks into millieconds + * + * Inputs: + * + * + * Outputs: + * + * + * Side Effects: + * + *---------------------------------------------------------------------------- +*/ +static void SMF_UpdateTime (S_SMF_DATA *pSMFData, EAS_U32 ticks) +{ + EAS_U32 temp1, temp2; + + if (pSMFData->flags & SMF_FLAGS_CHASE_MODE) + return; + + temp1 = (ticks >> 10) * pSMFData->tickConv; + temp2 = (ticks & 0x3ff) * pSMFData->tickConv; + pSMFData->time += (EAS_I32)((temp1 << 8) + (temp2 >> 2)); +} + diff --git a/arm-wt-22k/lib_src/eas_smf.h b/arm-wt-22k/lib_src/eas_smf.h new file mode 100644 index 0000000..9f66ab9 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_smf.h @@ -0,0 +1,49 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smf.h
+ *
+ * Contents and purpose:
+ * SMF Type 0 and 1 File Parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SMF_H
+#define _EAS_SMF_H
+
+/* prototypes for private interface to SMF parser */
+EAS_RESULT SMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+EAS_RESULT SMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+EAS_RESULT SMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+EAS_RESULT SMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+EAS_RESULT SMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+EAS_RESULT SMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+EAS_RESULT SMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+EAS_RESULT SMF_ParseHeader (EAS_HW_DATA_HANDLE hwInstData, S_SMF_DATA *pSMFData);
+
+#endif /* end _EAS_SMF_H */
+
+
diff --git a/arm-wt-22k/lib_src/eas_smfdata.c b/arm-wt-22k/lib_src/eas_smfdata.c new file mode 100644 index 0000000..5c27551 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_smfdata.c @@ -0,0 +1,66 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smfdata.c
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 778 $
+ * $Date: 2007-07-23 16:45:17 -0700 (Mon, 23 Jul 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_miditypes.h"
+#include "eas_smfdata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * S_SMF_STREAM
+ *
+ * Static memory allocation for SMF parser
+ *----------------------------------------------------------------------------
+*/
+static S_SMF_STREAM eas_SMFStreams[MAX_SMF_STREAMS];
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_SMFData
+ *
+ * Static memory allocation for SMF parser
+ *----------------------------------------------------------------------------
+*/
+S_SMF_DATA eas_SMFData =
+{
+ eas_SMFStreams, /* pointer to individual streams in file */
+ 0, /* pointer to next stream with event */
+ 0, /* pointer to synth */
+ 0, /* file handle */
+ { 0, 0, 0, 0}, /* metadata callback */
+ 0, /* file offset */
+ 0, /* current time in milliseconds/256 */
+ 0, /* actual number of streams */
+ 0, /* current MIDI tick to msec conversion */
+ 0, /* ticks per quarter note */
+ 0, /* current state EAS_STATE_XXXX */
+ 0 /* flags */
+};
+
diff --git a/arm-wt-22k/lib_src/eas_smfdata.h b/arm-wt-22k/lib_src/eas_smfdata.h new file mode 100644 index 0000000..cf59cdc --- /dev/null +++ b/arm-wt-22k/lib_src/eas_smfdata.h @@ -0,0 +1,66 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_smfdata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data definitions for the SMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 686 $
+ * $Date: 2007-05-03 14:10:54 -0700 (Thu, 03 May 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SMF_DATA_H
+#define _EAS_SMF_DATA_H
+
+#ifndef MAX_SMF_STREAMS
+#define MAX_SMF_STREAMS 17
+#endif
+
+/* offsets in to the SMF file */
+#define SMF_OFS_HEADER_SIZE 4
+#define SMF_OFS_FILE_TYPE 8
+#define SMF_OFS_NUM_TRACKS 10
+
+/* size of chunk info (chunk ID + chunk size) */
+#define SMF_CHUNK_INFO_SIZE 8
+
+/* 'MTrk' track chunk ID */
+#define SMF_CHUNK_TYPE_TRACK 0x4d54726bL
+
+/* some useful meta-events */
+#define SMF_META_TEXT 0x01
+#define SMF_META_COPYRIGHT 0x02
+#define SMF_META_SEQTRK_NAME 0x03
+#define SMF_META_LYRIC 0x05
+#define SMF_META_END_OF_TRACK 0x2f
+#define SMF_META_TEMPO 0x51
+#define SMF_META_TIME_SIGNATURE 0x58
+
+/* default timebase (120BPM) */
+#define SMF_DEFAULT_TIMEBASE 500000L
+
+/* value for pSMFStream->ticks to signify end of track */
+#define SMF_END_OF_TRACK 0xffffffff
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_sndlib.h b/arm-wt-22k/lib_src/eas_sndlib.h new file mode 100644 index 0000000..e05bee0 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_sndlib.h @@ -0,0 +1,406 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_sndlib.h
+ *
+ * Contents and purpose:
+ * Declarations for the sound library
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 550 $
+ * $Date: 2007-02-02 09:37:03 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SNDLIB_H
+#define _EAS_SNDLIB_H
+
+#include "eas_types.h"
+#include "eas_synthcfg.h"
+
+#ifdef _WT_SYNTH
+#include "eas_wtengine.h"
+#endif
+
+/*----------------------------------------------------------------------------
+ * This is bit of a hack to allow us to keep the same structure
+ * declarations for the DLS parser. Normally, the data is located
+ * in read-only memory, but for DLS, we store the data in RW
+ * memory.
+ *----------------------------------------------------------------------------
+*/
+#ifndef SCNST
+#define SCNST const
+#endif
+
+/*----------------------------------------------------------------------------
+ * sample size
+ *----------------------------------------------------------------------------
+*/
+#ifdef _16_BIT_SAMPLES
+typedef EAS_I16 EAS_SAMPLE;
+#else
+typedef EAS_I8 EAS_SAMPLE;
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS Library ID - quick check for valid library and version
+ *----------------------------------------------------------------------------
+*/
+#define _EAS_LIBRARY_VERSION 0x01534145
+
+#define NUM_PROGRAMS_IN_BANK 128
+#define INVALID_REGION_INDEX 0xffff
+
+/* this bit in region index indicates that region is for secondary synth */
+#define FLAG_RGN_IDX_FM_SYNTH 0x8000
+#define FLAG_RGN_IDX_DLS_SYNTH 0x4000
+#define REGION_INDEX_MASK 0x3fff
+
+/*----------------------------------------------------------------------------
+ * Generic region data structure
+ *
+ * This must be the first element in each region structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_region_tag
+{
+ EAS_U16 keyGroupAndFlags;
+ EAS_U8 rangeLow;
+ EAS_U8 rangeHigh;
+} S_REGION;
+
+/*
+ * Bit fields for m_nKeyGroupAndFlags
+ * Bits 0-2 are mode bits in FM synth
+ * Bits 8-11 are the key group
+ */
+#define REGION_FLAG_IS_LOOPED 0x01
+#define REGION_FLAG_USE_WAVE_GENERATOR 0x02
+#define REGION_FLAG_USE_ADPCM 0x04
+#define REGION_FLAG_ONE_SHOT 0x08
+#define REGION_FLAG_SQUARE_WAVE 0x10
+#define REGION_FLAG_OFF_CHIP 0x20
+#define REGION_FLAG_NON_SELF_EXCLUSIVE 0x40
+#define REGION_FLAG_LAST_REGION 0x8000
+
+/*----------------------------------------------------------------------------
+ * Envelope data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_envelope_tag
+{
+ EAS_I16 attackTime;
+ EAS_I16 decayTime;
+ EAS_I16 sustainLevel;
+ EAS_I16 releaseTime;
+} S_ENVELOPE;
+
+/*----------------------------------------------------------------------------
+ * DLS envelope data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_dls_envelope_tag
+{
+ EAS_I16 delayTime;
+ EAS_I16 attackTime;
+ EAS_I16 holdTime;
+ EAS_I16 decayTime;
+ EAS_I16 sustainLevel;
+ EAS_I16 releaseTime;
+ EAS_I16 velToAttack;
+ EAS_I16 keyNumToDecay;
+ EAS_I16 keyNumToHold;
+} S_DLS_ENVELOPE;
+
+/*----------------------------------------------------------------------------
+ * LFO data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_lfo_params_tag
+{
+ EAS_I16 lfoFreq;
+ EAS_I16 lfoDelay;
+} S_LFO_PARAMS;
+
+/*----------------------------------------------------------------------------
+ * Articulation data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_articulation_tag
+{
+ S_ENVELOPE eg1;
+ S_ENVELOPE eg2;
+ EAS_I16 lfoToPitch;
+ EAS_I16 lfoDelay;
+ EAS_I16 lfoFreq;
+ EAS_I16 eg2ToPitch;
+ EAS_I16 eg2ToFc;
+ EAS_I16 filterCutoff;
+ EAS_I8 lfoToGain;
+ EAS_U8 filterQ;
+ EAS_I8 pan;
+} S_ARTICULATION;
+
+/*----------------------------------------------------------------------------
+ * DLS articulation data structure
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct s_dls_articulation_tag
+{
+ S_LFO_PARAMS modLFO;
+ S_LFO_PARAMS vibLFO;
+
+ S_DLS_ENVELOPE eg1;
+ S_DLS_ENVELOPE eg2;
+
+ EAS_I16 eg1ShutdownTime;
+
+ EAS_I16 filterCutoff;
+ EAS_I16 modLFOToFc;
+ EAS_I16 modLFOCC1ToFc;
+ EAS_I16 modLFOChanPressToFc;
+ EAS_I16 eg2ToFc;
+ EAS_I16 velToFc;
+ EAS_I16 keyNumToFc;
+
+ EAS_I16 modLFOToGain;
+ EAS_I16 modLFOCC1ToGain;
+ EAS_I16 modLFOChanPressToGain;
+
+ EAS_I16 tuning;
+ EAS_I16 keyNumToPitch;
+ EAS_I16 vibLFOToPitch;
+ EAS_I16 vibLFOCC1ToPitch;
+ EAS_I16 vibLFOChanPressToPitch;
+ EAS_I16 modLFOToPitch;
+ EAS_I16 modLFOCC1ToPitch;
+ EAS_I16 modLFOChanPressToPitch;
+ EAS_I16 eg2ToPitch;
+
+ /* pad to 4-byte boundary */
+ EAS_U16 pad;
+
+ EAS_I8 pan;
+ EAS_U8 filterQandFlags;
+
+#ifdef _REVERB
+ EAS_I16 reverbSend;
+ EAS_I16 cc91ToReverbSend;
+#endif
+
+#ifdef _CHORUS
+ EAS_I16 chorusSend;
+ EAS_I16 cc93ToChorusSend;
+#endif
+} S_DLS_ARTICULATION;
+
+/* flags in filterQandFlags
+ * NOTE: Q is stored in bottom 5 bits
+ */
+#define FLAG_DLS_VELOCITY_SENSITIVE 0x80
+#define FILTER_Q_MASK 0x1f
+
+/*----------------------------------------------------------------------------
+ * Wavetable region data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_region_tag
+{
+ S_REGION region;
+ EAS_I16 tuning;
+ EAS_I16 gain;
+ EAS_U32 loopStart;
+ EAS_U32 loopEnd;
+ EAS_U16 waveIndex;
+ EAS_U16 artIndex;
+} S_WT_REGION;
+
+/*----------------------------------------------------------------------------
+ * DLS region data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_dls_region_tag
+{
+ S_WT_REGION wtRegion;
+ EAS_U8 velLow;
+ EAS_U8 velHigh;
+} S_DLS_REGION;
+
+/*----------------------------------------------------------------------------
+ * FM synthesizer data structures
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_fm_oper_tag
+{
+ EAS_I16 tuning;
+ EAS_U8 attackDecay;
+ EAS_U8 velocityRelease;
+ EAS_U8 egKeyScale;
+ EAS_U8 sustain;
+ EAS_U8 gain;
+ EAS_U8 flags;
+} S_FM_OPER;
+
+/* defines for S_FM_OPER.m_nFlags */
+#define FM_OPER_FLAG_MONOTONE 0x01
+#define FM_OPER_FLAG_NO_VIBRATO 0x02
+#define FM_OPER_FLAG_NOISE 0x04
+#define FM_OPER_FLAG_LINEAR_VELOCITY 0x08
+
+/* NOTE: The first two structure elements are common with S_WT_REGION
+ * and we will rely on that in the voice management code and must
+ * remain there unless the voice management code is revisited.
+ */
+typedef struct s_fm_region_tag
+{
+ S_REGION region;
+ EAS_U8 vibTrem;
+ EAS_U8 lfoFreqDelay;
+ EAS_U8 feedback;
+ EAS_I8 pan;
+ S_FM_OPER oper[4];
+} S_FM_REGION;
+
+/*----------------------------------------------------------------------------
+ * Common data structures
+ *----------------------------------------------------------------------------
+*/
+
+/*----------------------------------------------------------------------------
+ * Program data structure
+ * Used for individual programs not stored as a complete bank.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_program_tag
+{
+ EAS_U32 locale;
+ EAS_U16 regionIndex;
+} S_PROGRAM;
+
+/*----------------------------------------------------------------------------
+ * Bank data structure
+ *
+ * A bank always consists of 128 programs. If a bank is less than 128
+ * programs, it should be stored as a spare matrix in the pPrograms
+ * array.
+ *
+ * bankNum: MSB/LSB of MIDI bank select controller
+ * regionIndex: Index of first region in program
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_bank_tag
+{
+ EAS_U16 locale;
+ EAS_U16 regionIndex[NUM_PROGRAMS_IN_BANK];
+} S_BANK;
+
+
+/* defines for libFormat field
+ * bits 0-17 are the sample rate
+ * bit 18 is true if wavetable is present
+ * bit 19 is true if FM is present
+ * bit 20 is true if filter is enabled
+ * bit 21 is sample depth (0 = 8-bits, 1 = 16-bits)
+ * bits 22-31 are reserved
+ */
+#define LIBFORMAT_SAMPLE_RATE_MASK 0x0003ffff
+#define LIB_FORMAT_TYPE_MASK 0x000c0000
+#define LIB_FORMAT_WAVETABLE 0x00000000
+#define LIB_FORMAT_FM 0x00040000
+#define LIB_FORMAT_HYBRID 0x00080000
+#define LIB_FORMAT_FILTER_ENABLED 0x00100000
+#define LIB_FORMAT_16_BIT_SAMPLES 0x00200000
+
+#ifdef DLS_SYNTHESIZER
+/*----------------------------------------------------------------------------
+ * DLS data structure
+ *
+ * pDLSPrograms pointer to array of DLS programs
+ * pDLSRegions pointer to array of DLS regions
+ * pDLSArticulations pointer to array of DLS articulations
+ * pSampleLen pointer to array of sample lengths
+ * ppSamples pointer to array of sample pointers
+ * numDLSPrograms number of DLS programs
+ * numDLSRegions number of DLS regions
+ * numDLSArticulations number of DLS articulations
+ * numDLSSamples number of DLS samples
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_eas_dls_tag
+{
+ S_PROGRAM *pDLSPrograms;
+ S_DLS_REGION *pDLSRegions;
+ S_DLS_ARTICULATION *pDLSArticulations;
+ EAS_U32 *pDLSSampleLen;
+ EAS_U32 *pDLSSampleOffsets;
+ EAS_SAMPLE *pDLSSamples;
+ EAS_U16 numDLSPrograms;
+ EAS_U16 numDLSRegions;
+ EAS_U16 numDLSArticulations;
+ EAS_U16 numDLSSamples;
+ EAS_U8 refCount;
+} S_DLS;
+#endif
+
+/*----------------------------------------------------------------------------
+ * Sound library data structure
+ *
+ * pBanks pointer to array of banks
+ * pPrograms pointer to array of programs
+ * pWTRegions pointer to array of wavetable regions
+ * pFMRegions pointer to array of FM regions
+ * pArticulations pointer to array of articulations
+ * pSampleLen pointer to array of sample lengths
+ * ppSamples pointer to array of sample pointers
+ * numBanks number of banks
+ * numPrograms number of individual program
+ * numRegions number of regions
+ * numArticulations number of articulations
+ * numSamples number of samples
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_eas_sndlib_tag
+{
+ SCNST EAS_U32 identifier;
+ SCNST EAS_U32 libAttr;
+
+ SCNST S_BANK *pBanks;
+ SCNST S_PROGRAM *pPrograms;
+
+ SCNST S_WT_REGION *pWTRegions;
+ SCNST S_ARTICULATION *pArticulations;
+ SCNST EAS_U32 *pSampleLen;
+ SCNST EAS_U32 *pSampleOffsets;
+ SCNST EAS_SAMPLE *pSamples;
+
+ SCNST S_FM_REGION *pFMRegions;
+
+ SCNST EAS_U16 numBanks;
+ SCNST EAS_U16 numPrograms;
+
+ SCNST EAS_U16 numWTRegions;
+ SCNST EAS_U16 numArticulations;
+ SCNST EAS_U16 numSamples;
+
+ SCNST EAS_U16 numFMRegions;
+} S_EAS;
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_synth.h b/arm-wt-22k/lib_src/eas_synth.h new file mode 100644 index 0000000..b242b03 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_synth.h @@ -0,0 +1,395 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synth.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for synth.
+ *
+ * Copyright Sonic Network Inc. 2004, 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 718 $
+ * $Date: 2007-06-08 16:43:16 -0700 (Fri, 08 Jun 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTH_H
+#define _EAS_SYNTH_H
+
+#include "eas_types.h"
+#include "eas_sndlib.h"
+
+#ifdef _WT_SYNTH
+#include "eas_wtsynth.h"
+#endif
+
+#ifdef _FM_SYNTH
+#include "eas_fmsynth.h"
+#endif
+
+#ifndef NUM_OUTPUT_CHANNELS
+#define NUM_OUTPUT_CHANNELS 2
+#endif
+
+#ifndef MAX_SYNTH_VOICES
+#define MAX_SYNTH_VOICES 64
+#endif
+
+#ifndef MAX_VIRTUAL_SYNTHESIZERS
+#define MAX_VIRTUAL_SYNTHESIZERS 4
+#endif
+
+/* defines */
+#ifndef NUM_PRIMARY_VOICES
+#define NUM_PRIMARY_VOICES MAX_SYNTH_VOICES
+#elif !defined(NUM_SECONDARY_VOICES)
+#define NUM_SECONDARY_VOICES (MAX_SYNTH_VOICES - NUM_PRIMARY_VOICES)
+#endif
+
+#if defined(EAS_WT_SYNTH)
+#define NUM_WT_VOICES MAX_SYNTH_VOICES
+
+/* FM on MCU */
+#elif defined(EAS_FM_SYNTH)
+#define NUM_FM_VOICES MAX_SYNTH_VOICES
+
+/* wavetable drums on MCU, wavetable melodic on DSP */
+#elif defined(EAS_SPLIT_WT_SYNTH)
+#define NUM_WT_VOICES MAX_SYNTH_VOICES
+
+/* wavetable drums and FM melodic on MCU */
+#elif defined(EAS_HYBRID_SYNTH)
+#define NUM_WT_VOICES NUM_PRIMARY_VOICES
+#define NUM_FM_VOICES NUM_SECONDARY_VOICES
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_SPLIT_HYBRID_SYNTH)
+#define NUM_WT_VOICES NUM_PRIMARY_VOICES
+#define NUM_FM_VOICES NUM_SECONDARY_VOICES
+
+/* FM synth on DSP */
+#elif defined(EAS_SPLIT_FM_SYNTH)
+#define NUM_FM_VOICES MAX_SYNTH_VOICES
+
+#else
+#error "Unrecognized architecture option"
+#endif
+
+#define NUM_SYNTH_CHANNELS 16
+
+#define DEFAULT_SYNTH_VOICES MAX_SYNTH_VOICES
+
+/* use the following values to specify unassigned channels or voices */
+#define UNASSIGNED_SYNTH_CHANNEL NUM_SYNTH_CHANNELS
+#define UNASSIGNED_SYNTH_VOICE MAX_SYNTH_VOICES
+
+
+/* synth parameters are updated every SYNTH_UPDATE_PERIOD_IN_SAMPLES */
+#define SYNTH_UPDATE_PERIOD_IN_SAMPLES (EAS_I32)(0x1L << SYNTH_UPDATE_PERIOD_IN_BITS)
+
+/* stealing weighting factors */
+#define NOTE_AGE_STEAL_WEIGHT 1
+#define NOTE_GAIN_STEAL_WEIGHT 4
+#define CHANNEL_POLY_STEAL_WEIGHT 12
+#define CHANNEL_PRIORITY_STEAL_WEIGHT 2
+#define NOTE_MATCH_PENALTY 128
+#define SYNTH_PRIORITY_WEIGHT 8
+
+/* default synth master volume */
+#define DEFAULT_SYNTH_MASTER_VOLUME 0x7fff
+
+#define DEFAULT_SYNTH_PRIORITY 5
+
+/* default tuning values */
+#define DEFAULT_PITCH_BEND_SENSITIVITY 200 /* 2 semitones */
+#define DEFAULT_FINE_PITCH 0 /* 0 cents */
+#define DEFAULT_COARSE_PITCH 0 /* 0 semitones */
+
+/* default drum channel is 10, but is internally 9 due to unit offset */
+#define DEFAULT_DRUM_CHANNEL 9
+
+/* drum channel can simultaneously play this many voices at most */
+#define DEFAULT_CHANNEL_POLYPHONY_LIMIT 2
+
+/* default instrument is acoustic piano */
+#define DEFAULT_MELODY_BANK_MSB 0x79
+#define DEFAULT_RHYTHM_BANK_MSB 0x78
+#define DEFAULT_MELODY_BANK_NUMBER (DEFAULT_MELODY_BANK_MSB << 8)
+#define DEFAULT_RHYTHM_BANK_NUMBER (DEFAULT_RHYTHM_BANK_MSB << 8)
+#define DEFAULT_SYNTH_PROGRAM_NUMBER 0
+
+#define DEFAULT_PITCH_BEND 0x2000 /* 0x2000 == (0x40 << 7) | 0x00 */
+#define DEFAULT_MOD_WHEEL 0
+#define DEFAULT_CHANNEL_VOLUME 0x64
+#define DEFAULT_PAN 0x40 /* decimal 64, center */
+
+#ifdef _REVERB
+#define DEFAULT_REVERB_SEND 40 /* some reverb */
+#endif
+
+#ifdef _CHORUS
+#define DEFAULT_CHORUS_SEND 0 /* no chorus */
+#endif
+
+#define DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY 0 /* EAS synth uses a different default */
+#define DEFAULT_FILTER_RESONANCE 0
+#define DEFAULT_EXPRESSION 0x7F
+
+#define DEFAULT_CHANNEL_PRESSURE 0
+
+#define DEFAULT_REGISTERED_PARAM 0x3FFF
+
+#define DEFAULT_CHANNEL_STATIC_GAIN 0
+#define DEFAULT_CHANNEL_STATIC_PITCH 0
+
+#define DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS 50
+#define DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS 50
+
+#define DEFAULT_KEY_NUMBER 0x69
+#define DEFAULT_VELOCITY 0x64
+#define DEFAULT_REGION_INDEX 0
+#define DEFAULT_ARTICULATION_INDEX 0
+#define DEFAULT_VOICE_GAIN 0
+#define DEFAULT_AGE 0
+#define DEFAULT_SP_MIDI_PRIORITY 16
+
+
+/* filter defines */
+#define DEFAULT_FILTER_ZERO 0
+#define FILTER_CUTOFF_MAX_PITCH_CENTS 1919
+#define FILTER_CUTOFF_MIN_PITCH_CENTS -4467
+#define A5_PITCH_OFFSET_IN_CENTS 6900
+
+/*------------------------------------
+ * S_SYNTH_CHANNEL data structure
+ *------------------------------------
+*/
+
+/* S_SYNTH_CHANNEL.m_nFlags */
+#define CHANNEL_FLAG_SUSTAIN_PEDAL 0x01
+#define CHANNEL_FLAG_MUTE 0x02
+#define CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS 0x04
+#define CHANNEL_FLAG_RHYTHM_CHANNEL 0x08
+#define CHANNEL_FLAG_EXTERNAL_AUDIO 0x10
+#define DEFAULT_CHANNEL_FLAGS 0
+
+/* macros for extracting virtual synth and channel numbers */
+#define GET_VSYNTH(a) ((a) >> 4)
+#define GET_CHANNEL(a) ((a) & 15)
+
+typedef struct s_synth_channel_tag
+{
+ /* use static channel parameters to reduce MIPs */
+ /* parameters shared by multiple voices assigned to same channel */
+ EAS_I32 staticPitch; /* (pitch bend * pitch sens) + fine pitch */
+ EAS_I16 staticGain; /* (CC7 * CC11 * master vol)^2 */
+
+ EAS_U16 regionIndex; /* index of first region in program */
+
+ EAS_U16 bankNum; /* play programs from this bank */
+ EAS_I16 pitchBend; /* pitch wheel value */
+ EAS_I16 pitchBendSensitivity;
+ EAS_I16 registeredParam; /* currently selected registered param */
+
+
+#if defined(_FM_SYNTH)
+ EAS_I16 lfoAmt; /* amount of LFO to apply to voice */
+#endif
+
+ EAS_U8 programNum; /* play this instrument number */
+ EAS_U8 modWheel; /* CC1 */
+ EAS_U8 volume; /* CC7 */
+ EAS_U8 pan; /* CC10 */
+
+ EAS_U8 expression; /* CC11 */
+
+ /* the following parameters are controlled by RPNs */
+ EAS_I8 finePitch;
+ EAS_I8 coarsePitch;
+
+ EAS_U8 channelPressure; /* applied to all voices on a given channel */
+
+ EAS_U8 channelFlags; /* bit field channelFlags for */
+ /* CC64, SP-MIDI channel masking */
+
+ EAS_U8 pool; /* SPMIDI channel voice pool */
+ EAS_U8 mip; /* SPMIDI MIP setting */
+
+#ifdef _REVERB
+ EAS_U8 reverbSend; /* CC91 */
+#endif
+
+#ifdef _CHORUS
+ EAS_U8 chorusSend; /* CC93 */
+#endif
+} S_SYNTH_CHANNEL;
+
+/*------------------------------------
+ * S_SYNTH_VOICE data structure
+ *------------------------------------
+*/
+
+/* S_SYNTH_VOICE.m_nFlags */
+#define VOICE_FLAG_UPDATE_VOICE_PARAMETERS 0x01
+#define VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF 0x02
+#define VOICE_FLAG_DEFER_MIDI_NOTE_OFF 0x04
+#define VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET 0x08
+#define VOICE_FLAG_DEFER_MUTE 0x40
+#define DEFAULT_VOICE_FLAGS 0
+
+/* S_SYNTH_VOICE.m_eState */
+typedef enum {
+
+ eVoiceStateFree = 0,
+ eVoiceStateStart,
+ eVoiceStatePlay,
+ eVoiceStateRelease,
+ eVoiceStateMuting,
+ eVoiceStateStolen,
+ eVoiceStateInvalid /* should never be in this state! */
+
+} E_VOICE_STATE;
+#define DEFAULT_VOICE_STATE eVoiceStateFree
+
+typedef struct s_synth_voice_tag
+{
+
+/* These parameters are common to both wavetable and FM
+ * synthesizers. The voice manager should only access this data.
+ * Any other data should be manipulated by the code that is
+ * specific to that synthesizer and reflected back through the
+ * common state data available here.
+ */
+ EAS_U16 regionIndex; /* index to wave and playback params */
+ EAS_I16 gain; /* current gain */
+ EAS_U16 age; /* large value means old note */
+ EAS_U16 nextRegionIndex; /* index to wave and playback params */
+ EAS_U8 voiceState; /* current voice state */
+ EAS_U8 voiceFlags; /* misc flags/bit fields */
+ EAS_U8 channel; /* this voice plays on this synth channel */
+ EAS_U8 note; /* 12 <= key number <= 108 */
+ EAS_U8 velocity; /* 0 <= velocity <= 127 */
+ EAS_U8 nextChannel; /* play stolen voice on this channel */
+ EAS_U8 nextNote; /* 12 <= key number <= 108 */
+ EAS_U8 nextVelocity; /* 0 <= velocity <= 127 */
+} S_SYNTH_VOICE;
+
+/*------------------------------------
+ * S_SYNTH data structure
+ *
+ * One instance for each MIDI stream
+ *------------------------------------
+*/
+
+/* S_SYNTH.m_nFlags */
+#define SYNTH_FLAG_RESET_IS_REQUESTED 0x01
+#define SYNTH_FLAG_SP_MIDI_ON 0x02
+#define SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS 0x04
+#define SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING 0x08
+#define DEFAULT_SYNTH_FLAGS SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS
+
+typedef struct s_synth_tag
+{
+ struct s_eas_data_tag *pEASData;
+ const S_EAS *pEAS;
+
+#ifdef DLS_SYNTHESIZER
+ S_DLS *pDLS;
+#endif
+
+#ifdef EXTERNAL_AUDIO
+ EAS_EXT_PRG_CHG_FUNC cbProgChgFunc;
+ EAS_EXT_EVENT_FUNC cbEventFunc;
+ EAS_VOID_PTR *pExtAudioInstData;
+#endif
+
+ S_SYNTH_CHANNEL channels[NUM_SYNTH_CHANNELS];
+ EAS_I32 totalNoteCount;
+ EAS_U16 maxPolyphony;
+ EAS_U16 numActiveVoices;
+ EAS_U16 masterVolume;
+ EAS_U8 channelsByPriority[NUM_SYNTH_CHANNELS];
+ EAS_U8 poolCount[NUM_SYNTH_CHANNELS];
+ EAS_U8 poolAlloc[NUM_SYNTH_CHANNELS];
+ EAS_U8 synthFlags;
+ EAS_I8 globalTranspose;
+ EAS_U8 vSynthNum;
+ EAS_U8 refCount;
+ EAS_U8 priority;
+} S_SYNTH;
+
+/*------------------------------------
+ * S_VOICE_MGR data structure
+ *
+ * One instance for each EAS library instance
+ *------------------------------------
+*/
+typedef struct s_voice_mgr_tag
+{
+ S_SYNTH *pSynth[MAX_VIRTUAL_SYNTHESIZERS];
+ EAS_PCM voiceBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+
+#ifdef _FM_SYNTH
+ EAS_PCM operMixBuffer[SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+ S_FM_VOICE fmVoices[NUM_FM_VOICES];
+#endif
+
+#ifdef _WT_SYNTH
+ S_WT_VOICE wtVoices[NUM_WT_VOICES];
+#endif
+
+#ifdef _REVERB
+ EAS_PCM reverbSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+#endif
+
+#ifdef _CHORUS
+ EAS_PCM chorusSendBuffer[NUM_OUTPUT_CHANNELS * SYNTH_UPDATE_PERIOD_IN_SAMPLES];
+#endif
+ S_SYNTH_VOICE voices[MAX_SYNTH_VOICES];
+
+ EAS_SNDLIB_HANDLE pGlobalEAS;
+
+#ifdef DLS_SYNTHESIZER
+ S_DLS *pGlobalDLS;
+#endif
+
+#ifdef _SPLIT_ARCHITECTURE
+ EAS_FRAME_BUFFER_HANDLE pFrameBuffer;
+#endif
+
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+ EAS_U16 maxPolyphonyPrimary;
+ EAS_U16 maxPolyphonySecondary;
+#endif
+
+ EAS_I32 workload;
+ EAS_I32 maxWorkLoad;
+
+ EAS_U16 activeVoices;
+ EAS_U16 maxPolyphony;
+
+ EAS_U16 age;
+
+/* limits the number of voice starts in a frame for split architecture */
+#ifdef MAX_VOICE_STARTS
+ EAS_U16 numVoiceStarts;
+#endif
+} S_VOICE_MGR;
+
+#endif /* #ifdef _EAS_SYNTH_H */
+
+
diff --git a/arm-wt-22k/lib_src/eas_synth_protos.h b/arm-wt-22k/lib_src/eas_synth_protos.h new file mode 100644 index 0000000..a2ef10d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_synth_protos.h @@ -0,0 +1,60 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synth_protos.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for synth.
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTH_PROTOS_H
+#define _EAS_SYNTH_PROTOS_H
+
+/* includes */
+#include "eas_data.h"
+#include "eas_sndlib.h"
+
+#ifdef _SPLIT_ARCHITECTURE
+typedef struct s_frame_interface_tag
+{
+ EAS_BOOL (* EAS_CONST pfStartFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+ EAS_BOOL (* EAS_CONST pfEndFrame)(EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain);
+} S_FRAME_INTERFACE;
+#endif
+
+/* generic synthesizer interface */
+typedef struct
+{
+ EAS_RESULT (* EAS_CONST pfInitialize)(S_VOICE_MGR *pVoiceMgr);
+ EAS_RESULT (* EAS_CONST pfStartVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
+ EAS_BOOL (* EAS_CONST pfUpdateVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples);
+ void (* EAS_CONST pfReleaseVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+ void (* EAS_CONST pfMuteVoice)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+ void (* EAS_CONST pfSustainPedal)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
+ void (* EAS_CONST pfUpdateChannel)(S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+} S_SYNTH_INTERFACE;
+
+#endif
+
+
+
diff --git a/arm-wt-22k/lib_src/eas_synthcfg.h b/arm-wt-22k/lib_src/eas_synthcfg.h new file mode 100644 index 0000000..2491e6d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_synthcfg.h @@ -0,0 +1,70 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_synthcfg.h
+ *
+ * Contents and purpose:
+ * Defines for various synth configurations
+ *
+ * Copyright Sonic Network Inc. 2004, 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 664 $
+ * $Date: 2007-04-25 13:11:22 -0700 (Wed, 25 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_SYNTHCFG_H
+#define _EAS_SYNTHCFG_H
+
+#if defined(EAS_WT_SYNTH)
+#define _WT_SYNTH
+
+/* FM on MCU */
+#elif defined(EAS_FM_SYNTH)
+#define _FM_SYNTH
+
+/* wavetable drums and FM melodic on MCU */
+#elif defined(EAS_HYBRID_SYNTH)
+#define _WT_SYNTH
+#define _FM_SYNTH
+#define _SECONDARY_SYNTH
+#define _HYBRID_SYNTH
+
+/* wavetable drums on MCU, wavetable melodic on DSP */
+#elif defined(EAS_SPLIT_WT_SYNTH)
+#define _WT_SYNTH
+#define _SPLIT_ARCHITECTURE
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_SPLIT_HYBRID_SYNTH)
+#define _WT_SYNTH
+#define _FM_SYNTH
+#define _SECONDARY_SYNTH
+#define _SPLIT_ARCHITECTURE
+#define _HYBRID_SYNTH
+
+/* FM synth on DSP */
+#elif defined(EAS_SPLIT_FM_SYNTH)
+#define _FM_SYNTH
+#define _SPLIT_ARCHITECTURE
+
+#else
+#error "Unrecognized architecture option"
+#endif
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_tcdata.c b/arm-wt-22k/lib_src/eas_tcdata.c new file mode 100644 index 0000000..11ff9c7 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_tcdata.c @@ -0,0 +1,43 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_tcdata.c
+ *
+ * Contents and purpose:
+ * ToneControl Parser data
+ *
+ * This file contains static data for the ToneControl parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_types.h"
+#include "eas_tcdata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_iMelodyData
+ *
+ * Static memory allocation for iMelody parser
+ *----------------------------------------------------------------------------
+*/
+S_TC_DATA eas_TCData;
+
diff --git a/arm-wt-22k/lib_src/eas_tcdata.h b/arm-wt-22k/lib_src/eas_tcdata.h new file mode 100644 index 0000000..51a60b2 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_tcdata.h @@ -0,0 +1,65 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_tcdata.h
+ *
+ * Contents and purpose:
+ * SMF File Parser
+ *
+ * This file contains data declarations for the ToneControl parser.
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef EAS_TFDATA_H
+#define EAS_TCDATA_H
+
+#include "eas_data.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * S_TC_DATA
+ *
+ * This structure contains the state data for the ToneControl parser
+ *----------------------------------------------------------------------------
+*/
+typedef struct
+{
+ EAS_FILE_HANDLE fileHandle; /* file handle */
+ S_SYNTH *pSynth; /* synthesizer handle */
+ EAS_I32 fileOffset; /* offset to start of data */
+ EAS_I32 time; /* current time in 256ths of a msec */
+ EAS_I32 tick; /* tick based on current tempo and resolution */
+ EAS_I32 length; /* length of current note */
+ EAS_I32 restorePos; /* return to here after block */
+ EAS_U8 state; /* current state EAS_STATE_XXXX */
+ EAS_U8 volume; /* volume */
+ EAS_I8 note; /* current note */
+ EAS_I8 repeatCount; /* note repeat counter */
+ EAS_I8 tempo; /* tempo from file (bpm = tempo * 4) */
+ EAS_I8 resolution; /* resolution from file */
+ EAS_I8 dataByte; /* storage for characters that are "put back" */
+ EAS_BOOL8 byteAvail; /* char in "put back" buffer */
+} S_TC_DATA;
+
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_tonecontrol.c b/arm-wt-22k/lib_src/eas_tonecontrol.c new file mode 100644 index 0000000..8231b83 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_tonecontrol.c @@ -0,0 +1,941 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_tonecontrol.c
+ *
+ * Contents and purpose:
+ * MMAPI ToneControl parser
+ *
+ * Copyright Sonic Network Inc. 2006
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 795 $
+ * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_miditypes.h"
+#include "eas_parser.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_midi.h"
+#include "eas_config.h"
+#include "eas_vm_protos.h"
+#include "eas_tcdata.h"
+
+
+/* default channel and program for TC playback */
+#define TC_CHANNEL 0
+#define TC_PROGRAM 80
+#define TC_VELOCITY 127
+
+#define TC_FIELD_SILENCE -1
+#define TC_FIELD_VERSION -2
+#define TC_FIELD_TEMPO -3
+#define TC_FIELD_RESOLUTION -4
+#define TC_FIELD_BLOCK_START -5
+#define TC_FIELD_BLOCK_END -6
+#define TC_FIELD_PLAY_BLOCK -7
+#define TC_FIELD_SET_VOLUME -8
+#define TC_FIELD_REPEAT -9
+#define TC_FIELD_INVALID -10
+
+/* convert 0-100 volume to 0-127 velocity using fixed point */
+#define TC_VOLUME_CONV 21307064
+#define TC_VOLUME_SHIFT 24
+
+
+/* local prototypes */
+static EAS_RESULT TC_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+static EAS_RESULT TC_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT TC_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+static EAS_RESULT TC_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+static EAS_RESULT TC_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT TC_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT TC_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT TC_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT TC_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT TC_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT TC_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT TC_ParseHeader (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_StartNote (S_EAS_DATA *pEASData, S_TC_DATA* pData, EAS_INT parserMode, EAS_I8 note);
+static EAS_RESULT TC_GetRepeat (S_EAS_DATA *pEASData, S_TC_DATA* pData, EAS_INT parserMode);
+static EAS_RESULT TC_PlayBlock (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_BlockEnd (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_GetVolume (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_GetTempo (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_GetResolution (S_EAS_DATA *pEASData, S_TC_DATA* pData);
+static EAS_RESULT TC_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_TC_DATA *pData, EAS_I8 *pValue);
+static void TC_PutBackChar (S_TC_DATA *pData, EAS_I8 value);
+
+/* calculate a new tick time based on resolution & tempo */
+EAS_INLINE void TC_CalcTimeBase (S_TC_DATA *pData)
+{
+
+ /* ticks in 256ths of a millisecond */
+ pData->tick = ((60 * 1000) << 8) / (pData->tempo * pData->resolution);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_TC_Parser
+ *
+ * This structure contains the functional interface for the iMelody parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_TC_Parser =
+{
+ TC_CheckFileType,
+ TC_Prepare,
+ TC_Time,
+ TC_Event,
+ TC_State,
+ TC_Close,
+ TC_Reset,
+ TC_Pause,
+ TC_Resume,
+ NULL,
+ TC_SetData,
+ TC_GetData,
+ NULL
+};
+
+/*----------------------------------------------------------------------------
+ * TC_CheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset)
+{
+ S_TC_DATA data;
+ S_TC_DATA *pData;
+
+ /* init data */
+ EAS_HWMemSet(&data, 0, sizeof(S_TC_DATA));
+ data.fileHandle = fileHandle;
+ data.fileOffset = offset;
+ *ppHandle= NULL;
+
+ /* see if we can parse the header */
+ if (TC_ParseHeader(pEASData, &data) == EAS_SUCCESS)
+ {
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pData = EAS_CMEnumOptData(EAS_MODULE_MMAPI_TONE_CONTROL);
+ else
+ pData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_TC_DATA));
+ if (!pData)
+ return EAS_ERROR_MALLOC_FAILED;
+
+ /* copy data to persistent storage */
+ EAS_HWMemCpy(pData, &data, sizeof(S_TC_DATA));
+
+ /* return a pointer to the instance data */
+ pData->state = EAS_STATE_OPEN;
+ *ppHandle = pData;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Prepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_TC_DATA* pData;
+ EAS_RESULT result;
+
+ /* check for valid state */
+ pData = (S_TC_DATA*) pInstData;
+ if (pData->state != EAS_STATE_OPEN)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* instantiate a synthesizer */
+ if ((result = VMInitMIDI(pEASData, &pData->pSynth)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI returned %d\n", result); */ }
+ return result;
+ }
+
+ /* set to ready state */
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Time()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the time of the next event in msecs
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pTime - pointer to variable to hold time of next event (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT TC_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime)
+{
+ S_TC_DATA *pData;
+
+ pData = (S_TC_DATA*) pInstData;
+
+ /* return time in milliseconds */
+ /*lint -e{704} use shift instead of division */
+ *pTime = pData->time >> 8;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Event()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the next event in the file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode)
+{
+ S_TC_DATA* pData;
+ EAS_RESULT result;
+ EAS_I8 temp;
+
+ pData = (S_TC_DATA*) pInstData;
+ if (pData->state >= EAS_STATE_OPEN)
+ return EAS_SUCCESS;
+
+ /* initialize MIDI channel when the track starts playing */
+ if (pData->time == 0)
+ {
+ /* set program to square lead */
+ VMProgramChange(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, TC_PROGRAM);
+
+ /* set channel volume to max */
+ VMControlChange(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, 7, 127);
+ }
+
+ /* check for end of note */
+ if (pData->note >= 0)
+ {
+ /* stop the note */
+ VMStopNote(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, (EAS_U8) pData->note, 0);
+
+ /* check for repeat note */
+ if (pData->repeatCount)
+ {
+ pData->repeatCount--;
+ pData->time += pData->length;
+ if ((pData->note >= 0) && (parserMode == eParserModePlay))
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, (EAS_U8) pData->note, pData->volume);
+ return EAS_SUCCESS;
+ }
+
+ pData->note = TC_FIELD_SILENCE;
+ }
+
+ /* parse stream until we get a note or rest */
+ for (;;)
+ {
+
+ /* get next byte from stream */
+ if ((result = TC_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ {
+ if (result == EAS_EOF)
+ {
+ pData->state = EAS_STATE_STOPPING;
+ return EAS_SUCCESS;
+ }
+ break;
+ }
+
+ /* check for musical events */
+ if (temp >= TC_FIELD_SILENCE)
+ {
+ result = TC_StartNote(pEASData, pData, parserMode, temp);
+ break;
+ }
+
+ /* must be a control field */
+ switch (temp)
+ {
+ case TC_FIELD_TEMPO:
+ result = TC_GetTempo(pEASData, pData);
+ break;
+
+ case TC_FIELD_RESOLUTION:
+ result = TC_GetResolution(pEASData, pData);
+ break;
+
+ case TC_FIELD_SET_VOLUME:
+ result = TC_GetVolume(pEASData, pData);
+ break;
+
+ case TC_FIELD_REPEAT:
+ result = TC_GetRepeat(pEASData, pData, parserMode);
+ break;
+
+ case TC_FIELD_PLAY_BLOCK:
+ result = TC_PlayBlock(pEASData, pData);
+ break;
+
+ case TC_FIELD_BLOCK_START:
+ result = TC_GetNextChar(pEASData->hwInstData, pData, &temp);
+ break;
+
+ case TC_FIELD_BLOCK_END:
+ result = TC_BlockEnd(pEASData, pData);
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unexpected byte 0x%02x in ToneControl stream\n", temp); */ }
+ result = EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* check for error */
+ if (result != EAS_SUCCESS)
+ break;
+ }
+
+ /* check for error */
+ if (result != EAS_SUCCESS)
+ {
+ if (result == EAS_EOF)
+ result = EAS_ERROR_FILE_FORMAT;
+ pData->state = EAS_STATE_ERROR;
+ }
+ else
+ pData->state = EAS_STATE_PLAY;
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_State()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT TC_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState)
+{
+ S_TC_DATA* pData;
+
+ /* establish pointer to instance data */
+ pData = (S_TC_DATA*) pInstData;
+
+ /* if stopping, check to see if synth voices are active */
+ if (pData->state == EAS_STATE_STOPPING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_STOPPED;
+ }
+
+ if (pData->state == EAS_STATE_PAUSING)
+ {
+ if (VMActiveVoices(pData->pSynth) == 0)
+ pData->state = EAS_STATE_PAUSED;
+ }
+
+ /* return current state */
+ *pState = pData->state;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Close()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_TC_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_TC_DATA*) pInstData;
+
+ /* close the file */
+ if ((result = EAS_HWCloseFile(pEASData->hwInstData, pData->fileHandle)) != EAS_SUCCESS)
+ return result;
+
+ /* free the synth */
+ if (pData->pSynth != NULL)
+ VMMIDIShutdown(pEASData, pData->pSynth);
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pData);
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_TC_DATA* pData;
+ EAS_RESULT result;
+
+ pData = (S_TC_DATA*) pInstData;
+
+ /* reset the synth */
+ VMReset(pEASData->pVoiceMgr, pData->pSynth, EAS_TRUE);
+
+ /* reset time to zero */
+ pData->time = 0;
+
+ /* reset file position and re-parse header */
+ pData->state = EAS_STATE_ERROR;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+ if ((result = TC_ParseHeader (pEASData, pData)) != EAS_SUCCESS)
+ return result;
+
+ pData->state = EAS_STATE_READY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Pause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Pauses the sequencer. Mutes all voices and sets state to pause.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_TC_DATA *pData;
+
+ /* can't pause a stopped stream */
+ pData = (S_TC_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* mute the synthesizer */
+ VMMuteAllVoices(pEASData->pVoiceMgr, pData->pSynth);
+ pData->state = EAS_STATE_PAUSING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_Resume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume playing after a pause, sets state back to playing.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT TC_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_TC_DATA *pData;
+
+ /* can't resume a stopped stream */
+ pData = (S_TC_DATA*) pInstData;
+ if (pData->state == EAS_STATE_STOPPED)
+ return EAS_ERROR_ALREADY_STOPPED;
+
+ /* nothing to do but resume playback */
+ pData->state = EAS_STATE_PLAY;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_SetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData, pInstData, value) reserved for future use */
+static EAS_RESULT TC_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ /* we don't parse any metadata, but we need to return success here */
+ if (param == PARSER_DATA_METADATA_CB)
+ return EAS_SUCCESS;
+
+ return EAS_ERROR_INVALID_PARAMETER;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Return file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -e{715} common with other parsers */
+static EAS_RESULT TC_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_TC_DATA *pData;
+
+ pData = (S_TC_DATA *) pInstData;
+ switch (param)
+ {
+ /* return file type as TC */
+ case PARSER_DATA_FILE_TYPE:
+ *pValue = EAS_FILE_MMAPI_TONE_CONTROL;
+ break;
+
+ case PARSER_DATA_SYNTH_HANDLE:
+ *pValue = (EAS_I32) pData->pSynth;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_ParseHeader()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_ParseHeader (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_RESULT result;
+ EAS_I8 temp;
+
+ /* initialize some defaults */
+ pData->time = 0;
+ pData->tempo = 120;
+ pData->resolution = 64;
+ pData->volume = 127;
+ pData->repeatCount = 0;
+ pData->note = TC_FIELD_SILENCE;
+ pData->byteAvail = EAS_FALSE;
+
+ /* set default timebase */
+ TC_CalcTimeBase(pData);
+
+ /* seek to start of data */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* get version */
+ if ((result = TC_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for version number */
+ if (temp == TC_FIELD_VERSION)
+ {
+ TC_GetNextChar(pEASData->hwInstData, pData, &temp);
+// { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "ToneControl sequence version %d\n", temp); */ }
+ }
+ else
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* parse the header data until we find the first note or block */
+ for (;;)
+ {
+
+ /* get next byte from stream */
+ if ((result = TC_GetNextChar(pEASData->hwInstData, pData, &temp)) != EAS_SUCCESS)
+ return result;
+
+ /* check for tempo */
+ if (temp == TC_FIELD_TEMPO)
+ {
+ if ((result = TC_GetTempo(pEASData, pData)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* or resolution */
+ else if (temp == TC_FIELD_TEMPO)
+ {
+ if ((result = TC_GetResolution(pEASData, pData)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* must be music data */
+ else if (temp > TC_FIELD_INVALID)
+ {
+ TC_PutBackChar(pData, temp);
+ return EAS_SUCCESS;
+ }
+
+ /* unknown codes */
+ else
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unexpected byte 0x%02x in ToneControl stream\n", temp); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * TC_StartNote()
+ *----------------------------------------------------------------------------
+ * Process a note or silence event
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_StartNote (S_EAS_DATA *pEASData, S_TC_DATA* pData, EAS_INT parserMode, EAS_I8 note)
+{
+ EAS_I8 duration;
+
+ /* get the duration */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &duration) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* calculate time of next event */
+ pData->length = (EAS_I32) duration * pData->tick;
+ pData->time += pData->length;
+
+ /* start the note */
+ if ((note >= 0) && (parserMode == eParserModePlay))
+ {
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, (EAS_U8) note, pData->volume);
+ pData->note = note;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetRepeat()
+ *----------------------------------------------------------------------------
+ * Process a repeat code
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_GetRepeat (S_EAS_DATA *pEASData, S_TC_DATA* pData, EAS_INT parserMode)
+{
+ EAS_I8 count;
+
+ /* get the repeat count */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &count) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* validiate it */
+ if (count < 2)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* calculate time of next event */
+ pData->time += pData->length;
+ pData->repeatCount = count - 2;
+
+ /* start the note */
+ if ((pData->note >= 0) && (parserMode == eParserModePlay))
+ VMStartNote(pEASData->pVoiceMgr, pData->pSynth, TC_CHANNEL, (EAS_U8) pData->note, pData->volume);
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_PlayBlock()
+ *----------------------------------------------------------------------------
+ * Play a block of notes
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_PlayBlock (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_RESULT result;
+ EAS_I8 blockNum;
+ EAS_I8 temp;
+ EAS_I8 temp2;
+
+ /* get the block number */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &blockNum) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* validiate it */
+ if (blockNum < 0)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* save the current position */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, pData->fileHandle, &pData->restorePos)) != EAS_SUCCESS)
+ return result;
+
+ /* return to start of file */
+ pData->byteAvail = EAS_FALSE;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->fileOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* find the block */
+ for (;;)
+ {
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &temp) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &temp2) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ if ((temp == TC_FIELD_BLOCK_START) && (temp2 == blockNum))
+ return EAS_SUCCESS;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * TC_BlockEnd()
+ *----------------------------------------------------------------------------
+ * Handle end of block
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_BlockEnd (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_I8 blockNum;
+
+ /* get the block number */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &blockNum) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* validiate it */
+ if (blockNum < 0)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* if we were playing this block, restore to previous position */
+ pData->byteAvail = EAS_FALSE;
+ return EAS_HWFileSeek(pEASData->hwInstData, pData->fileHandle, pData->restorePos);
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetVolume()
+ *----------------------------------------------------------------------------
+ * Get the volume field and process it
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_GetVolume (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_I8 volume;
+
+ /* get volume */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &volume) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+ if ((volume < 0) || (volume > 100))
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* save volume */
+ pData->volume = (EAS_U8) ((EAS_I32) (volume * TC_VOLUME_CONV + 1) >> TC_VOLUME_SHIFT);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetTempo()
+ *----------------------------------------------------------------------------
+ * Get the tempo field and process it
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_GetTempo (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_I8 tempo;
+
+ /* get tempo */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &tempo) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+ if (tempo < 5)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* save tempo */
+ pData->tempo = tempo;
+
+ /* calculate new timebase */
+ TC_CalcTimeBase(pData);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetResolution()
+ *----------------------------------------------------------------------------
+ * Get the resolution field and process it
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_GetResolution (S_EAS_DATA *pEASData, S_TC_DATA* pData)
+{
+ EAS_I8 resolution;
+
+ /* get resolution */
+ if (TC_GetNextChar(pEASData->hwInstData, pData, &resolution) != EAS_SUCCESS)
+ return EAS_ERROR_FILE_FORMAT;
+ if (resolution < 0)
+ return EAS_ERROR_FILE_FORMAT;
+
+ /* save tempo */
+ pData->resolution = resolution;
+
+ /* calculate new timebase */
+ TC_CalcTimeBase(pData);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * TC_GetNextChar()
+ *----------------------------------------------------------------------------
+ * Fetch the next character from the stream
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT TC_GetNextChar (EAS_HW_DATA_HANDLE hwInstData, S_TC_DATA *pData, EAS_I8 *pValue)
+{
+
+ /* get character from "put back" buffer */
+ if (pData->byteAvail)
+ {
+ pData->byteAvail = EAS_FALSE;
+ *pValue = pData->dataByte;
+ return EAS_SUCCESS;
+ }
+
+ /* get character from file */
+ return EAS_HWGetByte(hwInstData, pData->fileHandle, pValue);
+}
+
+/*----------------------------------------------------------------------------
+ * TC_PutBackChar()
+ *----------------------------------------------------------------------------
+ * Put back the character
+ *----------------------------------------------------------------------------
+*/
+static void TC_PutBackChar (S_TC_DATA *pData, EAS_I8 value)
+{
+
+ pData->dataByte = value;
+ pData->byteAvail = EAS_TRUE;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_vm_protos.h b/arm-wt-22k/lib_src/eas_vm_protos.h new file mode 100644 index 0000000..eb49ba8 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_vm_protos.h @@ -0,0 +1,1086 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_vm_protos.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for voice manager.
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 736 $
+ * $Date: 2007-06-22 13:51:24 -0700 (Fri, 22 Jun 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_VM_PROTOS_H
+#define _EAS_VM_PROTOS_H
+
+// includes
+#include "eas_data.h"
+#include "eas_sndlib.h"
+
+/*----------------------------------------------------------------------------
+ * VMInitialize()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMInitialize (S_EAS_DATA *pEASData);
+
+/*----------------------------------------------------------------------------
+ * VMInitMIDI()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMInitMIDI (S_EAS_DATA *pEASData, S_SYNTH **ppSynth);
+
+/*----------------------------------------------------------------------------
+ * VMInitializeAllChannels()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitializeAllChannels (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMResetControllers()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMResetControllers (S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMInitMIPTable()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Initialize the SP-MIDI MIP table
+ *
+ * Inputs:
+ * pEASData - pointer to synthesizer instance data
+ * mute - EAS_FALSE to unmute channels, EAS_TRUE to mute
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitMIPTable (S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMSetMIPEntry()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the priority and MIP level for a MIDI channel
+ *
+ * Inputs:
+ * pEASData - pointer to synthesizer instance data
+ * channel - MIDI channel number
+ * priority - priority (0-15 with 0 = highest priority)
+ * mip - maximum instantaneous polyphony
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetMIPEntry (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 priority, EAS_U8 mip);
+
+/*----------------------------------------------------------------------------
+ * VMUpdateMIPTable()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine is called when the polyphony count in the synthesizer changes
+ *
+ * Inputs:
+ * pEASData - pointer to synthesizer instance data
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMUpdateMIPTable (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMInitializeAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitializeAllVoices (S_VOICE_MGR *pVoiceMgr, EAS_INT vSynthNum);
+
+/*----------------------------------------------------------------------------
+ * VMStartNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the synth's state to play the requested note on the requested
+ * channel if possible.
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nKeyNumber - the MIDI key number for this note
+ * nNoteVelocity - the key velocity for this note
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMStartNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity);
+
+/*----------------------------------------------------------------------------
+ * VMCheckKeyGroup()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * If the note that we've been asked to start is in the same key group as
+ * any currently playing notes, then we must shut down the currently playing
+ * note in the same key group and then start the newly requested note.
+ *
+ * Inputs:
+ * nChannel - synth channel that wants to start a new note
+ * nKeyNumber - new note's midi note number
+ * nRegionIndex - calling routine finds this index and gives to us
+ * nNoteVelocity - new note's velocity
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pbVoiceStealingRequired - flag: this routine sets true if we needed to
+ * steal a voice
+ *
+ * Side Effects:
+ * gsSynthObject.m_sVoice[free voice num].m_nKeyNumber may be assigned
+ * gsSynthObject.m_sVoice[free voice num].m_nVelocity may be assigned
+ *----------------------------------------------------------------------------
+*/
+void VMCheckKeyGroup (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U16 keyGroup, EAS_U8 channel);
+
+/*----------------------------------------------------------------------------
+ * VMCheckPolyphonyLimiting()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We only play at most 2 of the same note on a MIDI channel.
+ * E.g., if we are asked to start note 36, and there are already two voices
+ * that are playing note 36, then we must steal the voice playing
+ * the oldest note 36 and use that stolen voice to play the new note 36.
+ *
+ * Inputs:
+ * nChannel - synth channel that wants to start a new note
+ * nKeyNumber - new note's midi note number
+ * nNoteVelocity - new note's velocity
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pbVoiceStealingRequired - flag: this routine sets true if we needed to
+ * steal a voice
+ * *
+ * Side Effects:
+ * psSynthObject->m_sVoice[free voice num].m_nKeyNumber may be assigned
+ * psSynthObject->m_sVoice[free voice num].m_nVelocity may be assigned
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMCheckPolyphonyLimiting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex, EAS_I32 lowVoice, EAS_I32 highVoice);
+
+/*----------------------------------------------------------------------------
+ * VMStopNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the synth's state to end the requested note on the requested
+ * channel.
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nKeyNumber - the key number of the note to stop
+ * nNoteVelocity - the note-off velocity
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * gsSynthObject.m_sVoice[free voice num].m_nSynthChannel may be assigned
+ * gsSynthObject.m_sVoice[free voice num].m_nKeyNumber is assigned
+ * gsSynthObject.m_sVoice[free voice num].m_nVelocity is assigned
+ *----------------------------------------------------------------------------
+*/
+void VMStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 key, EAS_U8 velocity);
+
+/*----------------------------------------------------------------------------
+ * VMFindAvailableVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Find an available voice and return the voice number if available.
+ *
+ * Inputs:
+ * pnVoiceNumber - really an output, see below
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pnVoiceNumber - returns the voice number of available voice if found
+ * success - if there is an available voice
+ * failure - otherwise
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMFindAvailableVoice (S_VOICE_MGR *pVoiceMgr, EAS_INT *pVoiceNumber, EAS_I32 lowVoice, EAS_I32 highVoice);
+
+/*----------------------------------------------------------------------------
+ * VMStealVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Steal a voice and return the voice number
+ *
+ * Stealing algorithm: steal the best choice with minimal work, taking into
+ * account SP-Midi channel priorities and polyphony allocation.
+ *
+ * In one pass through all the voices, figure out which voice to steal
+ * taking into account a number of different factors:
+ * Priority of the voice's MIDI channel
+ * Number of voices over the polyphony allocation for voice's MIDI channel
+ * Amplitude of the voice
+ * Note age
+ * Key velocity (for voices that haven't been started yet)
+ * If any matching notes are found
+ *
+ * Inputs:
+ * nChannel - the channel that this voice wants to be started on
+ * nKeyNumber - the key number for this new voice
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pnVoiceNumber - voice stolen
+ * EAS_RESULT EAS_SUCCESS - always successful
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMStealVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_INT *pVoiceNumber, EAS_U8 channel, EAS_U8 note, EAS_I32 lowVoice, EAS_I32 highVoice);
+
+/*----------------------------------------------------------------------------
+ * VMAddSamples()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesize the requested number of samples.
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to write to buffer
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * number of samples actually written to buffer
+ *
+ * Side Effects:
+ * - samples are added to the presently free buffer
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 VMAddSamples (S_VOICE_MGR *pVoiceMgr, EAS_I32 *pMixBuffer, EAS_I32 numSamplesToAdd);
+
+/*----------------------------------------------------------------------------
+ * VMProgramChange()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the instrument (program) for the given channel.
+ *
+ * Depending on the program number, and the bank selected for this channel, the
+ * program may be in ROM, RAM (from SMAF or CMX related RAM wavetable), or
+ * Alternate wavetable (from mobile DLS or other DLS file)
+ *
+ * This function figures out what wavetable should be used, and sets it up as the
+ * wavetable to use for this channel. Also the channel may switch from a melodic
+ * channel to a rhythm channel, or vice versa.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel].m_nProgramNumber is likely changed
+ * gsSynthObject.m_sChannel[nChannel].m_psEAS may be changed
+ * gsSynthObject.m_sChannel[nChannel].m_bRhythmChannel may be changed
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMProgramChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 program);
+
+/*----------------------------------------------------------------------------
+ * VMChannelPressure()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the channel pressure for the given channel
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nVelocity - the channel pressure value
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel].m_nChannelPressure is updated
+ *----------------------------------------------------------------------------
+*/
+void VMChannelPressure (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 value);
+
+/*----------------------------------------------------------------------------
+ * VMPitchBend()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the pitch wheel value for the given channel.
+ * This routine constructs the proper 14-bit argument when the calling routine
+ * passes the pitch LSB and MSB.
+ *
+ * Note: some midi disassemblers display a bipolar pitch bend value.
+ * We can display the bipolar value using
+ * if m_nPitchBend >= 0x2000
+ * bipolar pitch bend = postive (m_nPitchBend - 0x2000)
+ * else
+ * bipolar pitch bend = negative (0x2000 - m_nPitchBend)
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nPitchLSB - the LSB byte from the pitch bend message
+ * nPitchMSB - the MSB byte from the message
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel].m_nPitchBend is changed
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMPitchBend (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 pitchLSB, EAS_U8 pitchMSB);
+
+/*----------------------------------------------------------------------------
+ * VMControlChange()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the controller (or mode) for the given channel.
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nControllerNumber - the controller number
+ * nControlValue - the controller number for this control change
+ * nControlValue - the value for this control change
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel] controller is changed
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMControlChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value);
+
+/*----------------------------------------------------------------------------
+ * VMUpdateRPNStateMachine()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this function when we want to parse a stream of RPN messages.
+ * NOTE: The synth has only one set of global RPN data instead of RPN data
+ * per channel.
+ * So actually, we don't really need to look at the nChannel parameter,
+ * but we pass it to facilitate future upgrades. Furthermore, we only
+ * support RPN0 (pitch bend sensitivity), RPN1 (fine tuning) and
+ * RPN2 (coarse tuning). Any other RPNs are rejected.
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nControllerNumber - the RPN controller number
+ * nControlValue - the value for this control change
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * gsSynthObject.m_RPN0 (or m_RPN1 or m_RPN2) may be updated if the
+ * proper RPN message sequence is parsed.
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMUpdateRPNStateMachine (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value);
+
+/*----------------------------------------------------------------------------
+ * VMUpdateStaticChannelParameters()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update all of the static channel parameters for channels that have had
+ * a controller change values
+ * Or if the synth has signalled that all channels must forcibly
+ * be updated
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * none
+ *
+ * Side Effects:
+ * - psSynthObject->m_sChannel[].m_nStaticGain and m_nStaticPitch
+ * are updated for channels whose controller values have changed
+ * or if the synth has signalled that all channels must forcibly
+ * be updated
+ *----------------------------------------------------------------------------
+*/
+void VMUpdateStaticChannelParameters (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMReleaseAllDeferredNoteOffs()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this functin when the sustain flag is presently set but
+ * we are now transitioning from damper pedal on to
+ * damper pedal off. This means all notes in this channel
+ * that received a note off while the damper pedal was on, and
+ * had their note-off requests deferred, should now proceed to
+ * the release state.
+ *
+ * Inputs:
+ * nChannel - this channel has its sustain pedal transitioning from on to off
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * any voice with deferred note offs on this channel are updated such that
+ *
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMReleaseAllDeferredNoteOffs (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+
+/*----------------------------------------------------------------------------
+ * VMCatchNotesForSustainPedal()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this function when the sustain flag is presently clear and
+ * the damper pedal is off and we are transitioning from damper pedal OFF to
+ * damper pedal ON. Currently sounding notes should be left
+ * unchanged. However, we should try to "catch" notes if possible.
+ * If any notes have levels >= sustain level, catch them,
+ * otherwise, let them continue to release.
+ *
+ * Inputs:
+ * nChannel - this channel has its sustain pedal transitioning from on to off
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * any voice with deferred note offs on this channel are updated such that
+ * psVoice->m_sEG1.m_eState = eEnvelopeStateSustainPedal
+ *----------------------------------------------------------------------------
+*/
+void VMCatchNotesForSustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+
+/*----------------------------------------------------------------------------
+ * VMUpdateAllNotesAge()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Increment the note age for all voices older than the age of the voice
+ * that is stopping, effectively making the voices "younger".
+ *
+ * Inputs:
+ * nAge - age of voice that is going away
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * m_nAge for some voices is incremented
+ *----------------------------------------------------------------------------
+*/
+void VMUpdateAllNotesAge (S_VOICE_MGR *pVoiceMgr, EAS_U16 nAge);
+
+/*----------------------------------------------------------------------------
+ * VMFindRegionIndex()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Find the region index for the given instrument using the midi key number
+ * and the RPN2 (coarse tuning) value. By using RPN2 as part of the
+ * region selection process, we reduce the amount a given sample has
+ * to be transposed by selecting the closest recorded root instead.
+ *
+ * Inputs:
+ * nChannel - current channel for this note
+ * nKeyNumber - current midi note number
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pnRegionIndex - valid only if we returned success
+ * success if we found the region index number, otherwise
+ * failure
+ *
+ * Side Effects:
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMFindRegionIndex (S_VOICE_MGR *pVoiceMgr, EAS_U8 channel, EAS_U8 note, EAS_U16 *pRegionIndex);
+
+/*----------------------------------------------------------------------------
+ * VMIncRefCount()
+ *----------------------------------------------------------------------------
+ * Increment reference count for virtual synth
+ *----------------------------------------------------------------------------
+*/
+void VMIncRefCount (S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMReset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this routine to start the process of reseting the synth.
+ * This routine sets a flag for the entire synth indicating that we want
+ * to reset.
+ * We also force all voices to mute quickly.
+ * However, we do not actually perform any synthesis in this routine. That
+ * is, we do not ramp the voices down from this routine, but instead, we
+ * let the "regular" synth processing steps take care of adding the ramp
+ * down samples to the output buffer. After we are sure that all voices
+ * have completed ramping down, we continue the process of resetting the
+ * synth (from another routine).
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - set a flag (in gsSynthObject.m_nFlags) indicating synth reset requested.
+ * - force all voices to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMReset (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_BOOL force);
+
+/*----------------------------------------------------------------------------
+ * VMMuteAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this in an emergency reset situation.
+ * This forces all voices to mute quickly.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMMuteVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum);
+void VMMuteAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMReleaseAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this after we've encountered the end of the Midi file.
+ * This ensures all voice are either in release (because we received their
+ * note off already) or forces them to mute quickly.
+ * We use this as a safety to prevent bad midi files from playing forever.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices to update their envelope states to release or mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMReleaseAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMAllNotesOff()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Quickly mute all notes on the given channel.
+ *
+ * Inputs:
+ * nChannel - quickly turn off all notes on this channel
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices on this channel to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMAllNotesOff (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+
+/*----------------------------------------------------------------------------
+ * VMDeferredStopNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Stop the notes that had deferred note-off requests.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None.
+ *
+ * Side Effects:
+ * voices that have had deferred note-off requests are now put into release
+ * gsSynthObject.m_sVoice[i].m_nFlags has the VOICE_FLAG_DEFER_MIDI_NOTE_OFF
+ * cleared
+ *----------------------------------------------------------------------------
+*/
+void VMDeferredStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMSetSynthPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the synth to a new polyphony value. Value must be >= 1 and
+ * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * synth synthesizer number (0 = onboard, 1 = DSP)
+ * polyphonyCount desired polyphony count
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 polyphonyCount);
+
+/*----------------------------------------------------------------------------
+ * VMGetSynthPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the synth to a new polyphony value. Value must be >= 1 and
+ * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * synth synthesizer number (0 = onboard, 1 = DSP)
+ * polyphonyCount desired polyphony count
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMGetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 *pPolyphonyCount);
+
+/*----------------------------------------------------------------------------
+ * VMSetPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the virtual synth polyphony. 0 = no limit (i.e. can use
+ * all available voices).
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * polyphonyCount desired polyphony count
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 polyphonyCount);
+
+/*----------------------------------------------------------------------------
+ * VMGetPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current polyphony setting
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * pSynth pointer to virtual synth
+ * pPolyphonyCount pointer to variable to receive data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMGetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPolyphonyCount);
+
+/*----------------------------------------------------------------------------
+ * VMSetPriority()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the virtual synth priority
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * priority new priority
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 priority);
+
+/*----------------------------------------------------------------------------
+ * VMGetPriority()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get the virtual synth priority
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * pPriority pointer to variable to hold priority
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMGetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPriority);
+
+/*----------------------------------------------------------------------------
+ * VMSetVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the master volume for this sequence
+ *
+ * Inputs:
+ * nSynthVolume - the desired master volume
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * overrides any previously set master volume from sysex
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetVolume (S_SYNTH *pSynth, EAS_U16 masterVolume);
+
+/*----------------------------------------------------------------------------
+ * VMSetPitchBendRange()
+ *----------------------------------------------------------------------------
+ * Set the pitch bend range for the given channel.
+ *----------------------------------------------------------------------------
+*/
+void VMSetPitchBendRange (S_SYNTH *pSynth, EAS_INT channel, EAS_I16 pitchBendRange);
+
+/*----------------------------------------------------------------------------
+ * VMSetEASLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the pointer to the sound library
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetGlobalEASLib (S_VOICE_MGR *pVoiceMgr, EAS_SNDLIB_HANDLE pEAS);
+EAS_RESULT VMSetEASLib (S_SYNTH *pSynth, EAS_SNDLIB_HANDLE pEAS);
+
+#ifdef DLS_SYNTHESIZER
+/*----------------------------------------------------------------------------
+ * VMSetDLSLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the pointer to the sound library
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetGlobalDLSLib (EAS_DATA_HANDLE pEASData, EAS_DLSLIB_HANDLE pDLS);
+EAS_RESULT VMSetDLSLib (S_SYNTH *pSynth, EAS_DLSLIB_HANDLE pDLS);
+#endif
+
+/*----------------------------------------------------------------------------
+ * VMSetTranposition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the global key transposition used by the synthesizer.
+ * Transposes all melodic instruments up or down by the specified
+ * amount. Range is limited to +/-12 semitones.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ * transposition - transpose amount (+/-12)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetTranposition (S_SYNTH *pSynth, EAS_I32 transposition);
+
+/*----------------------------------------------------------------------------
+ * VMGetTranposition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Gets the global key transposition used by the synthesizer.
+ * Transposes all melodic instruments up or down by the specified
+ * amount. Range is limited to +/-12 semitones.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMGetTranposition (S_SYNTH *pSynth, EAS_I32 *pTransposition);
+
+/*----------------------------------------------------------------------------
+ * VMGetNoteCount()
+ *----------------------------------------------------------------------------
+* Returns the total note count
+*----------------------------------------------------------------------------
+*/
+EAS_I32 VMGetNoteCount (S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMRender()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine renders a frame of audio
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pVoicesRendered - number of voices rendered this frame
+ *
+ * Side Effects:
+ * sets psMidiObject->m_nMaxWorkloadPerFrame
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMRender (S_VOICE_MGR *pVoiceMgr, EAS_I32 numSamples, EAS_I32 *pMixBuffer, EAS_I32 *pVoicesRendered);
+
+/*----------------------------------------------------------------------------
+ * VMInitWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clears the workload counter
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitWorkload (S_VOICE_MGR *pVoiceMgr);
+
+/*----------------------------------------------------------------------------
+ * VMSetWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the max workload for a single frame.
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetWorkload (S_VOICE_MGR *pVoiceMgr, EAS_I32 maxWorkLoad);
+
+/*----------------------------------------------------------------------------
+ * VMCheckWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Checks to see if work load has been exceeded on this frame.
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMCheckWorkload (S_VOICE_MGR *pVoiceMgr);
+
+/*----------------------------------------------------------------------------
+ * VMActiveVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the number of active voices in the synthesizer.
+ *
+ * Inputs:
+ * pEASData - pointer to instance data
+ *
+ * Outputs:
+ * Returns the number of active voices
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 VMActiveVoices (S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMMIDIShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clean up any Synth related system issues.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMMIDIShutdown (S_EAS_DATA *pEASData, S_SYNTH *pSynth);
+
+/*----------------------------------------------------------------------------
+ * VMShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clean up any Synth related system issues.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMShutdown (S_EAS_DATA *pEASData);
+
+#ifdef EXTERNAL_AUDIO
+/*----------------------------------------------------------------------------
+ * EAS_RegExtAudioCallback()
+ *----------------------------------------------------------------------------
+ * Register a callback for external audio processing
+ *----------------------------------------------------------------------------
+*/
+void VMRegExtAudioCallback (S_SYNTH *pSynth, EAS_VOID_PTR pInstData, EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, EAS_EXT_EVENT_FUNC cbEventFunc);
+
+/*----------------------------------------------------------------------------
+ * VMGetMIDIControllers()
+ *----------------------------------------------------------------------------
+ * Returns the MIDI controller values on the specified channel
+ *----------------------------------------------------------------------------
+*/
+void VMGetMIDIControllers (S_SYNTH *pSynth, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl);
+#endif
+
+#ifdef _SPLIT_ARCHITECTURE
+/*----------------------------------------------------------------------------
+ * VMStartFrame()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Starts an audio frame
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMStartFrame (S_EAS_DATA *pEASData);
+
+/*----------------------------------------------------------------------------
+ * VMEndFrame()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Stops an audio frame
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMEndFrame (S_EAS_DATA *pEASData);
+#endif
+
+#endif /* #ifdef _EAS_VM_PROTOS_H */
+
diff --git a/arm-wt-22k/lib_src/eas_voicemgt.c b/arm-wt-22k/lib_src/eas_voicemgt.c new file mode 100644 index 0000000..873f29d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_voicemgt.c @@ -0,0 +1,3971 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_voicemgt.c
+ *
+ * Contents and purpose:
+ * Implements the synthesizer functions.
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 794 $
+ * $Date: 2007-08-01 00:08:48 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* includes */
+#include "eas.h"
+#include "eas_data.h"
+#include "eas_config.h"
+#include "eas_report.h"
+#include "eas_midictrl.h"
+#include "eas_host.h"
+#include "eas_synth_protos.h"
+#include "eas_vm_protos.h"
+
+#ifdef DLS_SYNTHESIZER
+#include "eas_mdls.h"
+#endif
+
+// #define _DEBUG_VM
+
+/* some defines for workload */
+#define WORKLOAD_AMOUNT_SMALL_INCREMENT 5
+#define WORKLOAD_AMOUNT_START_NOTE 10
+#define WORKLOAD_AMOUNT_STOP_NOTE 10
+#define WORKLOAD_AMOUNT_KEY_GROUP 10
+#define WORKLOAD_AMOUNT_POLY_LIMIT 10
+
+/* pointer to base sound library */
+extern S_EAS easSoundLib;
+
+#ifdef TEST_HARNESS
+extern S_EAS easTestLib;
+EAS_SNDLIB_HANDLE VMGetLibHandle(EAS_INT libNum)
+{
+ switch (libNum)
+ {
+ case 0:
+ return &easSoundLib;
+#ifdef _WT_SYNTH
+ case 1:
+ return &easTestLib;
+#endif
+ default:
+ return NULL;
+ }
+}
+#endif
+
+/* pointer to synthesizer interface(s) */
+#ifdef _WT_SYNTH
+extern const S_SYNTH_INTERFACE wtSynth;
+#endif
+
+#ifdef _FM_SYNTH
+extern const S_SYNTH_INTERFACE fmSynth;
+#endif
+
+typedef S_SYNTH_INTERFACE *S_SYNTH_INTERFACE_HANDLE;
+
+/* wavetable on MCU */
+#if defined(EAS_WT_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth;
+
+/* FM on MCU */
+#elif defined(EAS_FM_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &fmSynth;
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_HYBRID_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth;
+const S_SYNTH_INTERFACE *const pSecondarySynth = &fmSynth;
+
+/* wavetable drums on MCU, wavetable melodic on DSP */
+#elif defined(EAS_SPLIT_WT_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth;
+extern const S_FRAME_INTERFACE wtFrameInterface;
+const S_FRAME_INTERFACE *const pFrameInterface = &wtFrameInterface;
+
+/* wavetable drums on MCU, FM melodic on DSP */
+#elif defined(EAS_SPLIT_HYBRID_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &wtSynth;
+const S_SYNTH_INTERFACE *const pSecondarySynth = &fmSynth;
+extern const S_FRAME_INTERFACE fmFrameInterface;
+const S_FRAME_INTERFACE *const pFrameInterface = &fmFrameInterface;
+
+/* FM on DSP */
+#elif defined(EAS_SPLIT_FM_SYNTH)
+const S_SYNTH_INTERFACE *const pPrimarySynth = &fmSynth;
+extern const S_FRAME_INTERFACE fmFrameInterface;
+const S_FRAME_INTERFACE *const pFrameInterface = &fmFrameInterface;
+
+#else
+#error "Undefined architecture option"
+#endif
+
+/*----------------------------------------------------------------------------
+ * inline functions
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE const S_REGION* GetRegionPtr (S_SYNTH *pSynth, EAS_U16 regionIndex)
+{
+#if defined(DLS_SYNTHESIZER)
+ if (regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ return &pSynth->pDLS->pDLSRegions[regionIndex & REGION_INDEX_MASK].wtRegion.region;
+#endif
+#if defined(_HYBRID_SYNTH)
+ if (regionIndex & FLAG_RGN_IDX_FM_SYNTH)
+ return &pSynth->pEAS->pFMRegions[regionIndex & REGION_INDEX_MASK].region;
+ else
+ return &pSynth->pEAS->pWTRegions[regionIndex].region;
+#elif defined(_WT_SYNTH)
+ return &pSynth->pEAS->pWTRegions[regionIndex].region;
+#elif defined(_FM_SYNTH)
+ return &pSynth->pEAS->pFMRegions[regionIndex].region;
+#endif
+}
+
+/*lint -esym(715, voiceNum) used in some implementation */
+EAS_INLINE const S_SYNTH_INTERFACE* GetSynthPtr (EAS_INT voiceNum)
+{
+#if defined(_HYBRID_SYNTH)
+ if (voiceNum < NUM_PRIMARY_VOICES)
+ return pPrimarySynth;
+ else
+ return pSecondarySynth;
+#else
+ return pPrimarySynth;
+#endif
+}
+
+EAS_INLINE EAS_INT GetAdjustedVoiceNum (EAS_INT voiceNum)
+{
+#if defined(_HYBRID_SYNTH)
+ if (voiceNum >= NUM_PRIMARY_VOICES)
+ return voiceNum - NUM_PRIMARY_VOICES;
+#endif
+ return voiceNum;
+}
+
+EAS_INLINE EAS_U8 VSynthToChannel (S_SYNTH *pSynth, EAS_U8 channel)
+{
+ /*lint -e{734} synthNum is always 0-15 */
+ return channel | (pSynth->vSynthNum << 4);
+}
+
+/*----------------------------------------------------------------------------
+ * InitVoice()
+ *----------------------------------------------------------------------------
+ * Initialize a synthesizer voice
+ *----------------------------------------------------------------------------
+*/
+void InitVoice (S_SYNTH_VOICE *pVoice)
+{
+ pVoice->channel = UNASSIGNED_SYNTH_CHANNEL;
+ pVoice->nextChannel = UNASSIGNED_SYNTH_CHANNEL;
+ pVoice->note = pVoice->nextNote = DEFAULT_KEY_NUMBER;
+ pVoice->velocity = pVoice->nextVelocity = DEFAULT_VELOCITY;
+ pVoice->regionIndex = DEFAULT_REGION_INDEX;
+ pVoice->age = DEFAULT_AGE;
+ pVoice->voiceFlags = DEFAULT_VOICE_FLAGS;
+ pVoice->voiceState = DEFAULT_VOICE_STATE;
+}
+
+/*----------------------------------------------------------------------------
+ * IncVoicePoolCount()
+ *----------------------------------------------------------------------------
+ * Updates the voice pool count when a voice changes state
+ *----------------------------------------------------------------------------
+*/
+static void IncVoicePoolCount (S_VOICE_MGR *pVoiceMgr, S_SYNTH_VOICE *pVoice)
+{
+ S_SYNTH *pSynth;
+ EAS_INT pool;
+
+ /* ignore muting voices */
+ if (pVoice->voiceState == eVoiceStateMuting)
+ return;
+
+ if (pVoice->voiceState == eVoiceStateStolen)
+ {
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)];
+ pool = pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool;
+ }
+ else
+ {
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)];
+ pool = pSynth->channels[GET_CHANNEL(pVoice->channel)].pool;
+ }
+
+ pSynth->poolCount[pool]++;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "IncVoicePoolCount: Synth=%d pool=%d\n", pSynth->vSynthNum, pool); */ }
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * DecVoicePoolCount()
+ *----------------------------------------------------------------------------
+ * Updates the voice pool count when a voice changes state
+ *----------------------------------------------------------------------------
+*/
+static void DecVoicePoolCount (S_VOICE_MGR *pVoiceMgr, S_SYNTH_VOICE *pVoice)
+{
+ S_SYNTH *pSynth;
+ EAS_INT pool;
+
+ /* ignore muting voices */
+ if (pVoice->voiceState == eVoiceStateMuting)
+ return;
+
+ if (pVoice->voiceState == eVoiceStateStolen)
+ {
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)];
+ pool = pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool;
+ }
+ else
+ {
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)];
+ pool = pSynth->channels[GET_CHANNEL(pVoice->channel)].pool;
+ }
+
+ pSynth->poolCount[pool]--;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "DecVoicePoolCount: Synth=%d pool=%d\n", pSynth->vSynthNum, pool); */ }
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitialize()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMInitialize (S_EAS_DATA *pEASData)
+{
+ S_VOICE_MGR *pVoiceMgr;
+ EAS_INT i;
+
+ /* check Configuration Module for data allocation */
+ if (pEASData->staticMemoryModel)
+ pVoiceMgr = EAS_CMEnumData(EAS_CM_SYNTH_DATA);
+ else
+ pVoiceMgr = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_VOICE_MGR));
+ if (!pVoiceMgr)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitialize: Failed to allocate synthesizer memory\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+ EAS_HWMemSet(pVoiceMgr, 0, sizeof(S_VOICE_MGR));
+
+ /* initialize non-zero variables */
+ pVoiceMgr->pGlobalEAS = (S_EAS*) &easSoundLib;
+ pVoiceMgr->maxPolyphony = (EAS_U16) MAX_SYNTH_VOICES;
+
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+ pVoiceMgr->maxPolyphonyPrimary = NUM_PRIMARY_VOICES;
+ pVoiceMgr->maxPolyphonySecondary = NUM_SECONDARY_VOICES;
+#endif
+
+ /* set max workload to zero */
+ pVoiceMgr->maxWorkLoad = 0;
+
+ /* initialize the voice manager parameters */
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ InitVoice(&pVoiceMgr->voices[i]);
+
+ /* initialize the synth */
+ /*lint -e{522} return unused at this time */
+ pPrimarySynth->pfInitialize(pVoiceMgr);
+
+ /* initialize the off-chip synth */
+#ifdef _HYBRID_SYNTH
+ /*lint -e{522} return unused at this time */
+ pSecondarySynth->pfInitialize(pVoiceMgr);
+#endif
+
+ pEASData->pVoiceMgr = pVoiceMgr;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitMIDI()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMInitMIDI (S_EAS_DATA *pEASData, S_SYNTH **ppSynth)
+{
+ EAS_RESULT result;
+ S_SYNTH *pSynth;
+ EAS_INT virtualSynthNum;
+
+ *ppSynth = NULL;
+
+ /* static memory model only allows one synth */
+ if (pEASData->staticMemoryModel)
+ {
+ if (pEASData->pVoiceMgr->pSynth[0] != NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: No virtual synthesizer support for static memory model\n"); */ }
+ return EAS_ERROR_NO_VIRTUAL_SYNTHESIZER;
+ }
+
+ /* check Configuration Module for data allocation */
+ pSynth = EAS_CMEnumData(EAS_CM_MIDI_DATA);
+ virtualSynthNum = 0;
+ }
+
+ /* dynamic memory model */
+ else
+ {
+ for (virtualSynthNum = 0; virtualSynthNum < MAX_VIRTUAL_SYNTHESIZERS; virtualSynthNum++)
+ if (pEASData->pVoiceMgr->pSynth[virtualSynthNum] == NULL)
+ break;
+ if (virtualSynthNum == MAX_VIRTUAL_SYNTHESIZERS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: Exceeded number of active virtual synthesizers"); */ }
+ return EAS_ERROR_NO_VIRTUAL_SYNTHESIZER;
+ }
+ pSynth = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_SYNTH));
+ }
+
+ /* make sure we have a valid memory pointer */
+ if (pSynth == NULL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMInitMIDI: Failed to allocate synthesizer memory\n"); */ }
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+ EAS_HWMemSet(pSynth, 0, sizeof(S_SYNTH));
+
+ /* set the sound library pointer */
+ if ((result = VMSetEASLib(pSynth, pEASData->pVoiceMgr->pGlobalEAS)) != EAS_SUCCESS)
+ {
+ VMMIDIShutdown(pEASData, pSynth);
+ return result;
+ }
+
+ /* link in DLS bank if downloaded */
+#ifdef DLS_SYNTHESIZER
+ if (pEASData->pVoiceMgr->pGlobalDLS)
+ {
+ pSynth->pDLS = pEASData->pVoiceMgr->pGlobalDLS;
+ DLSAddRef(pSynth->pDLS);
+ }
+#endif
+
+ /* initialize MIDI state variables */
+ pSynth->synthFlags = DEFAULT_SYNTH_FLAGS;
+ pSynth->masterVolume = DEFAULT_SYNTH_MASTER_VOLUME;
+ pSynth->refCount = 1;
+ pSynth->priority = DEFAULT_SYNTH_PRIORITY;
+ pSynth->poolAlloc[0] = (EAS_U8) pEASData->pVoiceMgr->maxPolyphony;
+
+ VMInitializeAllChannels(pEASData->pVoiceMgr, pSynth);
+
+ pSynth->vSynthNum = (EAS_U8) virtualSynthNum;
+ pEASData->pVoiceMgr->pSynth[virtualSynthNum] = pSynth;
+
+ *ppSynth = pSynth;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMIncRefCount()
+ *----------------------------------------------------------------------------
+ * Increment reference count for virtual synth
+ *----------------------------------------------------------------------------
+*/
+void VMIncRefCount (S_SYNTH *pSynth)
+{
+ pSynth->refCount++;
+}
+
+/*----------------------------------------------------------------------------
+ * VMReset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this routine to start the process of reseting the synth.
+ * This routine sets a flag for the entire synth indicating that we want
+ * to reset.
+ * We also force all voices to mute quickly.
+ * However, we do not actually perform any synthesis in this routine. That
+ * is, we do not ramp the voices down from this routine, but instead, we
+ * let the "regular" synth processing steps take care of adding the ramp
+ * down samples to the output buffer. After we are sure that all voices
+ * have completed ramping down, we continue the process of resetting the
+ * synth (from another routine).
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ * force - force reset even if voices are active
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - set a flag (in psSynthObject->m_nFlags) indicating synth reset requested.
+ * - force all voices to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMReset (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_BOOL force)
+{
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReset: request to reset synth. Force = %d\n", force); */ }
+#endif
+
+ /* force voices to off state - may cause audio artifacts */
+ if (force)
+ {
+ pVoiceMgr->activeVoices -= pSynth->numActiveVoices;
+ pSynth->numActiveVoices = 0;
+ VMInitializeAllVoices(pVoiceMgr, pSynth->vSynthNum);
+ }
+ else
+ VMMuteAllVoices(pVoiceMgr, pSynth);
+
+ /* don't reset if voices are still playing */
+ if (pSynth->numActiveVoices == 0)
+ {
+ EAS_INT i;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReset: complete the reset process\n"); */ }
+#endif
+
+ VMInitializeAllChannels(pVoiceMgr, pSynth);
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ pSynth->poolCount[i] = 0;
+
+ /* set polyphony */
+ if (pSynth->maxPolyphony < pVoiceMgr->maxPolyphony)
+ pSynth->poolAlloc[0] = (EAS_U8) pVoiceMgr->maxPolyphony;
+ else
+ pSynth->poolAlloc[0] = (EAS_U8) pSynth->maxPolyphony;
+
+ /* clear reset flag */
+ pSynth->synthFlags &= ~SYNTH_FLAG_RESET_IS_REQUESTED;
+ }
+
+ /* handle reset after voices are muted */
+ else
+ pSynth->synthFlags |= SYNTH_FLAG_RESET_IS_REQUESTED;
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitializeAllChannels()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitializeAllChannels (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_INT i;
+
+ VMResetControllers(pSynth);
+
+ /* init each channel */
+ pChannel = pSynth->channels;
+
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++, pChannel++)
+ {
+ pChannel->channelFlags = DEFAULT_CHANNEL_FLAGS;
+ pChannel->staticGain = DEFAULT_CHANNEL_STATIC_GAIN;
+ pChannel->staticPitch = DEFAULT_CHANNEL_STATIC_PITCH;
+ pChannel->pool = 0;
+
+ /* the drum channel needs a different init */
+ if (i == DEFAULT_DRUM_CHANNEL)
+ {
+ pChannel->bankNum = DEFAULT_RHYTHM_BANK_NUMBER;
+ pChannel->channelFlags |= CHANNEL_FLAG_RHYTHM_CHANNEL;
+ }
+ else
+ pChannel->bankNum = DEFAULT_MELODY_BANK_NUMBER;
+
+ VMProgramChange(pVoiceMgr, pSynth, (EAS_U8) i, DEFAULT_SYNTH_PROGRAM_NUMBER);
+ }
+
+}
+
+/*----------------------------------------------------------------------------
+ * VMResetControllers()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMResetControllers (S_SYNTH *pSynth)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_INT i;
+
+ pChannel = pSynth->channels;
+
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++, pChannel++)
+ {
+ pChannel->pitchBend = DEFAULT_PITCH_BEND;
+ pChannel->modWheel = DEFAULT_MOD_WHEEL;
+ pChannel->volume = DEFAULT_CHANNEL_VOLUME;
+ pChannel->pan = DEFAULT_PAN;
+ pChannel->expression = DEFAULT_EXPRESSION;
+
+#ifdef _REVERB
+ pSynth->channels[i].reverbSend = DEFAULT_REVERB_SEND;
+#endif
+
+#ifdef _CHORUS
+ pSynth->channels[i].chorusSend = DEFAULT_CHORUS_SEND;
+#endif
+
+ pChannel->channelPressure = DEFAULT_CHANNEL_PRESSURE;
+ pChannel->registeredParam = DEFAULT_REGISTERED_PARAM;
+ pChannel->pitchBendSensitivity = DEFAULT_PITCH_BEND_SENSITIVITY;
+ pChannel->finePitch = DEFAULT_FINE_PITCH;
+ pChannel->coarsePitch = DEFAULT_COARSE_PITCH;
+
+ /* update all voices on this channel */
+ pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitializeAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitializeAllVoices (S_VOICE_MGR *pVoiceMgr, EAS_INT vSynthNum)
+{
+ EAS_INT i;
+
+ /* initialize the voice manager parameters */
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ if (pVoiceMgr->voices[i].voiceState != eVoiceStateStolen)
+ {
+ if (GET_VSYNTH(pVoiceMgr->voices[i].channel) == vSynthNum)
+ InitVoice(&pVoiceMgr->voices[i]);
+ }
+ else
+ {
+ if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == vSynthNum)
+ InitVoice(&pVoiceMgr->voices[i]);
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMMuteVoice()
+ *----------------------------------------------------------------------------
+ * Mute the selected voice
+ *----------------------------------------------------------------------------
+*/
+void VMMuteVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum)
+{
+ S_SYNTH *pSynth;
+ S_SYNTH_VOICE *pVoice;
+
+ /* take no action if voice is already muted */
+ pVoice = &pVoiceMgr->voices[voiceNum];
+ if ((pVoice->voiceState == eVoiceStateMuting) || (pVoice->voiceState == eVoiceStateFree))
+ return;
+
+ /* one less voice in pool */
+ DecVoicePoolCount(pVoiceMgr, pVoice);
+
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)];
+ GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pSynth, pVoice, GetAdjustedVoiceNum(voiceNum));
+ pVoice->voiceState = eVoiceStateMuting;
+
+}
+
+/*----------------------------------------------------------------------------
+ * VMReleaseVoice()
+ *----------------------------------------------------------------------------
+ * Release the selected voice
+ *----------------------------------------------------------------------------
+*/
+void VMReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 voiceNum)
+{
+ S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum];
+
+ /* take no action if voice is already free, muting, or releasing */
+ if (( pVoice->voiceState == eVoiceStateMuting) ||
+ (pVoice->voiceState == eVoiceStateFree) ||
+ (pVoice->voiceState == eVoiceStateRelease))
+ return;
+
+ /* stolen voices should just be muted */
+ if (pVoice->voiceState == eVoiceStateStolen)
+ VMMuteVoice(pVoiceMgr, voiceNum);
+
+ /* release this voice */
+ GetSynthPtr(voiceNum)->pfReleaseVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum));
+ pVoice->voiceState = eVoiceStateRelease;
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitMIPTable()
+ *----------------------------------------------------------------------------
+ * Initialize the SP-MIDI MIP table in preparation for receiving MIP message
+ *----------------------------------------------------------------------------
+*/
+void VMInitMIPTable (S_SYNTH *pSynth)
+{
+ EAS_INT i;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMInitMIPTable\n"); */ }
+#endif
+
+ /* clear SP-MIDI flag */
+ pSynth->synthFlags &= ~SYNTH_FLAG_SP_MIDI_ON;
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ {
+ pSynth->channels[i].pool = 0;
+ pSynth->channels[i].mip = 0;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetMIPEntry()
+ *----------------------------------------------------------------------------
+ * Sets the priority and MIP level for a MIDI channel
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+void VMSetMIPEntry (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 priority, EAS_U8 mip)
+{
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMSetMIPEntry: channel=%d, priority=%d, MIP=%d\n", channel, priority, mip); */ }
+#endif
+
+ /* save data for use by MIP message processing */
+ if (priority < NUM_SYNTH_CHANNELS)
+ {
+ pSynth->channels[channel].pool = priority;
+ pSynth->channels[channel].mip = mip;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMMIPUpdateChannelMuting()
+ *----------------------------------------------------------------------------
+ * This routine is called after an SP-MIDI message is received and
+ * any time the allocated polyphony changes. It mutes or unmutes
+ * channels based on polyphony.
+ *----------------------------------------------------------------------------
+*/
+void VMMIPUpdateChannelMuting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ EAS_INT i;
+ EAS_INT maxPolyphony;
+ EAS_INT channel;
+ EAS_INT vSynthNum;
+ EAS_INT pool;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateMIPTable\n"); */ }
+#endif
+
+ /* determine max polyphony */
+ if (pSynth->maxPolyphony)
+ maxPolyphony = pSynth->maxPolyphony;
+ else
+ maxPolyphony = pVoiceMgr->maxPolyphony;
+
+ /* process channels */
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ {
+
+ /* channel must be in MIP message and must meet allocation target */
+ if ((pSynth->channels[i].mip != 0) && (pSynth->channels[i].mip <= maxPolyphony))
+ pSynth->channels[i].channelFlags &= ~CHANNEL_FLAG_MUTE;
+ else
+ pSynth->channels[i].channelFlags |= CHANNEL_FLAG_MUTE;
+
+ /* reset voice pool count */
+ pSynth->poolCount[i] = 0;
+ }
+
+ /* mute any voices on muted channels, and count unmuted voices */
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+
+ /* ignore free voices */
+ if (pVoiceMgr->voices[i].voiceState == eVoiceStateFree)
+ continue;
+
+ /* get channel and virtual synth */
+ if (pVoiceMgr->voices[i].voiceState != eVoiceStateStolen)
+ {
+ vSynthNum = GET_VSYNTH(pVoiceMgr->voices[i].channel);
+ channel = GET_CHANNEL(pVoiceMgr->voices[i].channel);
+ }
+ else
+ {
+ vSynthNum = GET_VSYNTH(pVoiceMgr->voices[i].nextChannel);
+ channel = GET_CHANNEL(pVoiceMgr->voices[i].nextChannel);
+ }
+
+ /* ignore voices on other synths */
+ if (vSynthNum != pSynth->vSynthNum)
+ continue;
+
+ /* count voices */
+ pool = pSynth->channels[channel].pool;
+
+ /* deal with muted channels */
+ if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_MUTE)
+ {
+ /* mute stolen voices scheduled to play on this channel */
+ if (pVoiceMgr->voices[i].voiceState == eVoiceStateStolen)
+ pVoiceMgr->voices[i].voiceState = eVoiceStateMuting;
+
+ /* release voices that aren't already muting */
+ else if (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting)
+ {
+ VMReleaseVoice(pVoiceMgr, pSynth, i);
+ pSynth->poolCount[pool]++;
+ }
+ }
+
+ /* not muted, count this voice */
+ else
+ pSynth->poolCount[pool]++;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMUpdateMIPTable()
+ *----------------------------------------------------------------------------
+ * This routine is called at the end of the SysEx message to allow
+ * the Voice Manager to complete the initialization of the MIP
+ * table. It assigns channels to the appropriate voice pool based
+ * on the MIP setting and calculates the voices allocated for each
+ * pool.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+void VMUpdateMIPTable (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_INT i;
+ EAS_INT currentMIP;
+ EAS_INT currentPool;
+ EAS_INT priority[NUM_SYNTH_CHANNELS];
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateMIPTable\n"); */ }
+#endif
+
+ /* set SP-MIDI flag */
+ pSynth->synthFlags |= SYNTH_FLAG_SP_MIDI_ON;
+
+ /* sort channels into priority order */
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ priority[i] = -1;
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ {
+ if (pSynth->channels[i].pool != DEFAULT_SP_MIDI_PRIORITY)
+ priority[pSynth->channels[i].pool] = i;
+ }
+
+ /* process channels in priority order */
+ currentMIP = 0;
+ currentPool = -1;
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ {
+ /* stop when we run out of channels */
+ if (priority[i] == -1)
+ break;
+
+ pChannel = &pSynth->channels[priority[i]];
+
+ /* when 2 or more channels have the same MIP setting, they
+ * share a common voice pool
+ */
+ if (pChannel->mip == currentMIP)
+ pChannel->pool = (EAS_U8) currentPool;
+
+ /* new voice pool */
+ else
+ {
+ currentPool++;
+ pSynth->poolAlloc[currentPool] = (EAS_U8) (pChannel->mip - currentMIP);
+ currentMIP = pChannel->mip;
+ }
+ }
+
+ /* set SP-MIDI flag */
+ pSynth->synthFlags |= SYNTH_FLAG_SP_MIDI_ON;
+
+ /* update channel muting */
+ VMMIPUpdateChannelMuting (pVoiceMgr, pSynth);
+}
+
+/*----------------------------------------------------------------------------
+ * VMMuteAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this in an emergency reset situation.
+ * This forces all voices to mute quickly.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMMuteAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ EAS_INT i;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMMuteAllVoices: about to mute all voices!!\n"); */ }
+#endif
+
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ /* for stolen voices, check new channel */
+ if (pVoiceMgr->voices[i].voiceState == eVoiceStateStolen)
+ {
+ if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == pSynth->vSynthNum)
+ VMMuteVoice(pVoiceMgr, i);
+ }
+
+ else if (pSynth->vSynthNum == GET_VSYNTH(pVoiceMgr->voices[i].channel))
+ VMMuteVoice(pVoiceMgr, i);
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMReleaseAllVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We call this after we've encountered the end of the Midi file.
+ * This ensures all voices are either in release (because we received their
+ * note off already) or forces them to mute quickly.
+ * We use this as a safety to prevent bad midi files from playing forever.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices to update their envelope states to release or mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMReleaseAllVoices (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ EAS_INT i;
+
+ /* release sustain pedal on all channels */
+ for (i = 0; i < NUM_SYNTH_CHANNELS; i++)
+ {
+ if (pSynth->channels[ i ].channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL)
+ {
+ VMReleaseAllDeferredNoteOffs(pVoiceMgr, pSynth, (EAS_U8) i);
+ pSynth->channels[i].channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL;
+ }
+ }
+
+ /* release all voices */
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+
+ switch (pVoiceMgr->voices[i].voiceState)
+ {
+ case eVoiceStateStart:
+ case eVoiceStatePlay:
+ /* only release voices on this synth */
+ if (GET_VSYNTH(pVoiceMgr->voices[i].channel) == pSynth->vSynthNum)
+ VMReleaseVoice(pVoiceMgr, pSynth, i);
+ break;
+
+ case eVoiceStateStolen:
+ if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) == pSynth->vSynthNum)
+ VMMuteVoice(pVoiceMgr, i);
+ break;
+
+ case eVoiceStateFree:
+ case eVoiceStateRelease:
+ case eVoiceStateMuting:
+ break;
+
+ case eVoiceStateInvalid:
+ default:
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReleaseAllVoices: error, %d is an unrecognized state\n",
+ pVoiceMgr->voices[i].voiceState); */ }
+#endif
+ break;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMAllNotesOff()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Quickly mute all notes on the given channel.
+ *
+ * Inputs:
+ * nChannel - quickly turn off all notes on this channel
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - forces all voices on this channel to update their envelope states to mute
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMAllNotesOff (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
+{
+ EAS_INT voiceNum;
+ S_SYNTH_VOICE *pVoice;
+
+#ifdef _DEBUG_VM
+ if (channel >= NUM_SYNTH_CHANNELS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMAllNotesOff: error, %d invalid channel number\n",
+ channel); */ }
+ return;
+ }
+#endif
+
+ /* increment workload */
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT;
+
+ /* check each voice */
+ channel = VSynthToChannel(pSynth, channel);
+ for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+ pVoice = &pVoiceMgr->voices[voiceNum];
+ if (pVoice->voiceState != eVoiceStateFree)
+ {
+ if (((pVoice->voiceState != eVoiceStateStolen) && (channel == pVoice->channel)) ||
+ ((pVoice->voiceState == eVoiceStateStolen) && (channel == pVoice->nextChannel)))
+ {
+ /* this voice is assigned to the requested channel */
+ GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pSynth, pVoice, GetAdjustedVoiceNum(voiceNum));
+ pVoice->voiceState = eVoiceStateMuting;
+ }
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMDeferredStopNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Stop the notes that had deferred note-off requests.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None.
+ *
+ * Side Effects:
+ * voices that have had deferred note-off requests are now put into release
+ * psSynthObject->m_sVoice[i].m_nFlags has the VOICE_FLAG_DEFER_MIDI_NOTE_OFF
+ * cleared
+ *----------------------------------------------------------------------------
+*/
+void VMDeferredStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ EAS_INT voiceNum;
+ EAS_INT channel;
+ EAS_BOOL deferredNoteOff;
+
+ deferredNoteOff = EAS_FALSE;
+
+ /* check each voice to see if it requires a deferred note off */
+ for (voiceNum=0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+ if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_DEFER_MIDI_NOTE_OFF)
+ {
+ /* check if this voice was stolen */
+ if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen)
+ {
+ /*
+ This voice was stolen, AND it also has a deferred note-off.
+ The stolen note must be completely ramped down at this point.
+ The note that caused the stealing to occur, however, must
+ have received a note-off request before the note that caused
+ stealing ever had a chance to even start. We want to give
+ the note that caused the stealing a chance to play, so we
+ start it on the next update interval, and we defer sending
+ the note-off request until the subsequent update interval.
+ So do not send the note-off request for this voice because
+ this voice was stolen and should have completed ramping down,
+ Also, do not clear the global flag nor this voice's flag
+ because we must indicate that the subsequent update interval,
+ after the note that caused stealing has started, should
+ then send the deferred note-off request.
+ */
+ deferredNoteOff = EAS_TRUE;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: defer request to stop voice %d (channel=%d note=%d) - voice not started\n",
+ voiceNum,
+ pVoiceMgr->voices[voiceNum].nextChannel,
+ pVoiceMgr->voices[voiceNum].note); */ }
+
+ /* sanity check: this stolen voice better be ramped to zero */
+ if (0 != pVoiceMgr->voices[voiceNum].gain)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: warning, this voice did not complete its ramp to zero\n"); */ }
+ }
+#endif // #ifdef _DEBUG_VM
+
+ }
+ else
+ {
+ /* clear the flag using exor */
+ pVoiceMgr->voices[voiceNum].voiceFlags ^=
+ VOICE_FLAG_DEFER_MIDI_NOTE_OFF;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMDeferredStopNote: Stop voice %d (channel=%d note=%d)\n",
+ voiceNum,
+ pVoiceMgr->voices[voiceNum].nextChannel,
+ pVoiceMgr->voices[voiceNum].note); */ }
+#endif
+
+ channel = pVoiceMgr->voices[voiceNum].channel & 15;
+
+ /* check if sustain pedal is on */
+ if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL)
+ {
+ GetSynthPtr(voiceNum)->pfSustainPedal(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], &pSynth->channels[channel], GetAdjustedVoiceNum(voiceNum));
+ }
+
+ /* release this voice */
+ else
+ VMReleaseVoice(pVoiceMgr, pSynth, voiceNum);
+
+ }
+
+ }
+
+ }
+
+ /* clear the deferred note-off flag, unless there's another one pending */
+ if (deferredNoteOff == EAS_FALSE)
+ pSynth->synthFlags ^= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING;
+}
+
+/*----------------------------------------------------------------------------
+ * VMReleaseAllDeferredNoteOffs()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this functin when the sustain flag is presently set but
+ * we are now transitioning from damper pedal on to
+ * damper pedal off. This means all notes in this channel
+ * that received a note off while the damper pedal was on, and
+ * had their note-off requests deferred, should now proceed to
+ * the release state.
+ *
+ * Inputs:
+ * nChannel - this channel has its sustain pedal transitioning from on to off
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * any voice with deferred note offs on this channel are updated such that
+ * pVoice->m_sEG1.m_eState = eEnvelopeStateRelease
+ * pVoice->m_sEG1.m_nIncrement = release increment
+ * pVoice->m_nFlags = clear the deferred note off flag
+ *----------------------------------------------------------------------------
+*/
+void VMReleaseAllDeferredNoteOffs (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
+{
+ S_SYNTH_VOICE *pVoice;
+ EAS_INT voiceNum;
+
+#ifdef _DEBUG_VM
+ if (channel >= NUM_SYNTH_CHANNELS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMReleaseAllDeferredNoteOffs: error, %d invalid channel number\n",
+ channel); */ }
+ return;
+ }
+#endif /* #ifdef _DEBUG_VM */
+
+ /* increment workload */
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT;
+
+ /* find all the voices assigned to this channel */
+ channel = VSynthToChannel(pSynth, channel);
+ for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+
+ pVoice = &pVoiceMgr->voices[voiceNum];
+ if (channel == pVoice->channel)
+ {
+
+ /* does this voice have a deferred note off? */
+ if (pVoice->voiceFlags & VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF)
+ {
+ /* release voice */
+ VMReleaseVoice(pVoiceMgr, pSynth, voiceNum);
+
+ /* use exor to flip bit, clear the flag */
+ pVoice->voiceFlags &= ~VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF;
+
+ }
+
+ }
+ }
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * VMCatchNotesForSustainPedal()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this function when the sustain flag is presently clear and
+ * the damper pedal is off and we are transitioning from damper pedal OFF to
+ * damper pedal ON. Currently sounding notes should be left
+ * unchanged. However, we should try to "catch" notes if possible.
+ * If any notes are in release and have levels >= sustain level, catch them,
+ * otherwise, let them continue to release.
+ *
+ * Inputs:
+ * nChannel - this channel has its sustain pedal transitioning from on to off
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *----------------------------------------------------------------------------
+*/
+void VMCatchNotesForSustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
+{
+ EAS_INT voiceNum;
+
+#ifdef _DEBUG_VM
+ if (channel >= NUM_SYNTH_CHANNELS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCatchNotesForSustainPedal: error, %d invalid channel number\n",
+ channel); */ }
+ return;
+ }
+#endif
+
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_SMALL_INCREMENT;
+ channel = VSynthToChannel(pSynth, channel);
+
+ /* find all the voices assigned to this channel */
+ for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+ if (channel == pVoiceMgr->voices[voiceNum].channel)
+ {
+ if (eVoiceStateRelease == pVoiceMgr->voices[voiceNum].voiceState)
+ GetSynthPtr(voiceNum)->pfSustainPedal(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], &pSynth->channels[channel], GetAdjustedVoiceNum(voiceNum));
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMUpdateAllNotesAge()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Increment the note age for all of the active voices.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * m_nAge for all voices is incremented
+ *----------------------------------------------------------------------------
+*/
+void VMUpdateAllNotesAge (S_VOICE_MGR *pVoiceMgr, EAS_U16 age)
+{
+ EAS_INT i;
+
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ if (age - pVoiceMgr->voices[i].age > 0)
+ pVoiceMgr->voices[i].age++;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMStolenVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice is being stolen. Sets the parameters so that the
+ * voice will begin playing the new sound on the next buffer.
+ *
+ * Inputs:
+ * pVoice - pointer to voice to steal
+ * nChannel - the channel to start a note on
+ * nKeyNumber - the key number to start a note for
+ * nNoteVelocity - the key velocity from this note
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+static void VMStolenVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 voiceNum, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex)
+{
+ S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum];
+
+ /* one less voice in old pool */
+ DecVoicePoolCount(pVoiceMgr, pVoice);
+
+ /* mute the sound that is currently playing */
+ GetSynthPtr(voiceNum)->pfMuteVoice(pVoiceMgr, pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)], &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum));
+ pVoice->voiceState = eVoiceStateStolen;
+
+ /* set new note data */
+ pVoice->nextChannel = VSynthToChannel(pSynth, channel);
+ pVoice->nextNote = note;
+ pVoice->nextVelocity = velocity;
+ pVoice->nextRegionIndex = regionIndex;
+
+ /* one more voice in new pool */
+ IncVoicePoolCount(pVoiceMgr, pVoice);
+
+ /* clear the deferred flags */
+ pVoice->voiceFlags &=
+ ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
+ VOICE_FLAG_DEFER_MUTE |
+ VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF);
+
+ /* all notes older than this one get "younger" */
+ VMUpdateAllNotesAge(pVoiceMgr, pVoice->age);
+
+ /* assign current age to this note and increment for the next note */
+ pVoice->age = pVoiceMgr->age++;
+}
+
+/*----------------------------------------------------------------------------
+ * VMFreeVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice is done playing and being returned to the
+ * pool of free voices
+ *
+ * Inputs:
+ * pVoice - pointer to voice to free
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+static void VMFreeVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice)
+{
+
+ /* do nothing if voice is already free */
+ if (pVoice->voiceState == eVoiceStateFree)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMFreeVoice: Attempt to free voice that is already free\n"); */ }
+ return;
+ }
+
+ /* if we jump directly to free without passing muting stage,
+ * we need to adjust the voice count */
+ DecVoicePoolCount(pVoiceMgr, pVoice);
+
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMFreeVoice: Synth=%d\n", pSynth->vSynthNum); */ }
+#endif
+
+ /* return to free voice pool */
+ pVoiceMgr->activeVoices--;
+ pSynth->numActiveVoices--;
+ InitVoice(pVoice);
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMFreeVoice: free voice %d\n", pVoice - pVoiceMgr->voices); */ }
+#endif
+
+ /* all notes older than this one get "younger" */
+ VMUpdateAllNotesAge(pVoiceMgr, pVoice->age);
+ }
+
+/*----------------------------------------------------------------------------
+ * VMRetargetStolenVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice has been stolen and needs to be initalized with
+ * the paramters of its new note.
+ *
+ * Inputs:
+ * pVoice - pointer to voice to retarget
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL VMRetargetStolenVoice (S_VOICE_MGR *pVoiceMgr, EAS_I32 voiceNum)
+{
+ EAS_U8 flags;
+ S_SYNTH_CHANNEL *pMIDIChannel;
+ S_SYNTH_VOICE *pVoice;
+ S_SYNTH *pSynth;
+ S_SYNTH *pNextSynth;
+
+ /* establish some pointers */
+ pVoice = &pVoiceMgr->voices[voiceNum];
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)];
+ pMIDIChannel = &pSynth->channels[pVoice->channel & 15];
+ pNextSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->nextChannel)];
+
+#ifdef _DEBUG_VM
+{ /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetStolenVoice: retargeting stolen voice %d on channel %d\n",
+ voiceNum, pVoice->channel); */ }
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\to channel %d note: %d velocity: %d\n",
+ pVoice->nextChannel, pVoice->nextNote, pVoice->nextVelocity); */ }
+#endif
+
+ /* make sure new channel hasn't been muted by SP-MIDI since the voice was stolen */
+ if ((pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON) &&
+ (pMIDIChannel->channelFlags & CHANNEL_FLAG_MUTE))
+ {
+ VMFreeVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum]);
+ return EAS_FALSE;
+ }
+
+ /* if assigned to a new synth, correct the active voice count */
+ if (pVoice->channel != pVoice->nextChannel)
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetStolenVoice: Note assigned to different virtual synth, adjusting numActiveVoices\n"); */ }
+#endif
+ pSynth->numActiveVoices--;
+ pNextSynth->numActiveVoices++;
+ }
+
+ /* assign new channel number, and increase channel voice count */
+ pVoice->channel = pVoice->nextChannel;
+ pMIDIChannel = &pNextSynth->channels[pVoice->channel & 15];
+
+ /* assign other data */
+ pVoice->note = pVoice->nextNote;
+ pVoice->velocity = pVoice->nextVelocity;
+ pVoice->nextChannel = UNASSIGNED_SYNTH_CHANNEL;
+ pVoice->regionIndex = pVoice->nextRegionIndex;
+
+ /* save the flags, pfStartVoice() will clear them */
+ flags = pVoice->voiceFlags;
+
+ /* keep track of the note-start related workload */
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_START_NOTE;
+
+ /* setup the voice parameters */
+ pVoice->voiceState = eVoiceStateStart;
+
+ /*lint -e{522} return not used at this time */
+ GetSynthPtr(voiceNum)->pfStartVoice(pVoiceMgr, pNextSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), pVoice->regionIndex);
+
+ /* did the new note already receive a MIDI note-off request? */
+ if (flags & VOICE_FLAG_DEFER_MIDI_NOTE_OFF)
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMRetargetVoice: stolen note note-off request deferred\n"); */ }
+#endif
+ pVoice->voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF;
+ pNextSynth->synthFlags |= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING;
+ }
+
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * VMCheckKeyGroup()
+ *----------------------------------------------------------------------------
+ * If the note that we've been asked to start is in the same key group as
+ * any currently playing notes, then we must shut down the currently playing
+ * note in the same key group
+ *----------------------------------------------------------------------------
+*/
+void VMCheckKeyGroup (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U16 keyGroup, EAS_U8 channel)
+{
+ const S_REGION *pRegion;
+ EAS_INT voiceNum;
+
+ /* increment frame workload */
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_KEY_GROUP;
+
+ /* need to check all voices in case this is a layered sound */
+ channel = VSynthToChannel(pSynth, channel);
+ for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+ if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateStolen)
+ {
+ /* voice must be on the same channel */
+ if (channel == pVoiceMgr->voices[voiceNum].channel)
+ {
+ /* check key group */
+ pRegion = GetRegionPtr(pSynth, pVoiceMgr->voices[voiceNum].regionIndex);
+ if (keyGroup == (pRegion->keyGroupAndFlags & 0x0f00))
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckKeyGroup: voice %d matches key group %d\n", voiceNum, keyGroup >> 8); */ }
+#endif
+
+ /* if this voice was just started, set it to mute on the next buffer */
+ if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)
+ pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MUTE;
+
+ /* mute immediately */
+ else
+ VMMuteVoice(pVoiceMgr, voiceNum);
+ }
+ }
+ }
+
+ /* for stolen voice, check new values */
+ else
+ {
+ /* voice must be on the same channel */
+ if (channel == pVoiceMgr->voices[voiceNum].nextChannel)
+ {
+ /* check key group */
+ pRegion = GetRegionPtr(pSynth, pVoiceMgr->voices[voiceNum].nextRegionIndex);
+ if (keyGroup == (pRegion->keyGroupAndFlags & 0x0f00))
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckKeyGroup: voice %d matches key group %d\n", voiceNum, keyGroup >> 8); */ }
+#endif
+
+ /* if this voice was just started, set it to mute on the next buffer */
+ if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)
+ pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MUTE;
+
+ /* mute immediately */
+ else
+ VMMuteVoice(pVoiceMgr, voiceNum);
+ }
+ }
+
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMCheckPolyphonyLimiting()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * We only play at most 2 of the same note on a MIDI channel.
+ * E.g., if we are asked to start note 36, and there are already two voices
+ * that are playing note 36, then we must steal the voice playing
+ * the oldest note 36 and use that stolen voice to play the new note 36.
+ *
+ * Inputs:
+ * nChannel - synth channel that wants to start a new note
+ * nKeyNumber - new note's midi note number
+ * nNoteVelocity - new note's velocity
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pbVoiceStealingRequired - flag: this routine sets true if we needed to
+ * steal a voice
+ * *
+ * Side Effects:
+ * psSynthObject->m_sVoice[free voice num].m_nKeyNumber may be assigned
+ * psSynthObject->m_sVoice[free voice num].m_nVelocity may be assigned
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMCheckPolyphonyLimiting (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex, EAS_I32 lowVoice, EAS_I32 highVoice)
+{
+ EAS_INT voiceNum;
+ EAS_INT oldestVoiceNum;
+ EAS_INT numVoicesPlayingNote;
+ EAS_U16 age;
+ EAS_U16 oldestNoteAge;
+
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_POLY_LIMIT;
+
+ numVoicesPlayingNote = 0;
+ oldestVoiceNum = MAX_SYNTH_VOICES;
+ oldestNoteAge = 0;
+ channel = VSynthToChannel(pSynth, channel);
+
+ /* examine each voice on this channel playing this note */
+ for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++)
+ {
+ /* check stolen notes separately */
+ if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateStolen)
+ {
+
+ /* same channel and note ? */
+ if ((channel == pVoiceMgr->voices[voiceNum].channel) && (note == pVoiceMgr->voices[voiceNum].note))
+ {
+ numVoicesPlayingNote++;
+ age = pVoiceMgr->age - pVoiceMgr->voices[voiceNum].age;
+
+ /* is this the oldest voice for this note? */
+ if (age >= oldestNoteAge)
+ {
+ oldestNoteAge = age;
+ oldestVoiceNum = voiceNum;
+ }
+ }
+ }
+
+ /* handle stolen voices */
+ else
+ {
+ /* same channel and note ? */
+ if ((channel == pVoiceMgr->voices[voiceNum].nextChannel) && (note == pVoiceMgr->voices[voiceNum].nextNote))
+ {
+ numVoicesPlayingNote++;
+ }
+ }
+ }
+
+ /* check to see if we exceeded poly limit */
+ if (numVoicesPlayingNote < DEFAULT_CHANNEL_POLYPHONY_LIMIT)
+ return EAS_FALSE;
+
+ /* make sure we have a voice to steal */
+ if (oldestVoiceNum != MAX_SYNTH_VOICES)
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMCheckPolyphonyLimiting: voice %d has the oldest note\n", oldestVoiceNum); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMCheckPolyphonyLimiting: polyphony limiting requires shutting down note %d \n", pVoiceMgr->voices[oldestVoiceNum].note); */ }
+#endif
+ VMStolenVoice(pVoiceMgr, pSynth, oldestVoiceNum, channel, note, velocity, regionIndex);
+ return EAS_TRUE;
+ }
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMCheckPolyphonyLimiting: No oldest voice to steal\n"); */ }
+#endif
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * VMStartVoice()
+ *----------------------------------------------------------------------------
+ * Starts a voice given a region index
+ *----------------------------------------------------------------------------
+*/
+void VMStartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity, EAS_U16 regionIndex)
+{
+ const S_REGION *pRegion;
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_INT voiceNum;
+ EAS_INT maxSynthPoly;
+ EAS_I32 lowVoice, highVoice;
+ EAS_U16 keyGroup;
+
+ pChannel = &pSynth->channels[channel];
+ pRegion = GetRegionPtr(pSynth, regionIndex);
+
+ /* select correct synth */
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+ {
+#ifdef EAS_SPLIT_WT_SYNTH
+ if ((pRegion->keyGroupAndFlags & REGION_FLAG_OFF_CHIP) == 0)
+#else
+ if ((regionIndex & FLAG_RGN_IDX_FM_SYNTH) == 0)
+#endif
+ {
+ lowVoice = 0;
+ highVoice = NUM_PRIMARY_VOICES - 1;
+ }
+ else
+ {
+ lowVoice = NUM_PRIMARY_VOICES;
+ highVoice = MAX_SYNTH_VOICES - 1;
+ }
+ }
+#else
+ lowVoice = 0;
+ highVoice = MAX_SYNTH_VOICES - 1;
+#endif
+
+ /* keep track of the note-start related workload */
+ pVoiceMgr->workload+= WORKLOAD_AMOUNT_START_NOTE;
+
+ /* other voices in pool, check for key group and poly limiting */
+ if (pSynth->poolCount[pChannel->pool] != 0)
+ {
+
+ /* check for key group exclusivity */
+ keyGroup = pRegion->keyGroupAndFlags & 0x0f00;
+ if (keyGroup!= 0)
+ VMCheckKeyGroup(pVoiceMgr, pSynth, keyGroup, channel);
+
+ /* check polyphony limit and steal a voice if necessary */
+ if ((pRegion->keyGroupAndFlags & REGION_FLAG_NON_SELF_EXCLUSIVE) == 0)
+ {
+ if (VMCheckPolyphonyLimiting(pVoiceMgr, pSynth, channel, note, velocity, regionIndex, lowVoice, highVoice) == EAS_TRUE)
+ return;
+ }
+ }
+
+ /* check max poly allocation */
+ if ((pSynth->maxPolyphony == 0) || (pVoiceMgr->maxPolyphony < pSynth->maxPolyphony))
+ maxSynthPoly = pVoiceMgr->maxPolyphony;
+ else
+ maxSynthPoly = pSynth->maxPolyphony;
+
+ /* any free voices? */
+ if ((pVoiceMgr->activeVoices < pVoiceMgr->maxPolyphony) &&
+ (pSynth->numActiveVoices < maxSynthPoly) &&
+ (EAS_SUCCESS == VMFindAvailableVoice(pVoiceMgr, &voiceNum, lowVoice, highVoice)))
+ {
+ S_SYNTH_VOICE *pVoice = &pVoiceMgr->voices[voiceNum];
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMStartVoice: Synth=%d\n", pSynth->vSynthNum); */ }
+#endif
+
+ /* bump voice counts */
+ pVoiceMgr->activeVoices++;
+ pSynth->numActiveVoices++;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStartVoice: voice %d assigned to channel %d note %d velocity %d\n",
+ voiceNum, channel, note, velocity); */ }
+#endif
+
+ /* save parameters */
+ pVoiceMgr->voices[voiceNum].channel = VSynthToChannel(pSynth, channel);
+ pVoiceMgr->voices[voiceNum].note = note;
+ pVoiceMgr->voices[voiceNum].velocity = velocity;
+
+ /* establish note age for voice stealing */
+ pVoiceMgr->voices[voiceNum].age = pVoiceMgr->age++;
+
+ /* setup the synthesis parameters */
+ pVoiceMgr->voices[voiceNum].voiceState = eVoiceStateStart;
+
+ /* increment voice pool count */
+ IncVoicePoolCount(pVoiceMgr, pVoice);
+
+ /* start voice on correct synth */
+ /*lint -e{522} return not used at this time */
+ GetSynthPtr(voiceNum)->pfStartVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), regionIndex);
+ return;
+ }
+
+ /* no free voices, we have to steal one using appropriate algorithm */
+ if (VMStealVoice(pVoiceMgr, pSynth, &voiceNum, channel, note, lowVoice, highVoice) == EAS_SUCCESS)
+ VMStolenVoice(pVoiceMgr, pSynth, voiceNum, channel, note, velocity, regionIndex);
+
+#ifdef _DEBUG_VM
+ else
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStartVoice: Could not steal a voice for channel %d note %d velocity %d\n",
+ channel, note, velocity); */ }
+ }
+#endif
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * VMStartNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the synth's state to play the requested note on the requested
+ * channel if possible.
+ *
+ * Inputs:
+ * nChannel - the channel to start a note on
+ * nKeyNumber - the key number to start a note for
+ * nNoteVelocity - the key velocity from this note
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * psSynthObject->m_nNumActiveVoices may be incremented
+ * psSynthObject->m_sVoice[free voice num].m_nSynthChannel may be assigned
+ * psSynthObject->m_sVoice[free voice num].m_nKeyNumber is assigned
+ * psSynthObject->m_sVoice[free voice num].m_nVelocity is assigned
+ *----------------------------------------------------------------------------
+*/
+void VMStartNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_U16 regionIndex;
+ EAS_I16 adjustedNote;
+
+ /* bump note count */
+ pSynth->totalNoteCount++;
+
+ pChannel = &pSynth->channels[channel];
+
+ /* check channel mute */
+ if (pChannel->channelFlags & CHANNEL_FLAG_MUTE)
+ return;
+
+#ifdef EXTERNAL_AUDIO
+ /* pass event to external audio when requested */
+ if ((pChannel->channelFlags & CHANNEL_FLAG_EXTERNAL_AUDIO) && (pSynth->cbEventFunc != NULL))
+ {
+ S_EXT_AUDIO_EVENT event;
+ event.channel = channel;
+ event.note = note;
+ event.velocity = velocity;
+ event.noteOn = EAS_TRUE;
+ if (pSynth->cbEventFunc(pSynth->pExtAudioInstData, &event))
+ return;
+ }
+#endif
+
+ /* start search at first region */
+ regionIndex = pChannel->regionIndex;
+
+ /* handle transposition */
+ adjustedNote = note;
+ if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)
+ adjustedNote += pChannel->coarsePitch;
+ else
+ adjustedNote += pChannel->coarsePitch + pSynth->globalTranspose;
+
+ /* limit adjusted key number so it does not wraparound, over/underflow */
+ if (adjustedNote < 0)
+ {
+ adjustedNote = 0;
+ }
+ else if (adjustedNote > 127)
+ {
+ adjustedNote = 127;
+ }
+
+#if defined(DLS_SYNTHESIZER)
+ if (regionIndex & FLAG_RGN_IDX_DLS_SYNTH)
+ {
+ /* DLS voice */
+ for (;;)
+ {
+ /*lint -e{740,826} cast OK, we know this is actually a DLS region */
+ const S_DLS_REGION *pDLSRegion = (S_DLS_REGION*) GetRegionPtr(pSynth, regionIndex);
+
+ /* check key against this region's key and velocity range */
+ if (((adjustedNote >= pDLSRegion->wtRegion.region.rangeLow) && (adjustedNote <= pDLSRegion->wtRegion.region.rangeHigh)) &&
+ ((velocity >= pDLSRegion->velLow) && (velocity <= pDLSRegion->velHigh)))
+ {
+ VMStartVoice(pVoiceMgr, pSynth, channel, note, velocity, regionIndex);
+ }
+
+ /* last region in program? */
+ if (pDLSRegion->wtRegion.region.keyGroupAndFlags & REGION_FLAG_LAST_REGION)
+ break;
+
+ /* advance to next region */
+ regionIndex++;
+ }
+ }
+ else
+#endif
+
+ /* braces here for #if clause */
+ {
+ /* EAS voice */
+ for (;;)
+ {
+ const S_REGION *pRegion = GetRegionPtr(pSynth, regionIndex);
+
+ /* check key against this region's keyrange */
+ if ((adjustedNote >= pRegion->rangeLow) && (adjustedNote <= pRegion->rangeHigh))
+ {
+ VMStartVoice(pVoiceMgr, pSynth, channel, note, velocity, regionIndex);
+ break;
+ }
+
+ /* last region in program? */
+ if (pRegion->keyGroupAndFlags & REGION_FLAG_LAST_REGION)
+ break;
+
+ /* advance to next region */
+ regionIndex++;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMStopNote()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update the synth's state to end the requested note on the requested
+ * channel.
+ *
+ * Inputs:
+ * nChannel - the channel to stop a note on
+ * nKeyNumber - the key number for this note off
+ * nNoteVelocity - the note-off velocity
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * psSynthObject->m_sVoice[free voice num].m_nSynthChannel may be assigned
+ * psSynthObject->m_sVoice[free voice num].m_nKeyNumber is assigned
+ * psSynthObject->m_sVoice[free voice num].m_nVelocity is assigned
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, velocity) reserved for future use */
+void VMStopNote (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 note, EAS_U8 velocity)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_INT voiceNum;
+
+ pChannel = &(pSynth->channels[channel]);
+
+#ifdef EXTERNAL_AUDIO
+ if ((pChannel->channelFlags & CHANNEL_FLAG_EXTERNAL_AUDIO) && (pSynth->cbEventFunc != NULL))
+ {
+ S_EXT_AUDIO_EVENT event;
+ event.channel = channel;
+ event.note = note;
+ event.velocity = velocity;
+ event.noteOn = EAS_FALSE;
+ if (pSynth->cbEventFunc(pSynth->pExtAudioInstData, &event))
+ return;
+ }
+#endif
+
+ /* keep track of the note-start workload */
+ pVoiceMgr->workload += WORKLOAD_AMOUNT_STOP_NOTE;
+
+ channel = VSynthToChannel(pSynth, channel);
+
+ for (voiceNum=0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+
+ /* stolen notes are handled separately */
+ if (eVoiceStateStolen != pVoiceMgr->voices[voiceNum].voiceState)
+ {
+
+ /* channel and key number must match */
+ if ((channel == pVoiceMgr->voices[voiceNum].channel) && (note == pVoiceMgr->voices[voiceNum].note))
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStopNote: voice %d channel %d note %d\n",
+ voiceNum, channel, note); */ }
+#endif
+
+ /* if sustain pedal is down, set deferred note-off flag */
+ if (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL)
+ {
+ pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF;
+ continue;
+ }
+
+ /* if this note just started, wait before we stop it */
+ if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)
+ {
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "\tDeferred: Not started yet\n"); */ }
+#endif
+ pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF;
+ pSynth->synthFlags |= SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING;
+ }
+
+ /* release voice */
+ else
+ VMReleaseVoice(pVoiceMgr, pSynth, voiceNum);
+
+ }
+ }
+
+ /* process stolen notes, new channel and key number must match */
+ else if ((channel == pVoiceMgr->voices[voiceNum].nextChannel) && (note == pVoiceMgr->voices[voiceNum].nextNote))
+ {
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStopNote: voice %d channel %d note %d\n\tDeferred: Stolen voice\n",
+ voiceNum, channel, note); */ }
+#endif
+ pVoiceMgr->voices[voiceNum].voiceFlags |= VOICE_FLAG_DEFER_MIDI_NOTE_OFF;
+ }
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * VMFindAvailableVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Find an available voice and return the voice number if available.
+ *
+ * Inputs:
+ * pnVoiceNumber - really an output, returns the voice number found
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * success - if there is an available voice
+ * failure - otherwise
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMFindAvailableVoice (S_VOICE_MGR *pVoiceMgr, EAS_INT *pVoiceNumber, EAS_I32 lowVoice, EAS_I32 highVoice)
+{
+ EAS_INT voiceNum;
+
+ /* Check each voice to see if it has been assigned to a synth channel */
+ for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++)
+ {
+ /* check if this voice has been assigned to a synth channel */
+ if ( pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateFree)
+ {
+ *pVoiceNumber = voiceNum; /* this voice is available */
+ return EAS_SUCCESS;
+ }
+ }
+
+ /* if we reach here, we have not found a free voice */
+ *pVoiceNumber = UNASSIGNED_SYNTH_VOICE;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMFindAvailableVoice: error, could not find an available voice\n"); */ }
+#endif
+ return EAS_FAILURE;
+}
+
+/*----------------------------------------------------------------------------
+ * VMStealVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Steal a voice and return the voice number
+ *
+ * Stealing algorithm: steal the best choice with minimal work, taking into
+ * account SP-Midi channel priorities and polyphony allocation.
+ *
+ * In one pass through all the voices, figure out which voice to steal
+ * taking into account a number of different factors:
+ * Priority of the voice's MIDI channel
+ * Number of voices over the polyphony allocation for voice's MIDI channel
+ * Amplitude of the voice
+ * Note age
+ * Key velocity (for voices that haven't been started yet)
+ * If any matching notes are found
+ *
+ * Inputs:
+ * pnVoiceNumber - really an output, see below
+ * nChannel - the channel that this voice wants to be started on
+ * nKeyNumber - the key number for this new voice
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pnVoiceNumber - voice number of the voice that was stolen
+ * EAS_RESULT EAS_SUCCESS - always successful
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMStealVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_INT *pVoiceNumber, EAS_U8 channel, EAS_U8 note, EAS_I32 lowVoice, EAS_I32 highVoice)
+{
+ S_SYNTH_VOICE *pCurrVoice;
+ S_SYNTH *pCurrSynth;
+ EAS_INT voiceNum;
+ EAS_INT bestCandidate;
+ EAS_U8 currChannel;
+ EAS_U8 currNote;
+ EAS_I32 bestPriority;
+ EAS_I32 currentPriority;
+
+ /* determine which voice to steal */
+ bestPriority = 0;
+ bestCandidate = MAX_SYNTH_VOICES;
+
+ for (voiceNum = lowVoice; voiceNum <= highVoice; voiceNum++)
+ {
+ pCurrVoice = &pVoiceMgr->voices[voiceNum];
+
+ /* ignore free voices */
+ if (pCurrVoice->voiceState == eVoiceStateFree)
+ continue;
+
+ /* for stolen voices, use the new parameters, not the old */
+ if (pCurrVoice->voiceState == eVoiceStateStolen)
+ {
+ pCurrSynth = pVoiceMgr->pSynth[GET_VSYNTH(pCurrVoice->nextChannel)];
+ currChannel = pCurrVoice->nextChannel;
+ currNote = pCurrVoice->nextNote;
+ }
+ else
+ {
+ pCurrSynth = pVoiceMgr->pSynth[GET_VSYNTH(pCurrVoice->channel)];
+ currChannel = pCurrVoice->channel;
+ currNote = pCurrVoice->note;
+ }
+
+ /* ignore voices that are higher priority */
+ if (pSynth->priority > pCurrSynth->priority)
+ continue;
+#ifdef _DEBUG_VM
+// { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: New priority = %d exceeds old priority = %d\n", pSynth->priority, pCurrSynth->priority); */ }
+#endif
+
+ /* if voice is stolen or just started, reduce the likelihood it will be stolen */
+ if (( pCurrVoice->voiceState == eVoiceStateStolen) || (pCurrVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET))
+ {
+ currentPriority = 128 - pCurrVoice->nextVelocity;
+ }
+ else
+ {
+ /* compute the priority of this voice, higher means better for stealing */
+ /* use not age */
+ currentPriority = (EAS_I32) pCurrVoice->age << NOTE_AGE_STEAL_WEIGHT;
+
+ /* include note gain -higher gain is lower steal value */
+ /*lint -e{704} use shift for performance */
+ currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) -
+ ((EAS_I32) pCurrVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT));
+ }
+
+ /* in SP-MIDI mode, include over poly allocation and channel priority */
+ if (pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON)
+ {
+ S_SYNTH_CHANNEL *pChannel = &pCurrSynth->channels[GET_CHANNEL(currChannel)];
+ /*lint -e{701} use shift for performance */
+ if (pSynth->poolCount[pChannel->pool] >= pSynth->poolAlloc[pChannel->pool])
+ currentPriority += (pSynth->poolCount[pChannel->pool] -pSynth->poolAlloc[pChannel->pool] + 1) << CHANNEL_POLY_STEAL_WEIGHT;
+
+ /* include channel priority */
+ currentPriority += (EAS_I32)(pChannel->pool << CHANNEL_PRIORITY_STEAL_WEIGHT);
+ }
+
+ /* if a note is already playing that matches this note, consider stealing it more readily */
+ if ((note == currNote) && (channel == currChannel))
+ currentPriority += NOTE_MATCH_PENALTY;
+
+ /* is this the best choice so far? */
+ if (currentPriority >= bestPriority)
+ {
+ bestPriority = currentPriority;
+ bestCandidate = voiceNum;
+ }
+ }
+
+ /* may happen if all voices are allocated to a higher priority virtual synth */
+ if (bestCandidate == MAX_SYNTH_VOICES)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: Unable to allocate a voice\n"); */ }
+ return EAS_ERROR_NO_VOICE_ALLOCATED;
+ }
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMStealVoice: Voice %d stolen\n", bestCandidate); */ }
+
+ /* are we stealing a stolen voice? */
+ if (pVoiceMgr->voices[bestCandidate].voiceState == eVoiceStateStolen)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMStealVoice: Voice %d is already marked as stolen and was scheduled to play ch: %d note: %d vel: %d\n",
+ bestCandidate,
+ pVoiceMgr->voices[bestCandidate].nextChannel,
+ pVoiceMgr->voices[bestCandidate].nextNote,
+ pVoiceMgr->voices[bestCandidate].nextVelocity); */ }
+ }
+#endif
+
+ *pVoiceNumber = (EAS_U16) bestCandidate;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMChannelPressure()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the channel pressure for the given channel
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nVelocity - the channel pressure value
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * psSynthObject->m_sChannel[nChannel].m_nChannelPressure is updated
+ *----------------------------------------------------------------------------
+*/
+void VMChannelPressure (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 value)
+{
+ S_SYNTH_CHANNEL *pChannel;
+
+ pChannel = &(pSynth->channels[channel]);
+ pChannel->channelPressure = value;
+
+ /*
+ set a channel flag to request parameter updates
+ for all the voices associated with this channel
+ */
+ pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMPitchBend()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the pitch wheel value for the given channel.
+ * This routine constructs the proper 14-bit argument when the calling routine
+ * passes the pitch LSB and MSB.
+ *
+ * Note: some midi disassemblers display a bipolar pitch bend value.
+ * We can display the bipolar value using
+ * if m_nPitchBend >= 0x2000
+ * bipolar pitch bend = postive (m_nPitchBend - 0x2000)
+ * else
+ * bipolar pitch bend = negative (0x2000 - m_nPitchBend)
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nPitchLSB - the LSB byte of the pitch bend message
+ * nPitchMSB - the MSB byte of the pitch bend message
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * psSynthObject->m_sChannel[nChannel].m_nPitchBend is changed
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMPitchBend (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 nPitchLSB, EAS_U8 nPitchMSB)
+{
+ S_SYNTH_CHANNEL *pChannel;
+
+ pChannel = &(pSynth->channels[channel]);
+ pChannel->pitchBend = (EAS_I16) ((nPitchMSB << 7) | nPitchLSB);
+
+ /*
+ set a channel flag to request parameter updates
+ for all the voices associated with this channel
+ */
+ pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMControlChange()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the controller (or mode) for the given channel.
+ *
+ * Inputs:
+ * nChannel - the MIDI channel
+ * nControllerNumber - the MIDI controller number
+ * nControlValue - the value for this controller message
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Side Effects:
+ * psSynthObject->m_sChannel[nChannel] controller is changed
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMControlChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value)
+{
+ S_SYNTH_CHANNEL *pChannel;
+
+ pChannel = &(pSynth->channels[channel]);
+
+ /*
+ set a channel flag to request parameter updates
+ for all the voices associated with this channel
+ */
+ pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+
+ switch ( controller )
+ {
+ case MIDI_CONTROLLER_BANK_SELECT_MSB:
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: Bank Select MSB: msb 0x%X\n", value); */ }
+#endif
+ /* use this MSB with a zero LSB, until we get an LSB message */
+ pChannel->bankNum = value << 8;
+ break;
+
+ case MIDI_CONTROLLER_MOD_WHEEL:
+ /* we treat mod wheel as a 7-bit controller and only use the MSB */
+ pChannel->modWheel = value;
+ break;
+
+ case MIDI_CONTROLLER_VOLUME:
+ /* we treat volume as a 7-bit controller and only use the MSB */
+ pChannel->volume = value;
+ break;
+
+ case MIDI_CONTROLLER_PAN:
+ /* we treat pan as a 7-bit controller and only use the MSB */
+ pChannel->pan = value;
+ break;
+
+ case MIDI_CONTROLLER_EXPRESSION:
+ /* we treat expression as a 7-bit controller and only use the MSB */
+ pChannel->expression = value;
+ break;
+
+ case MIDI_CONTROLLER_BANK_SELECT_LSB:
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: Bank Select LSB: lsb 0x%X\n", value); */ }
+#endif
+ /*
+ construct bank number as 7-bits (stored as 8) of existing MSB
+ and 7-bits of new LSB (also stored as 8(
+ */
+ pChannel->bankNum =
+ (pChannel->bankNum & 0xFF00) | value;
+
+ break;
+
+ case MIDI_CONTROLLER_SUSTAIN_PEDAL:
+ /* we treat sustain pedal as a boolean on/off bit flag */
+ if (value < 64)
+ {
+ /*
+ we are requested to turn the pedal off, but first check
+ if the pedal is already on
+ */
+ if (0 !=
+ (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL)
+ )
+ {
+ /*
+ The sustain flag is presently set and the damper pedal is on.
+ We are therefore transitioning from damper pedal ON to
+ damper pedal OFF. This means all notes in this channel
+ that received a note off while the damper pedal was on, and
+ had their note-off requests deferred, should now proceed to
+ the release state.
+ */
+ VMReleaseAllDeferredNoteOffs(pVoiceMgr, pSynth, channel);
+ } /* end if sustain pedal is already on */
+
+ /* turn the sustain pedal off */
+ pChannel->channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL;
+ }
+ else
+ {
+ /*
+ we are requested to turn the pedal on, but first check
+ if the pedal is already off
+ */
+ if (0 ==
+ (pChannel->channelFlags & CHANNEL_FLAG_SUSTAIN_PEDAL)
+ )
+ {
+ /*
+ The sustain flag is presently clear and the damper pedal is off.
+ We are therefore transitioning from damper pedal OFF to
+ damper pedal ON. Currently sounding notes should be left
+ unchanged. However, we should try to "catch" notes if possible.
+ If any notes have levels >= sustain level, catch them,
+ otherwise, let them continue to release.
+ */
+ VMCatchNotesForSustainPedal(pVoiceMgr, pSynth, channel);
+ }
+
+ /* turn the sustain pedal on */
+ pChannel->channelFlags |= CHANNEL_FLAG_SUSTAIN_PEDAL;
+ }
+
+ break;
+#ifdef _REVERB
+ case MIDI_CONTROLLER_REVERB_SEND:
+ /* we treat send as a 7-bit controller and only use the MSB */
+ pSynth->channels[channel].reverbSend = value;
+ break;
+#endif
+#ifdef _CHORUS
+ case MIDI_CONTROLLER_CHORUS_SEND:
+ /* we treat send as a 7-bit controller and only use the MSB */
+ pSynth->channels[channel].chorusSend = value;
+ break;
+#endif
+ case MIDI_CONTROLLER_RESET_CONTROLLERS:
+ /* despite the Midi message name, not ALL controllers are reset */
+ pChannel->modWheel = DEFAULT_MOD_WHEEL;
+ pChannel->expression = DEFAULT_EXPRESSION;
+
+ /* turn the sustain pedal off as default/reset */
+ pChannel->channelFlags &= ~CHANNEL_FLAG_SUSTAIN_PEDAL;
+ pChannel->pitchBend = DEFAULT_PITCH_BEND;
+
+ /* reset channel pressure */
+ pChannel->channelPressure = DEFAULT_CHANNEL_PRESSURE;
+
+ /* reset RPN values */
+ pChannel->registeredParam = DEFAULT_REGISTERED_PARAM;
+ pChannel->pitchBendSensitivity = DEFAULT_PITCH_BEND_SENSITIVITY;
+ pChannel->finePitch = DEFAULT_FINE_PITCH;
+ pChannel->coarsePitch = DEFAULT_COARSE_PITCH;
+
+ /*
+ program change, bank select, channel volume CC7, pan CC10
+ are NOT reset
+ */
+ break;
+
+ /*
+ For logical reasons, the RPN data entry are grouped together.
+ However, keep in mind that these cases are not necessarily in
+ ascending order.
+ e.g., MIDI_CONTROLLER_DATA_ENTRY_MSB == 6,
+ whereas MIDI_CONTROLLER_SUSTAIN_PEDAL == 64.
+ So arrange these case statements in whatever manner is more efficient for
+ the processor / compiler.
+ */
+ case MIDI_CONTROLLER_ENTER_DATA_MSB:
+ case MIDI_CONTROLLER_ENTER_DATA_LSB:
+ case MIDI_CONTROLLER_SELECT_RPN_LSB:
+ case MIDI_CONTROLLER_SELECT_RPN_MSB:
+ case MIDI_CONTROLLER_SELECT_NRPN_MSB:
+ case MIDI_CONTROLLER_SELECT_NRPN_LSB:
+ VMUpdateRPNStateMachine(pSynth, channel, controller, value);
+ break;
+
+ case MIDI_CONTROLLER_ALL_SOUND_OFF:
+ case MIDI_CONTROLLER_ALL_NOTES_OFF:
+ case MIDI_CONTROLLER_OMNI_OFF:
+ case MIDI_CONTROLLER_OMNI_ON:
+ case MIDI_CONTROLLER_MONO_ON_POLY_OFF:
+ case MIDI_CONTROLLER_POLY_ON_MONO_OFF:
+ /* NOTE: we treat all sounds off the same as all notes off */
+ VMAllNotesOff(pVoiceMgr, pSynth, channel);
+ break;
+
+ default:
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMControlChange: controller %d not yet implemented\n", controller); */ }
+#endif
+ break;
+
+ }
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * VMUpdateRPNStateMachine()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Call this function when we want to parse RPN related controller messages.
+ * We only support RPN0 (pitch bend sensitivity), RPN1 (fine tuning) and
+ * RPN2 (coarse tuning). Any other RPNs or NRPNs are ignored for now.
+ *.
+ * Supports any order, so not a state machine anymore. This function was
+ * rewritten to work correctly regardless of order.
+ *
+ * Inputs:
+ * nChannel - the channel this controller message is coming from
+ * nControllerNumber - which RPN related controller
+ * nControlValue - the value of the RPN related controller
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * returns EAS_RESULT, which is typically EAS_SUCCESS, since there are
+ * few possible errors
+ *
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel].m_nPitchBendSensitivity
+ * (or m_nFinePitch or m_nCoarsePitch)
+ * will be updated if the proper RPN message is received.
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMUpdateRPNStateMachine (S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 controller, EAS_U8 value)
+{
+ S_SYNTH_CHANNEL *pChannel;
+
+#ifdef _DEBUG_VM
+ if (channel >= NUM_SYNTH_CHANNELS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMUpdateRPNStateMachines: error, %d invalid channel number\n",
+ channel); */ }
+ return EAS_FAILURE;
+ }
+#endif
+
+ pChannel = &(pSynth->channels[channel]);
+
+ switch (controller)
+ {
+ case MIDI_CONTROLLER_SELECT_NRPN_MSB:
+ case MIDI_CONTROLLER_SELECT_NRPN_LSB:
+ pChannel->registeredParam = DEFAULT_REGISTERED_PARAM;
+ break;
+ case MIDI_CONTROLLER_SELECT_RPN_MSB:
+ pChannel->registeredParam =
+ (pChannel->registeredParam & 0x7F) | (value<<7);
+ break;
+ case MIDI_CONTROLLER_SELECT_RPN_LSB:
+ pChannel->registeredParam =
+ (pChannel->registeredParam & 0x7F00) | value;
+ break;
+ case MIDI_CONTROLLER_ENTER_DATA_MSB:
+ switch (pChannel->registeredParam)
+ {
+ case 0:
+ pChannel->pitchBendSensitivity = value * 100;
+ break;
+ case 1:
+ /*lint -e{702} <avoid division for performance reasons>*/
+ pChannel->finePitch = (EAS_I8)((((value << 7) - 8192) * 100) >> 13);
+ break;
+ case 2:
+ pChannel->coarsePitch = (EAS_I8)(value - 64);
+ break;
+ default:
+ break;
+ }
+ break;
+ case MIDI_CONTROLLER_ENTER_DATA_LSB:
+ switch (pChannel->registeredParam)
+ {
+ case 0:
+ //ignore lsb
+ break;
+ case 1:
+ //ignore lsb
+ break;
+ case 2:
+ //ignore lsb
+ break;
+ default:
+ break;
+ }
+ break;
+ default:
+ return EAS_FAILURE; //not a RPN related controller
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMUpdateStaticChannelParameters()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Update all of the static channel parameters for channels that have had
+ * a controller change values
+ * Or if the synth has signalled that all channels must forcibly
+ * be updated
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * none
+ *
+ * Side Effects:
+ * - psSynthObject->m_sChannel[].m_nStaticGain and m_nStaticPitch
+ * are updated for channels whose controller values have changed
+ * or if the synth has signalled that all channels must forcibly
+ * be updated
+ *----------------------------------------------------------------------------
+*/
+void VMUpdateStaticChannelParameters (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth)
+{
+ EAS_INT channel;
+
+ if (pSynth->synthFlags & SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS)
+ {
+ /*
+ the synth wants us to forcibly update all channel
+ parameters. This event occurs when we are about to
+ finish resetting the synth
+ */
+ for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++)
+ {
+#ifdef _HYBRID_SYNTH
+ if (pSynth->channels[channel].regionIndex & FLAG_RGN_IDX_FM_SYNTH)
+ pSecondarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+ else
+ pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+#else
+ pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+#endif
+ }
+
+ /*
+ clear the flag to indicates we have now forcibly
+ updated all channel parameters
+ */
+ pSynth->synthFlags &= ~SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS;
+ }
+ else
+ {
+
+ /* only update channel params if signalled by a channel flag */
+ for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++)
+ {
+ if ( 0 != (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS))
+ {
+#ifdef _HYBRID_SYNTH
+ if (pSynth->channels[channel].regionIndex & FLAG_RGN_IDX_FM_SYNTH)
+ pSecondarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+ else
+ pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+#else
+ pPrimarySynth->pfUpdateChannel(pVoiceMgr, pSynth, (EAS_U8) channel);
+#endif
+ }
+ }
+
+ }
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * VMFindProgram()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Look up an individual program in sound library. This function
+ * searches the bank list for a program, then the individual program
+ * list.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT VMFindProgram (const S_EAS *pEAS, EAS_U32 bank, EAS_U8 programNum, EAS_U16 *pRegionIndex)
+{
+ EAS_U32 locale;
+ const S_PROGRAM *p;
+ EAS_U16 i;
+ EAS_U16 regionIndex;
+
+ /* make sure we have a valid sound library */
+ if (pEAS == NULL)
+ return EAS_FAILURE;
+
+ /* search the banks */
+ for (i = 0; i < pEAS->numBanks; i++)
+ {
+ if (bank == (EAS_U32) pEAS->pBanks[i].locale)
+ {
+ regionIndex = pEAS->pBanks[i].regionIndex[programNum];
+ if (regionIndex != INVALID_REGION_INDEX)
+ {
+ *pRegionIndex = regionIndex;
+ return EAS_SUCCESS;
+ }
+ break;
+ }
+ }
+
+ /* establish locale */
+ locale = ( bank << 8) | programNum;
+
+ /* search for program */
+ for (i = 0, p = pEAS->pPrograms; i < pEAS->numPrograms; i++, p++)
+ {
+ if (p->locale == locale)
+ {
+ *pRegionIndex = p->regionIndex;
+ return EAS_SUCCESS;
+ }
+ }
+
+ return EAS_FAILURE;
+}
+
+#ifdef DLS_SYNTHESIZER
+/*----------------------------------------------------------------------------
+ * VMFindDLSProgram()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Look up an individual program in sound library. This function
+ * searches the bank list for a program, then the individual program
+ * list.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT VMFindDLSProgram (const S_DLS *pDLS, EAS_U32 bank, EAS_U8 programNum, EAS_U16 *pRegionIndex)
+{
+ EAS_U32 locale;
+ const S_PROGRAM *p;
+ EAS_U16 i;
+
+ /* make sure we have a valid sound library */
+ if (pDLS == NULL)
+ return EAS_FAILURE;
+
+ /* establish locale */
+ locale = (bank << 8) | programNum;
+
+ /* search for program */
+ for (i = 0, p = pDLS->pDLSPrograms; i < pDLS->numDLSPrograms; i++, p++)
+ {
+ if (p->locale == locale)
+ {
+ *pRegionIndex = p->regionIndex;
+ return EAS_SUCCESS;
+ }
+ }
+
+ return EAS_FAILURE;
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * VMProgramChange()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Change the instrument (program) for the given channel.
+ *
+ * Depending on the program number, and the bank selected for this channel, the
+ * program may be in ROM, RAM (from SMAF or CMX related RAM wavetable), or
+ * Alternate wavetable (from mobile DLS or other DLS file)
+ *
+ * This function figures out what wavetable should be used, and sets it up as the
+ * wavetable to use for this channel. Also the channel may switch from a melodic
+ * channel to a rhythm channel, or vice versa.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Side Effects:
+ * gsSynthObject.m_sChannel[nChannel].m_nProgramNumber is likely changed
+ * gsSynthObject.m_sChannel[nChannel].m_psEAS may be changed
+ * gsSynthObject.m_sChannel[nChannel].m_bRhythmChannel may be changed
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+void VMProgramChange (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel, EAS_U8 program)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_U32 bank;
+ EAS_U16 regionIndex;
+
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "VMProgramChange: vSynthNum=%d, channel=%d, program=%d\n", pSynth->vSynthNum, channel, program); */ }
+#endif
+
+ /* setup pointer to MIDI channel data */
+ pChannel = &pSynth->channels[channel];
+ bank = pChannel->bankNum;
+
+ /* allow channels to switch between being melodic or rhythm channels, using GM2 CC values */
+ if ((bank & 0xFF00) == DEFAULT_RHYTHM_BANK_NUMBER)
+ {
+ /* make it a rhythm channel */
+ pChannel->channelFlags |= CHANNEL_FLAG_RHYTHM_CHANNEL;
+ }
+ else if ((bank & 0xFF00) == DEFAULT_MELODY_BANK_NUMBER)
+ {
+ /* make it a melody channel */
+ pChannel->channelFlags &= ~CHANNEL_FLAG_RHYTHM_CHANNEL;
+ }
+
+ regionIndex = DEFAULT_REGION_INDEX;
+
+#ifdef EXTERNAL_AUDIO
+ /* give the external audio interface a chance to handle it */
+ if (pSynth->cbProgChgFunc != NULL)
+ {
+ S_EXT_AUDIO_PRG_CHG prgChg;
+ prgChg.channel = channel;
+ prgChg.bank = (EAS_U16) bank;
+ prgChg.program = program;
+ if (pSynth->cbProgChgFunc(pSynth->pExtAudioInstData, &prgChg))
+ pChannel->channelFlags |= CHANNEL_FLAG_EXTERNAL_AUDIO;
+ }
+
+#endif
+
+
+#ifdef DLS_SYNTHESIZER
+ /* first check for DLS program that may overlay the internal instrument */
+ if (VMFindDLSProgram(pSynth->pDLS, bank, program, ®ionIndex) != EAS_SUCCESS)
+#endif
+
+ /* braces to support 'if' clause above */
+ {
+
+ /* look in the internal banks */
+ if (VMFindProgram(pSynth->pEAS, bank, program, ®ionIndex) != EAS_SUCCESS)
+
+ /* fall back to default bank */
+ {
+ if (pSynth->channels[channel].channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)
+ bank = DEFAULT_RHYTHM_BANK_NUMBER;
+ else
+ bank = DEFAULT_MELODY_BANK_NUMBER;
+
+ if (VMFindProgram(pSynth->pEAS, bank, program, ®ionIndex) != EAS_SUCCESS)
+
+ /* switch to program 0 in the default bank */
+ {
+ if (VMFindProgram(pSynth->pEAS, bank, 0, ®ionIndex) != EAS_SUCCESS)
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMProgramChange: No program @ %03d:%03d:%03d\n",
+ (bank >> 8) & 0x7f, bank & 0x7f, program); */ }
+ }
+ }
+ }
+
+ /* we have our new program change for this channel */
+ pChannel->programNum = program;
+ pChannel->regionIndex = regionIndex;
+
+ /*
+ set a channel flag to request parameter updates
+ for all the voices associated with this channel
+ */
+ pChannel->channelFlags |= CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * VMAddSamples()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesize the requested number of samples (block based processing)
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to write to buffer
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * number of voices rendered
+ *
+ * Side Effects:
+ * - samples are added to the presently free buffer
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 VMAddSamples (S_VOICE_MGR *pVoiceMgr, EAS_I32 *pMixBuffer, EAS_I32 numSamples)
+{
+ S_SYNTH *pSynth;
+ EAS_INT voicesRendered;
+ EAS_INT voiceNum;
+ EAS_BOOL done;
+
+#ifdef _REVERB
+ EAS_PCM *pReverbSendBuffer;
+#endif // ifdef _REVERB
+
+#ifdef _CHORUS
+ EAS_PCM *pChorusSendBuffer;
+#endif // ifdef _CHORUS
+
+ voicesRendered = 0;
+ for (voiceNum = 0; voiceNum < MAX_SYNTH_VOICES; voiceNum++)
+ {
+
+ /* retarget stolen voices */
+ if ((pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen) && (pVoiceMgr->voices[voiceNum].gain <= 0))
+ VMRetargetStolenVoice(pVoiceMgr, voiceNum);
+
+ /* get pointer to virtual synth */
+ pSynth = pVoiceMgr->pSynth[pVoiceMgr->voices[voiceNum].channel >> 4];
+
+ /* synthesize active voices */
+ if (pVoiceMgr->voices[voiceNum].voiceState != eVoiceStateFree)
+ {
+ done = GetSynthPtr(voiceNum)->pfUpdateVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum], GetAdjustedVoiceNum(voiceNum), pMixBuffer, numSamples);
+ voicesRendered++;
+
+ /* voice is finished */
+ if (done == EAS_TRUE)
+ {
+ /* set gain of stolen voice to zero so it will be restarted */
+ if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStolen)
+ pVoiceMgr->voices[voiceNum].gain = 0;
+
+ /* or return it to the free voice pool */
+ else
+ VMFreeVoice(pVoiceMgr, pSynth, &pVoiceMgr->voices[voiceNum]);
+ }
+
+ /* if this voice is scheduled to be muted, set the mute flag */
+ if (pVoiceMgr->voices[voiceNum].voiceFlags & VOICE_FLAG_DEFER_MUTE)
+ {
+ pVoiceMgr->voices[voiceNum].voiceFlags &= ~(VOICE_FLAG_DEFER_MUTE | VOICE_FLAG_DEFER_MIDI_NOTE_OFF);
+ VMMuteVoice(pVoiceMgr, voiceNum);
+ }
+
+ /* if voice just started, advance state to play */
+ if (pVoiceMgr->voices[voiceNum].voiceState == eVoiceStateStart)
+ pVoiceMgr->voices[voiceNum].voiceState = eVoiceStatePlay;
+ }
+ }
+
+ return voicesRendered;
+}
+
+/*----------------------------------------------------------------------------
+ * VMRender()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This routine renders a frame of audio
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * pVoicesRendered - number of voices rendered this frame
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMRender (S_VOICE_MGR *pVoiceMgr, EAS_I32 numSamples, EAS_I32 *pMixBuffer, EAS_I32 *pVoicesRendered)
+{
+ S_SYNTH *pSynth;
+ EAS_INT i;
+ EAS_INT channel;
+
+#ifdef _CHECKED_BUILD
+ SanityCheck(pVoiceMgr);
+#endif
+
+ /* update MIDI channel parameters */
+ *pVoicesRendered = 0;
+ for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++)
+ {
+ if (pVoiceMgr->pSynth[i] != NULL)
+ VMUpdateStaticChannelParameters(pVoiceMgr, pVoiceMgr->pSynth[i]);
+ }
+
+ /* synthesize a buffer of audio */
+ *pVoicesRendered = VMAddSamples(pVoiceMgr, pMixBuffer, numSamples);
+
+ /*
+ * check for deferred note-off messages
+ * If flag is set, that means one or more voices are expecting deferred
+ * midi note-off messages because the midi note-on and corresponding midi
+ * note-off requests occurred during the same update interval. The goal
+ * is the defer the note-off request so that the note can at least start.
+ */
+ for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++)
+ {
+ pSynth = pVoiceMgr->pSynth[i];
+
+ if (pSynth== NULL)
+ continue;
+
+ if (pSynth->synthFlags & SYNTH_FLAG_DEFERRED_MIDI_NOTE_OFF_PENDING)
+ VMDeferredStopNote(pVoiceMgr, pSynth);
+
+ /* check if we need to reset the synth */
+ if ((pSynth->synthFlags & SYNTH_FLAG_RESET_IS_REQUESTED) &&
+ (pSynth->numActiveVoices == 0))
+ {
+ /*
+ complete the process of resetting the synth now that
+ all voices have muted
+ */
+#ifdef _DEBUG_VM
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "VMAddSamples: complete the reset process\n"); */ }
+#endif
+
+ VMInitializeAllChannels(pVoiceMgr, pSynth);
+ VMInitializeAllVoices(pVoiceMgr, pSynth->vSynthNum);
+
+ /* clear the reset flag */
+ pSynth->synthFlags &= ~SYNTH_FLAG_RESET_IS_REQUESTED;
+ }
+
+ /* clear channel update flags */
+ for (channel = 0; channel < NUM_SYNTH_CHANNELS; channel++)
+ pSynth->channels[channel].channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+
+ }
+
+#ifdef _CHECKED_BUILD
+ SanityCheck(pVoiceMgr);
+#endif
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMInitWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clears the workload counter
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMInitWorkload (S_VOICE_MGR *pVoiceMgr)
+{
+ pVoiceMgr->workload = 0;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the max workload for a single frame.
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetWorkload (S_VOICE_MGR *pVoiceMgr, EAS_I32 maxWorkLoad)
+{
+ pVoiceMgr->maxWorkLoad = maxWorkLoad;
+}
+
+/*----------------------------------------------------------------------------
+ * VMCheckWorkload()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Checks to see if work load has been exceeded on this frame.
+ *
+ * Inputs:
+ * pVoiceMgr - pointer to instance data
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMCheckWorkload (S_VOICE_MGR *pVoiceMgr)
+{
+ if (pVoiceMgr->maxWorkLoad > 0)
+ return (EAS_BOOL) (pVoiceMgr->workload >= pVoiceMgr->maxWorkLoad);
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * VMActiveVoices()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the number of active voices in the synthesizer.
+ *
+ * Inputs:
+ * pEASData - pointer to instance data
+ *
+ * Outputs:
+ * Returns the number of active voices
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 VMActiveVoices (S_SYNTH *pSynth)
+{
+ return pSynth->numActiveVoices;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetSynthPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the synth to a new polyphony value. Value must be >= 1 and
+ * <= MAX_SYNTH_VOICES. This function will pin the polyphony at those limits
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * polyphonyCount desired polyphony count
+ * synth synthesizer number (0 = onboard, 1 = DSP)
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 polyphonyCount)
+{
+ EAS_INT i;
+ EAS_INT activeVoices;
+
+ /* lower limit */
+ if (polyphonyCount < 1)
+ polyphonyCount = 1;
+
+ /* split architecture */
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+ if (synth == EAS_MCU_SYNTH)
+ {
+ if (polyphonyCount > NUM_PRIMARY_VOICES)
+ polyphonyCount = NUM_PRIMARY_VOICES;
+ if (pVoiceMgr->maxPolyphonyPrimary == polyphonyCount)
+ return EAS_SUCCESS;
+ pVoiceMgr->maxPolyphonyPrimary = (EAS_U16) polyphonyCount;
+ }
+ else if (synth == EAS_DSP_SYNTH)
+ {
+ if (polyphonyCount > NUM_SECONDARY_VOICES)
+ polyphonyCount = NUM_SECONDARY_VOICES;
+ if (pVoiceMgr->maxPolyphonySecondary == polyphonyCount)
+ return EAS_SUCCESS;
+ pVoiceMgr->maxPolyphonySecondary = (EAS_U16) polyphonyCount;
+ }
+ else
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* setting for SP-MIDI */
+ pVoiceMgr->maxPolyphony = pVoiceMgr->maxPolyphonyPrimary + pVoiceMgr->maxPolyphonySecondary;
+
+ /* standard architecture */
+#else
+ if (synth != EAS_MCU_SYNTH)
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* pin desired value to possible limits */
+ if (polyphonyCount > MAX_SYNTH_VOICES)
+ polyphonyCount = MAX_SYNTH_VOICES;
+
+ /* set polyphony, if value is different than current value */
+ if (pVoiceMgr->maxPolyphony == polyphonyCount)
+ return EAS_SUCCESS;
+
+ pVoiceMgr->maxPolyphony = (EAS_U16) polyphonyCount;
+#endif
+
+ /* if SPMIDI enabled, update channel masking based on new polyphony */
+ for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++)
+ {
+ if (pVoiceMgr->pSynth[i])
+ {
+ if (pVoiceMgr->pSynth[i]->synthFlags & SYNTH_FLAG_SP_MIDI_ON)
+ VMMIPUpdateChannelMuting(pVoiceMgr, pVoiceMgr->pSynth[i]);
+ else
+ pVoiceMgr->pSynth[i]->poolAlloc[0] = (EAS_U8) polyphonyCount;
+ }
+ }
+
+ /* are we under polyphony limit? */
+ if (pVoiceMgr->activeVoices <= polyphonyCount)
+ return EAS_SUCCESS;
+
+ /* count the number of active voices */
+ activeVoices = 0;
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+
+ /* is voice active? */
+ if ((pVoiceMgr->voices[i].voiceState != eVoiceStateFree) && (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting))
+ activeVoices++;
+ }
+
+ /* we may have to mute voices to reach new target */
+ while (activeVoices > polyphonyCount)
+ {
+ S_SYNTH *pSynth;
+ S_SYNTH_VOICE *pVoice;
+ EAS_I32 currentPriority, bestPriority;
+ EAS_INT bestCandidate;
+
+ /* find the lowest priority voice */
+ bestPriority = bestCandidate = -1;
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+
+ pVoice = &pVoiceMgr->voices[i];
+
+ /* ignore free and muting voices */
+ if ((pVoice->voiceState == eVoiceStateFree) || (pVoice->voiceState == eVoiceStateMuting))
+ continue;
+
+ pSynth = pVoiceMgr->pSynth[GET_VSYNTH(pVoice->channel)];
+
+ /* if voice is stolen or just started, reduce the likelihood it will be stolen */
+ if (( pVoice->voiceState == eVoiceStateStolen) || (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET))
+ {
+ /* include velocity */
+ currentPriority = 128 - pVoice->nextVelocity;
+
+ /* include channel priority */
+ currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT;
+ }
+ else
+ {
+ /* include age */
+ currentPriority = (EAS_I32) pVoice->age << NOTE_AGE_STEAL_WEIGHT;
+
+ /* include note gain -higher gain is lower steal value */
+ /*lint -e{704} use shift for performance */
+ currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) -
+ ((EAS_I32) pVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT));
+
+ /* include channel priority */
+ currentPriority += pSynth->channels[GET_CHANNEL(pVoice->channel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT;
+ }
+
+ /* include synth priority */
+ currentPriority += pSynth->priority << SYNTH_PRIORITY_WEIGHT;
+
+ /* is this the best choice so far? */
+ if (currentPriority > bestPriority)
+ {
+ bestPriority = currentPriority;
+ bestCandidate = i;
+ }
+ }
+
+ /* shutdown best candidate */
+ if (bestCandidate < 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMSetPolyphony: Unable to reduce polyphony\n"); */ }
+ break;
+ }
+
+ /* shut down this voice */
+ /*lint -e{771} pSynth is initialized if bestCandidate >= 0 */
+ VMMuteVoice(pVoiceMgr, bestCandidate);
+ activeVoices--;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetSynthPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current polyphony setting
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * synth synthesizer number (0 = onboard, 1 = DSP)
+ *
+ * Outputs:
+ * Returns actual polyphony value set, as pinned by limits
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMGetSynthPolyphony (S_VOICE_MGR *pVoiceMgr, EAS_I32 synth, EAS_I32 *pPolyphonyCount)
+{
+
+#if defined(_SECONDARY_SYNTH) || defined(EAS_SPLIT_WT_SYNTH)
+ if (synth == EAS_MCU_SYNTH)
+ *pPolyphonyCount = pVoiceMgr->maxPolyphonyPrimary;
+ else if (synth == EAS_DSP_SYNTH)
+ *pPolyphonyCount = pVoiceMgr->maxPolyphonySecondary;
+ else
+ return EAS_ERROR_PARAMETER_RANGE;
+#else
+ if (synth != EAS_MCU_SYNTH)
+ return EAS_ERROR_PARAMETER_RANGE;
+ *pPolyphonyCount = pVoiceMgr->maxPolyphony;
+#endif
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the virtual synth polyphony. 0 = no limit (i.e. can use
+ * all available voices).
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * polyphonyCount desired polyphony count
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 polyphonyCount)
+{
+ EAS_INT i;
+ EAS_INT activeVoices;
+
+ /* check limits */
+ if (polyphonyCount < 0)
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* zero is max polyphony */
+ if ((polyphonyCount == 0) || (polyphonyCount > MAX_SYNTH_VOICES))
+ {
+ pSynth->maxPolyphony = 0;
+ return EAS_SUCCESS;
+ }
+
+ /* set new polyphony */
+ pSynth->maxPolyphony = (EAS_U16) polyphonyCount;
+
+ /* max polyphony is minimum of virtual synth and actual synth */
+ if (polyphonyCount > pVoiceMgr->maxPolyphony)
+ polyphonyCount = pVoiceMgr->maxPolyphony;
+
+ /* if SP-MIDI mode, update the channel muting */
+ if (pSynth->synthFlags & SYNTH_FLAG_SP_MIDI_ON)
+ VMMIPUpdateChannelMuting(pVoiceMgr, pSynth);
+ else
+ pSynth->poolAlloc[0] = (EAS_U8) polyphonyCount;
+
+ /* are we under polyphony limit? */
+ if (pSynth->numActiveVoices <= polyphonyCount)
+ return EAS_SUCCESS;
+
+ /* count the number of active voices */
+ activeVoices = 0;
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ /* this synth? */
+ if (GET_VSYNTH(pVoiceMgr->voices[i].nextChannel) != pSynth->vSynthNum)
+ continue;
+
+ /* is voice active? */
+ if ((pVoiceMgr->voices[i].voiceState != eVoiceStateFree) && (pVoiceMgr->voices[i].voiceState != eVoiceStateMuting))
+ activeVoices++;
+ }
+
+ /* we may have to mute voices to reach new target */
+ while (activeVoices > polyphonyCount)
+ {
+ S_SYNTH_VOICE *pVoice;
+ EAS_I32 currentPriority, bestPriority;
+ EAS_INT bestCandidate;
+
+ /* find the lowest priority voice */
+ bestPriority = bestCandidate = -1;
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ pVoice = &pVoiceMgr->voices[i];
+
+ /* this synth? */
+ if (GET_VSYNTH(pVoice->nextChannel) != pSynth->vSynthNum)
+ continue;
+
+ /* if voice is stolen or just started, reduce the likelihood it will be stolen */
+ if (( pVoice->voiceState == eVoiceStateStolen) || (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET))
+ {
+ /* include velocity */
+ currentPriority = 128 - pVoice->nextVelocity;
+
+ /* include channel priority */
+ currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT;
+ }
+ else
+ {
+ /* include age */
+ currentPriority = (EAS_I32) pVoice->age << NOTE_AGE_STEAL_WEIGHT;
+
+ /* include note gain -higher gain is lower steal value */
+ /*lint -e{704} use shift for performance */
+ currentPriority += ((32768 >> (12 - NOTE_GAIN_STEAL_WEIGHT)) + 256) -
+ ((EAS_I32) pVoice->gain >> (12 - NOTE_GAIN_STEAL_WEIGHT));
+
+ /* include channel priority */
+ currentPriority += pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool << CHANNEL_PRIORITY_STEAL_WEIGHT;
+ }
+
+ /* is this the best choice so far? */
+ if (currentPriority > bestPriority)
+ {
+ bestPriority = currentPriority;
+ bestCandidate = i;
+ }
+ }
+
+ /* shutdown best candidate */
+ if (bestCandidate < 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "VMSetPolyphony: Unable to reduce polyphony\n"); */ }
+ break;
+ }
+
+ /* shut down this voice */
+ VMMuteVoice(pVoiceMgr, bestCandidate);
+ activeVoices--;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetPolyphony()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get the virtual synth polyphony
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * pPolyphonyCount pointer to variable to hold polyphony count
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+EAS_RESULT VMGetPolyphony (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPolyphonyCount)
+{
+ *pPolyphonyCount = (EAS_U16) pSynth->maxPolyphony;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetPriority()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the virtual synth priority
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * priority new priority
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+EAS_RESULT VMSetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 priority)
+{
+ pSynth->priority = (EAS_U8) priority ;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetPriority()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get the virtual synth priority
+ *
+ * Inputs:
+ * pVoiceMgr pointer to synthesizer data
+ * pPriority pointer to variable to hold priority
+ * pSynth pointer to virtual synth
+ *
+ * Outputs:
+ * Returns error code
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) reserved for future use */
+EAS_RESULT VMGetPriority (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_I32 *pPriority)
+{
+ *pPriority = pSynth->priority;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetVolume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Set the master volume for this synthesizer for this sequence.
+ *
+ * Inputs:
+ * nSynthVolume - the desired master volume
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ * overrides any previously set master volume from sysex
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetVolume (S_SYNTH *pSynth, EAS_U16 masterVolume)
+{
+ pSynth->masterVolume = masterVolume;
+ pSynth->synthFlags |= SYNTH_FLAG_UPDATE_ALL_CHANNEL_PARAMETERS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetPitchBendRange()
+ *----------------------------------------------------------------------------
+ * Set the pitch bend range for the given channel.
+ *----------------------------------------------------------------------------
+*/
+void VMSetPitchBendRange (S_SYNTH *pSynth, EAS_INT channel, EAS_I16 pitchBendRange)
+{
+ pSynth->channels[channel].pitchBendSensitivity = pitchBendRange;
+}
+
+/*----------------------------------------------------------------------------
+ * VMValidateEASLib()
+ *----------------------------------------------------------------------------
+ * Validates an EAS library
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMValidateEASLib (EAS_SNDLIB_HANDLE pEAS)
+{
+ /* validate the sound library */
+ if (pEAS)
+ {
+ if (pEAS->identifier != _EAS_LIBRARY_VERSION)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Sound library mismatch in sound library: Read 0x%08x, expected 0x%08x\n",
+ pEAS->identifier, _EAS_LIBRARY_VERSION); */ }
+ return EAS_ERROR_SOUND_LIBRARY;
+ }
+
+ /* check sample rate */
+ if ((pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK) != _OUTPUT_SAMPLE_RATE)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Sample rate mismatch in sound library: Read %lu, expected %lu\n",
+ pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ }
+ return EAS_ERROR_SOUND_LIBRARY;
+ }
+
+#ifdef _WT_SYNTH
+ /* check sample bit depth */
+#ifdef _8_BIT_SAMPLES
+ if (pEAS->libAttr & LIB_FORMAT_16_BIT_SAMPLES)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Expected 8-bit samples and found 16-bit\n",
+ pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ }
+ return EAS_ERROR_SOUND_LIBRARY;
+ }
+#endif
+#ifdef _16_BIT_SAMPLES
+ if ((pEAS->libAttr & LIB_FORMAT_16_BIT_SAMPLES) == 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMValidateEASLib: Expected 16-bit samples and found 8-bit\n",
+ pEAS->libAttr & LIBFORMAT_SAMPLE_RATE_MASK, _OUTPUT_SAMPLE_RATE); */ }
+ return EAS_ERROR_SOUND_LIBRARY;
+ }
+#endif
+#endif
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetGlobalEASLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the EAS library to be used by the synthesizer
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetGlobalEASLib (S_VOICE_MGR *pVoiceMgr, EAS_SNDLIB_HANDLE pEAS)
+{
+ EAS_RESULT result;
+
+ result = VMValidateEASLib(pEAS);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ pVoiceMgr->pGlobalEAS = pEAS;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetEASLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the EAS library to be used by the synthesizer
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetEASLib (S_SYNTH *pSynth, EAS_SNDLIB_HANDLE pEAS)
+{
+ EAS_RESULT result;
+
+ result = VMValidateEASLib(pEAS);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ pSynth->pEAS = pEAS;
+ return EAS_SUCCESS;
+}
+
+#ifdef DLS_SYNTHESIZER
+/*----------------------------------------------------------------------------
+ * VMSetGlobalDLSLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the DLS library to be used by the synthesizer
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetGlobalDLSLib (EAS_DATA_HANDLE pEASData, EAS_DLSLIB_HANDLE pDLS)
+{
+
+ if (pEASData->pVoiceMgr->pGlobalDLS)
+ DLSCleanup(pEASData->hwInstData, pEASData->pVoiceMgr->pGlobalDLS);
+
+ pEASData->pVoiceMgr->pGlobalDLS = pDLS;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * VMSetDLSLib()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the DLS library to be used by the synthesizer
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSetDLSLib (S_SYNTH *pSynth, EAS_DLSLIB_HANDLE pDLS)
+{
+ pSynth->pDLS = pDLS;
+ return EAS_SUCCESS;
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * VMSetTranposition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the global key transposition used by the synthesizer.
+ * Transposes all melodic instruments up or down by the specified
+ * amount. Range is limited to +/-12 semitones.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMSetTranposition (S_SYNTH *pSynth, EAS_I32 transposition)
+{
+ pSynth->globalTranspose = (EAS_I8) transposition;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetTranposition()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Gets the global key transposition used by the synthesizer.
+ * Transposes all melodic instruments up or down by the specified
+ * amount. Range is limited to +/-12 semitones.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMGetTranposition (S_SYNTH *pSynth, EAS_I32 *pTransposition)
+{
+ *pTransposition = pSynth->globalTranspose;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetNoteCount()
+ *----------------------------------------------------------------------------
+* Returns the total note count
+*----------------------------------------------------------------------------
+*/
+EAS_I32 VMGetNoteCount (S_SYNTH *pSynth)
+{
+ return pSynth->totalNoteCount;
+}
+
+/*----------------------------------------------------------------------------
+ * VMMIDIShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clean up any Synth related system issues.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMMIDIShutdown (S_EAS_DATA *pEASData, S_SYNTH *pSynth)
+{
+ EAS_INT vSynthNum;
+
+ /* decrement reference count, free if all references are gone */
+ if (--pSynth->refCount > 0)
+ return;
+
+ vSynthNum = pSynth->vSynthNum;
+
+ /* cleanup DLS load */
+#ifdef DLS_SYNTHESIZER
+ /*lint -e{550} result used only in debugging code */
+ if (pSynth->pDLS != NULL)
+ {
+ EAS_RESULT result;
+ if ((result = DLSCleanup(pEASData->hwInstData, pSynth->pDLS)) != EAS_SUCCESS)
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMMIDIShutdown: Error %ld cleaning up DLS collection\n", result); */ }
+ pSynth->pDLS = NULL;
+ }
+#endif
+
+ VMReset(pEASData->pVoiceMgr, pSynth, EAS_TRUE);
+
+ /* check Configuration Module for static memory allocation */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pSynth);
+
+ /* clear pointer to MIDI state */
+ pEASData->pVoiceMgr->pSynth[vSynthNum] = NULL;
+}
+
+/*----------------------------------------------------------------------------
+ * VMShutdown()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Clean up any Synth related system issues.
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * None
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void VMShutdown (S_EAS_DATA *pEASData)
+{
+
+ /* don't free a NULL pointer */
+ if (pEASData->pVoiceMgr == NULL)
+ return;
+
+#ifdef DLS_SYNTHESIZER
+ /* if we have a global DLS collection, clean it up */
+ if (pEASData->pVoiceMgr->pGlobalDLS)
+ {
+ DLSCleanup(pEASData->hwInstData, pEASData->pVoiceMgr->pGlobalDLS);
+ pEASData->pVoiceMgr->pGlobalDLS = NULL;
+ }
+#endif
+
+ /* check Configuration Module for static memory allocation */
+ if (!pEASData->staticMemoryModel)
+ EAS_HWFree(pEASData->hwInstData, pEASData->pVoiceMgr);
+ pEASData->pVoiceMgr = NULL;
+}
+
+#ifdef EXTERNAL_AUDIO
+/*----------------------------------------------------------------------------
+ * EAS_RegExtAudioCallback()
+ *----------------------------------------------------------------------------
+ * Register a callback for external audio processing
+ *----------------------------------------------------------------------------
+*/
+void VMRegExtAudioCallback (S_SYNTH *pSynth, EAS_VOID_PTR pInstData, EAS_EXT_PRG_CHG_FUNC cbProgChgFunc, EAS_EXT_EVENT_FUNC cbEventFunc)
+{
+ pSynth->pExtAudioInstData = pInstData;
+ pSynth->cbProgChgFunc = cbProgChgFunc;
+ pSynth->cbEventFunc = cbEventFunc;
+}
+
+/*----------------------------------------------------------------------------
+ * VMGetMIDIControllers()
+ *----------------------------------------------------------------------------
+ * Returns the MIDI controller values on the specified channel
+ *----------------------------------------------------------------------------
+*/
+void VMGetMIDIControllers (S_SYNTH *pSynth, EAS_U8 channel, S_MIDI_CONTROLLERS *pControl)
+{
+ pControl->modWheel = pSynth->channels[channel].modWheel;
+ pControl->volume = pSynth->channels[channel].volume;
+ pControl->pan = pSynth->channels[channel].pan;
+ pControl->expression = pSynth->channels[channel].expression;
+ pControl->channelPressure = pSynth->channels[channel].channelPressure;
+
+#ifdef _REVERB
+ pControl->reverbSend = pSynth->channels[channel].reverbSend;
+#endif
+
+#ifdef _CHORUSE
+ pControl->chorusSend = pSynth->channels[channel].chorusSend;
+#endif
+}
+#endif
+
+#ifdef _SPLIT_ARCHITECTURE
+/*----------------------------------------------------------------------------
+ * VMStartFrame()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Starts an audio frame
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Returns true if EAS_MixEnginePrep should be called (onboard mixing)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMStartFrame (S_EAS_DATA *pEASData)
+{
+
+ /* init counter for voices starts in split architecture */
+#ifdef MAX_VOICE_STARTS
+ pVoiceMgr->numVoiceStarts = 0;
+#endif
+
+ return pFrameInterface->pfStartFrame(pEASData->pVoiceMgr->pFrameBuffer);
+}
+
+/*----------------------------------------------------------------------------
+ * VMEndFrame()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Stops an audio frame
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Returns true if EAS_MixEnginePost should be called (onboard mixing)
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL VMEndFrame (S_EAS_DATA *pEASData)
+{
+
+ return pFrameInterface->pfEndFrame(pEASData->pVoiceMgr->pFrameBuffer, pEASData->pMixBuffer, pEASData->masterGain);
+}
+#endif
+
+#ifdef TEST_HARNESS
+/*----------------------------------------------------------------------------
+ * SanityCheck()
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT VMSanityCheck (EAS_DATA_HANDLE pEASData)
+{
+ S_SYNTH_VOICE *pVoice;
+ S_SYNTH *pSynth;
+ EAS_INT i;
+ EAS_INT j;
+ EAS_INT freeVoices;
+ EAS_INT activeVoices;
+ EAS_INT playingVoices;
+ EAS_INT stolenVoices;
+ EAS_INT releasingVoices;
+ EAS_INT mutingVoices;
+ EAS_INT poolCount[MAX_VIRTUAL_SYNTHESIZERS][NUM_SYNTH_CHANNELS];
+ EAS_INT vSynthNum;
+ EAS_RESULT result = EAS_SUCCESS;
+
+ /* initialize counts */
+ EAS_HWMemSet(poolCount, 0, sizeof(poolCount));
+ freeVoices = activeVoices = playingVoices = stolenVoices = releasingVoices = mutingVoices = 0;
+
+ /* iterate through all voices */
+ for (i = 0; i < MAX_SYNTH_VOICES; i++)
+ {
+ pVoice = &pEASData->pVoiceMgr->voices[i];
+ if (pVoice->voiceState != eVoiceStateFree)
+ {
+ vSynthNum = GET_VSYNTH(pVoice->channel);
+ if (vSynthNum >= MAX_VIRTUAL_SYNTHESIZERS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Voice %d has invalid virtual synth number %d\n", i, vSynthNum); */ }
+ result = EAS_FAILURE;
+ continue;
+ }
+ pSynth = pEASData->pVoiceMgr->pSynth[vSynthNum];
+
+ switch (pVoice->voiceState)
+ {
+ case eVoiceStateMuting:
+ activeVoices++;
+ mutingVoices++;
+ break;
+
+ case eVoiceStateStolen:
+ vSynthNum = GET_VSYNTH(pVoice->nextChannel);
+ if (vSynthNum >= MAX_VIRTUAL_SYNTHESIZERS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Voice %d has invalid virtual synth number %d\n", i, vSynthNum); */ }
+ result = EAS_FAILURE;
+ continue;
+ }
+ pSynth = pEASData->pVoiceMgr->pSynth[vSynthNum];
+ activeVoices++;
+ stolenVoices++;
+ poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->nextChannel)].pool]++;
+ break;
+
+ case eVoiceStateStart:
+ case eVoiceStatePlay:
+ activeVoices++;
+ playingVoices++;
+ poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->channel)].pool]++;
+ break;
+
+ case eVoiceStateRelease:
+ activeVoices++;
+ releasingVoices++;
+ poolCount[vSynthNum][pSynth->channels[GET_CHANNEL(pVoice->channel)].pool]++;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck : voice %d in invalid state\n", i); */ }
+ result = EAS_FAILURE;
+ break;
+ }
+ }
+
+ /* count free voices */
+ else
+ freeVoices++;
+ }
+
+ /* dump state info */
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d free\n", freeVoices); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d active\n", activeVoices); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d playing\n", playingVoices); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d releasing\n", releasingVoices); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d muting\n", mutingVoices); */ }
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "%d stolen\n", stolenVoices); */ }
+
+ if (pEASData->pVoiceMgr->activeVoices != activeVoices)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Active voice mismatch was %d should be %d\n",
+ pEASData->pVoiceMgr->activeVoices, activeVoices); */ }
+ result = EAS_FAILURE;
+ }
+
+ /* check virtual synth status */
+ for (i = 0; i < MAX_VIRTUAL_SYNTHESIZERS; i++)
+ {
+ if (pEASData->pVoiceMgr->pSynth[i] == NULL)
+ continue;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_DETAIL, "Synth %d numActiveVoices: %d\n", i, pEASData->pVoiceMgr->pSynth[i]->numActiveVoices); */ }
+ if (pEASData->pVoiceMgr->pSynth[i]->numActiveVoices > MAX_SYNTH_VOICES)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "VMSanityCheck: Synth %d illegal count for numActiveVoices: %d\n", i, pEASData->pVoiceMgr->pSynth[i]->numActiveVoices); */ }
+ result = EAS_FAILURE;
+ }
+ for (j = 0; j < NUM_SYNTH_CHANNELS; j++)
+ {
+ if (poolCount[i][j] != pEASData->pVoiceMgr->pSynth[i]->poolCount[j])
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Pool count mismatch synth %d pool %d, was %d, should be %d\n",
+ i, j, pEASData->pVoiceMgr->pSynth[i]->poolCount[j], poolCount[i][j]); */ }
+ result = EAS_FAILURE;
+ }
+ }
+ }
+
+ return result;
+}
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_wavefile.c b/arm-wt-22k/lib_src/eas_wavefile.c new file mode 100644 index 0000000..d3f3ba0 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wavefile.c @@ -0,0 +1,867 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wavefile.c
+ *
+ * Contents and purpose:
+ * This file implements the wave file parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 852 $
+ * $Date: 2007-09-04 11:43:49 -0700 (Tue, 04 Sep 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_config.h"
+#include "eas_parser.h"
+#include "eas_pcm.h"
+#include "eas_wavefile.h"
+
+/* lint is choking on the ARM math.h file, so we declare the log10 function here */
+extern double log10(double x);
+
+/* increase gain to compensate for loss in mixer */
+#define WAVE_GAIN_OFFSET 6
+
+/* constant for 1200 / log10(2.0) */
+#define PITCH_CENTS_CONVERSION 3986.313714
+
+/*----------------------------------------------------------------------------
+ * WAVE file defines
+ *----------------------------------------------------------------------------
+*/
+/* RIFF chunks */
+#define CHUNK_TYPE(a,b,c,d) ( \
+ ( ((EAS_U32)(a) & 0xFF) << 24 ) \
+ + ( ((EAS_U32)(b) & 0xFF) << 16 ) \
+ + ( ((EAS_U32)(c) & 0xFF) << 8 ) \
+ + ( ((EAS_U32)(d) & 0xFF) ) )
+
+#define CHUNK_RIFF CHUNK_TYPE('R','I','F','F')
+#define CHUNK_WAVE CHUNK_TYPE('W','A','V','E')
+#define CHUNK_FMT CHUNK_TYPE('f','m','t',' ')
+#define CHUNK_DATA CHUNK_TYPE('d','a','t','a')
+#define CHUNK_LIST CHUNK_TYPE('L','I','S','T')
+#define CHUNK_INFO CHUNK_TYPE('I','N','F','O')
+#define CHUNK_INAM CHUNK_TYPE('I','N','A','M')
+#define CHUNK_ICOP CHUNK_TYPE('I','C','O','P')
+#define CHUNK_IART CHUNK_TYPE('I','A','R','T')
+
+/* wave file format identifiers */
+#define WAVE_FORMAT_PCM 0x0001
+#define WAVE_FORMAT_IMA_ADPCM 0x0011
+
+/* file size for streamed file */
+#define FILE_SIZE_STREAMING 0x80000000
+
+/*----------------------------------------------------------------------------
+ * prototypes
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveCheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *pHandle, EAS_I32 offset);
+static EAS_RESULT WavePrepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT WaveState (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT WaveClose (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT WaveReset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT WaveLocate (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate);
+static EAS_RESULT WavePause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT WaveResume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT WaveSetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT WaveGetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT WaveParseHeader (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData);
+static EAS_RESULT WaveGetMetaData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength);
+
+#ifdef MMAPI_SUPPORT
+static EAS_RESULT SaveFmtChunk (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData, EAS_I32 size);
+#endif
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_Wave_Parser
+ *
+ * This structure contains the functional interface for the Wave file parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_Wave_Parser =
+{
+ WaveCheckFileType,
+ WavePrepare,
+ NULL,
+ NULL,
+ WaveState,
+ WaveClose,
+ WaveReset,
+ WavePause,
+ WaveResume,
+ WaveLocate,
+ WaveSetData,
+ WaveGetData,
+ WaveGetMetaData
+};
+
+/*----------------------------------------------------------------------------
+ * WaveCheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveCheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *pHandle, EAS_I32 offset)
+{
+ S_WAVE_STATE *pWaveData;
+
+ /* zero the memory to insure complete initialization */
+ *pHandle = NULL;
+
+ /* read the file header */
+ if (WaveParseHeader(pEASData, fileHandle, NULL) == EAS_SUCCESS)
+ {
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pWaveData = EAS_CMEnumData(EAS_CM_WAVE_DATA);
+ else
+ pWaveData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_WAVE_STATE));
+ if (!pWaveData)
+ return EAS_ERROR_MALLOC_FAILED;
+ EAS_HWMemSet(pWaveData, 0, sizeof(S_WAVE_STATE));
+
+ /* return a pointer to the instance data */
+ pWaveData->fileHandle = fileHandle;
+ pWaveData->fileOffset = offset;
+ *pHandle = pWaveData;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * WavePrepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WavePrepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_WAVE_STATE *pWaveData;
+ EAS_RESULT result;
+
+ /* validate parser state */
+ pWaveData = (S_WAVE_STATE*) pInstData;
+ if (pWaveData->streamHandle != NULL)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* back to start of file */
+ pWaveData->time = 0;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, pWaveData->fileOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* parse the file header */
+ if ((result = WaveParseHeader(pEASData, pWaveData->fileHandle, pWaveData)) != EAS_SUCCESS)
+ return result;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveState()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * This interface is also exposed in the internal library for use by the other modules.
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveState (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState)
+{
+ S_WAVE_STATE *pWaveData;
+
+ /* return current state */
+ pWaveData = (S_WAVE_STATE*) pInstData;
+ if (pWaveData->streamHandle)
+ return EAS_PEState(pEASData, pWaveData->streamHandle, pState);
+
+ /* if no stream handle, and time is not zero, we are done */
+ if (pWaveData->time > 0)
+ *pState = EAS_STATE_STOPPED;
+ else
+ *pState = EAS_STATE_OPEN;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveClose()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveClose (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_WAVE_STATE *pWaveData;
+ EAS_RESULT result;
+
+ pWaveData = (S_WAVE_STATE*) pInstData;
+
+ /* close the stream */
+ if (pWaveData->streamHandle)
+ {
+ if ((result = EAS_PEClose(pEASData, pWaveData->streamHandle)) != EAS_SUCCESS)
+ return result;
+ pWaveData->streamHandle = NULL;
+ }
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ {
+
+#ifdef MMAPI_SUPPORT
+ /* need to free the fmt chunk */
+ if (pWaveData->fmtChunk != NULL)
+ EAS_HWFree(pEASData->hwInstData, pWaveData->fmtChunk);
+#endif
+
+ /* free the instance data */
+ EAS_HWFree(pEASData->hwInstData, pWaveData);
+
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveReset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveReset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ EAS_PCM_HANDLE streamHandle;
+
+ /* reset to first byte of data in the stream */
+ streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle;
+ if (streamHandle)
+ return EAS_PEReset(pEASData, streamHandle);
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveLocate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Rewind/fast-forward in file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * time - time (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pParserLocate) reserved for future use */
+static EAS_RESULT WaveLocate (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 time, EAS_BOOL *pParserLocate)
+{
+ EAS_PCM_HANDLE streamHandle;
+
+ /* reset to first byte of data in the stream */
+ streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle;
+ if (streamHandle)
+ return EAS_PELocate(pEASData, streamHandle, time);
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+}
+
+/*----------------------------------------------------------------------------
+ * WavePause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mute and stop rendering a PCM stream. Sets the gain target to zero and stops the playback
+ * at the end of the next audio frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_WAVE_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT WavePause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ EAS_PCM_HANDLE streamHandle;
+
+ /* pause the stream */
+ streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle;
+ if (streamHandle)
+ return EAS_PEPause(pEASData, streamHandle);
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveResume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume rendering a PCM stream. Sets the gain target back to its
+ * previous setting and restarts playback at the end of the next audio
+ * frame.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_WAVE_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT WaveResume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ EAS_PCM_HANDLE streamHandle;
+
+ /* resume the stream */
+ streamHandle = ((S_WAVE_STATE*)pInstData)->streamHandle;
+ if (streamHandle)
+ return EAS_PEResume(pEASData, streamHandle);
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveSetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_WAVE_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveSetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ S_WAVE_STATE *pWaveData = (S_WAVE_STATE*) pInstData;
+
+ switch (param)
+ {
+ /* set metadata callback */
+ case PARSER_DATA_METADATA_CB:
+ EAS_HWMemCpy(&pWaveData->metadata, (void*) value, sizeof(S_METADATA_CB));
+ return EAS_SUCCESS;
+
+ case PARSER_DATA_PLAYBACK_RATE:
+ value = (EAS_I32) (PITCH_CENTS_CONVERSION * log10((double) value / (double) (1 << 28)));
+ return EAS_PEUpdatePitch(pEASData, pWaveData->streamHandle, (EAS_I16) value);
+
+ case PARSER_DATA_VOLUME:
+ return EAS_PEUpdateVolume(pEASData, pWaveData->streamHandle, (EAS_I16) value);
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * WaveGetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_WAVE_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pEASData) reserved for future use */
+static EAS_RESULT WaveGetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ S_WAVE_STATE *pWaveData;
+
+ pWaveData = (S_WAVE_STATE*) pInstData;
+ switch (param)
+ {
+ /* return file type as WAVE */
+ case PARSER_DATA_FILE_TYPE:
+ *pValue = pWaveData->fileType;
+ break;
+
+#ifdef MMAPI_SUPPORT
+ /* return pointer to 'fmt' chunk */
+ case PARSER_DATA_FORMAT:
+ *pValue = (EAS_I32) pWaveData->fmtChunk;
+ break;
+#endif
+
+ case PARSER_DATA_GAIN_OFFSET:
+ *pValue = WAVE_GAIN_OFFSET;
+ break;
+
+ default:
+ return EAS_ERROR_INVALID_PARAMETER;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * WaveParseHeader()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the WAVE file header.
+ *
+ * Inputs:
+ * pEASData - pointer to EAS library instance data
+ * handle - pointer to S_WAVE_STATE for this stream
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveParseHeader (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData)
+{
+ S_PCM_OPEN_PARAMS params;
+ EAS_RESULT result;
+ EAS_U32 tag;
+ EAS_U32 fileSize;
+ EAS_U32 size;
+ EAS_I32 pos;
+ EAS_I32 audioOffset;
+ EAS_U16 usTemp;
+ EAS_BOOL parseDone;
+ EAS_U32 avgBytesPerSec;
+
+ /* init some data (and keep lint happy) */
+ params.sampleRate = 0;
+ params.size = 0;
+ audioOffset = 0;
+ params.decoder = 0;
+ params.blockSize = 0;
+ params.pCallbackFunc = NULL;
+ params.cbInstData = NULL;
+ params.loopSamples = 0;
+ params.fileHandle = fileHandle;
+ params.volume = 0x7fff;
+ params.envData = 0;
+ avgBytesPerSec = 8000;
+
+ /* check for 'RIFF' tag */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE)
+ return result;
+ if (tag != CHUNK_RIFF)
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+
+ /* get size */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &fileSize, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* check for 'WAVE' tag */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE)
+ return result;
+ if (tag != CHUNK_WAVE)
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+
+ /* this is enough to say we recognize the file */
+ if (pWaveData == NULL)
+ return EAS_SUCCESS;
+
+ /* check for streaming mode */
+ pWaveData->flags = 0;
+ pWaveData->mediaLength = -1;
+ pWaveData->infoChunkPos = -1;
+ pWaveData->infoChunkSize = -1;
+ if (fileSize== FILE_SIZE_STREAMING)
+ {
+ pWaveData->flags |= PCM_FLAGS_STREAMING;
+ fileSize = 0x7fffffff;
+ }
+
+ /* find out where we're at */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &pos)) != EAS_SUCCESS)
+ return result;
+ fileSize -= 4;
+
+ parseDone = EAS_FALSE;
+ for (;;)
+ {
+ /* get tag and size for next chunk */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE)
+ return result;
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &size, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* process chunk */
+ pos += 8;
+ switch (tag)
+ {
+ case CHUNK_FMT:
+
+#ifdef MMAPI_SUPPORT
+ if ((result = SaveFmtChunk(pEASData, fileHandle, pWaveData, (EAS_I32) size)) != EAS_SUCCESS)
+ return result;
+#endif
+
+ /* get audio format */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE)
+ return result;
+ if (usTemp == WAVE_FORMAT_PCM)
+ {
+ params.decoder = EAS_DECODER_PCM;
+ pWaveData->fileType = EAS_FILE_WAVE_PCM;
+ }
+ else if (usTemp == WAVE_FORMAT_IMA_ADPCM)
+ {
+ params.decoder = EAS_DECODER_IMA_ADPCM;
+ pWaveData->fileType = EAS_FILE_WAVE_IMA_ADPCM;
+ }
+ else
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+
+ /* get number of channels */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE)
+ return result;
+ if (usTemp == 2)
+ pWaveData->flags |= PCM_FLAGS_STEREO;
+ else if (usTemp != 1)
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+
+ /* get sample rate */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, ¶ms.sampleRate, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* get stream rate */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &avgBytesPerSec, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* get block alignment */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE)
+ return result;
+ params.blockSize = usTemp;
+
+ /* get bits per sample */
+ if ((result = EAS_HWGetWord(pEASData->hwInstData, fileHandle, &usTemp, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* PCM, must be 8 or 16 bit samples */
+ if (params.decoder == EAS_DECODER_PCM)
+ {
+ if (usTemp == 8)
+ pWaveData->flags |= PCM_FLAGS_8_BIT | PCM_FLAGS_UNSIGNED;
+ else if (usTemp != 16)
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+ }
+
+ /* for IMA ADPCM, we only support mono 4-bit ADPCM */
+ else
+ {
+ if ((usTemp != 4) || (pWaveData->flags & PCM_FLAGS_STEREO))
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+ }
+
+ break;
+
+ case CHUNK_DATA:
+ audioOffset = pos;
+ if (pWaveData->flags & PCM_FLAGS_STREAMING)
+ {
+ params.size = 0x7fffffff;
+ parseDone = EAS_TRUE;
+ }
+ else
+ {
+ params.size = (EAS_I32) size;
+ params.loopStart = size;
+ /* use more accurate method if possible */
+ if (size <= (0x7fffffff / 1000))
+ pWaveData->mediaLength = (EAS_I32) ((size * 1000) / avgBytesPerSec);
+ else
+ pWaveData->mediaLength = (EAS_I32) (size / (avgBytesPerSec / 1000));
+ }
+ break;
+
+ case CHUNK_LIST:
+ /* get the list type */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &tag, EAS_TRUE)) != EAS_FALSE)
+ return result;
+ if (tag == CHUNK_INFO)
+ {
+ pWaveData->infoChunkPos = pos + 4;
+ pWaveData->infoChunkSize = (EAS_I32) size - 4;
+ }
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: %c%c%c%c chunk - %d byte(s) ignored\n",
+ (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ }
+ break;
+ }
+
+ if (parseDone)
+ break;
+
+ /* subtract header size */
+ fileSize -= 8;
+
+ /* account for zero-padding on odd length chunks */
+ if (size & 1)
+ size++;
+
+ /* this check works for files with odd length last chunk and no zero-pad */
+ if (size >= fileSize)
+ {
+ if (size > fileSize)
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: '%c%c%c%c' chunk size exceeds length of file or is not zero-padded\n",
+ (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ }
+ break;
+ }
+
+ /* subtract size of data chunk (including any zero-pad) */
+ fileSize -= size;
+
+ /* seek to next chunk */
+ pos += (EAS_I32) size;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, pos)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* check for valid header */
+ if ((params.sampleRate == 0) || (params.size == 0))
+ return EAS_ERROR_UNRECOGNIZED_FORMAT;
+
+ /* save the pertinent information */
+ pWaveData->audioOffset = audioOffset;
+ params.flags = pWaveData->flags;
+
+ /* seek to data */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, audioOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* open a stream in the PCM engine */
+ return EAS_PEOpenStream(pEASData, ¶ms, &pWaveData->streamHandle);
+}
+
+/*----------------------------------------------------------------------------
+ * WaveGetMetaData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Process the INFO chunk and return metadata to host
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT WaveGetMetaData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pMediaLength)
+{
+ S_WAVE_STATE *pWaveData;
+ EAS_RESULT result;
+ EAS_I32 pos;
+ EAS_U32 size;
+ EAS_I32 infoSize;
+ EAS_U32 tag;
+ EAS_I32 restorePos;
+ E_EAS_METADATA_TYPE metaType;
+ EAS_I32 metaLen;
+
+ /* get current position so we can restore it */
+ pWaveData = (S_WAVE_STATE*) pInstData;
+
+ /* return media length */
+ *pMediaLength = pWaveData->mediaLength;
+
+ /* did we encounter an INFO chunk? */
+ if (pWaveData->infoChunkPos < 0)
+ return EAS_SUCCESS;
+
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, pWaveData->fileHandle, &restorePos)) != EAS_SUCCESS)
+ return result;
+
+ /* offset to start of first chunk in INFO chunk */
+ pos = pWaveData->infoChunkPos;
+ infoSize = pWaveData->infoChunkSize;
+
+ /* read all the chunks in the INFO chunk */
+ for (;;)
+ {
+
+ /* seek to next chunk */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, pos)) != EAS_SUCCESS)
+ return result;
+
+ /* get tag and size for next chunk */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, pWaveData->fileHandle, &tag, EAS_TRUE)) != EAS_FALSE)
+ return result;
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, pWaveData->fileHandle, &size, EAS_FALSE)) != EAS_FALSE)
+ return result;
+
+ /* process chunk */
+ pos += 8;
+ metaType = EAS_METADATA_UNKNOWN;
+ switch (tag)
+ {
+ case CHUNK_INAM:
+ metaType = EAS_METADATA_TITLE;
+ break;
+
+ case CHUNK_IART:
+ metaType = EAS_METADATA_AUTHOR;
+ break;
+
+ case CHUNK_ICOP:
+ metaType = EAS_METADATA_COPYRIGHT;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WaveParseHeader: %c%c%c%c chunk - %d byte(s) ignored\n",
+ (char) (tag >> 24), (char) (tag >> 16), (char) (tag >> 8), (char) tag, size); */ }
+ break;
+ }
+
+ /* process known metadata */
+ if (metaType != EAS_METADATA_UNKNOWN)
+ {
+ metaLen = pWaveData->metadata.bufferSize - 1;
+ if (metaLen > (EAS_I32) size)
+ metaLen = (EAS_I32) size;
+ if ((result = EAS_HWReadFile(pEASData->hwInstData, pWaveData->fileHandle, pWaveData->metadata.buffer, metaLen, &metaLen)) != EAS_SUCCESS)
+ return result;
+ pWaveData->metadata.buffer[metaLen] = 0;
+ pWaveData->metadata.callback(metaType, pWaveData->metadata.buffer, pWaveData->metadata.pUserData);
+ }
+
+ /* subtract this block */
+ if (size & 1)
+ size++;
+ infoSize -= (EAS_I32) size + 8;
+ if (infoSize == 0)
+ break;
+ pos += (EAS_I32) size;
+ }
+
+
+ /* restore original position */
+ return EAS_HWFileSeek(pEASData->hwInstData, pWaveData->fileHandle, restorePos);
+}
+
+#ifdef MMAPI_SUPPORT
+/*----------------------------------------------------------------------------
+ * SaveFmtChunk()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Save the fmt chunk for the MMAPI library
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT SaveFmtChunk (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, S_WAVE_STATE *pWaveData, EAS_I32 fmtSize)
+{
+ EAS_RESULT result;
+ EAS_I32 pos;
+ EAS_I32 count;
+
+ /* save current file position */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &pos)) != EAS_SUCCESS)
+ return result;
+
+ /* allocate a chunk of memory */
+ pWaveData->fmtChunk = EAS_HWMalloc(pEASData->hwInstData, fmtSize);
+ if (!pWaveData->fmtChunk)
+ return EAS_ERROR_MALLOC_FAILED;
+
+ /* read the fmt chunk into memory */
+ if ((result = EAS_HWReadFile(pEASData->hwInstData, fileHandle, pWaveData->fmtChunk, fmtSize, &count)) != EAS_SUCCESS)
+ return result;
+ if (count != fmtSize)
+ return EAS_ERROR_FILE_READ_FAILED;
+
+ /* restore file position */
+ return EAS_HWFileSeek(pEASData->hwInstData, fileHandle, pos);
+}
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_wavefile.h b/arm-wt-22k/lib_src/eas_wavefile.h new file mode 100644 index 0000000..b8b76df --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wavefile.h @@ -0,0 +1,63 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wavefile.h
+ *
+ * Contents and purpose:
+ * Static data block for wave file parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 439 $
+ * $Date: 2006-10-26 11:53:18 -0700 (Thu, 26 Oct 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_WAVEFILE_H
+#define _EAS_WAVEFILE_H
+
+#include "eas_data.h"
+#include "eas_pcm.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * S_WAVE_STATE
+ *
+ * This structure contains the WAVE file parser state information
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wave_state_tag
+{
+ EAS_FILE_HANDLE fileHandle;
+ EAS_PCM_HANDLE streamHandle;
+ S_METADATA_CB metadata;
+ EAS_U32 time;
+ EAS_I32 fileOffset;
+ EAS_I32 audioOffset;
+ EAS_I32 mediaLength;
+ EAS_U32 audioSize;
+ EAS_U32 flags;
+ EAS_I16 fileType;
+#ifdef MMAPI_SUPPORT
+ EAS_VOID_PTR fmtChunk;
+#endif
+ EAS_I32 infoChunkPos;
+ EAS_I32 infoChunkSize;
+} S_WAVE_STATE;
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_wavefiledata.c b/arm-wt-22k/lib_src/eas_wavefiledata.c new file mode 100644 index 0000000..3742aa6 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wavefiledata.c @@ -0,0 +1,33 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wavefiledata.c
+ *
+ * Contents and purpose:
+ * Static data block for wave file parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_wavefile.h"
+
+S_WAVE_STATE eas_WaveData;
+
diff --git a/arm-wt-22k/lib_src/eas_wt_IPC_frame.h b/arm-wt-22k/lib_src/eas_wt_IPC_frame.h new file mode 100644 index 0000000..376fd3a --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wt_IPC_frame.h @@ -0,0 +1,82 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wt_IPC_frame.h
+ *
+ * Contents and purpose:
+ * This module contains data definitions for the interprocessor
+ * communications framework for a split-architecture synthesizer.
+ *
+ * This sample version writes IPC data to a file that can be used
+ * as a test vector for the DSP simulator. For a real-time system
+ * the file I/O is replaced with an IPC protocol in the hardware.
+ *
+ * Synchronization with the DSP is accomplished at the API level,
+ * i.e. the host code should call EAS_Render when it is ready to
+ * buffer another block of data for transmission to the DSP.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 818 $
+ * $Date: 2007-08-02 15:19:41 -0700 (Thu, 02 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_WT_IPC_FRAME_H
+#define _EAS_WT_IPC_FRAME_H
+
+/*----------------------------------------------------------------------------
+ * S_WT_FRAME
+ *
+ * This structure contains the common parameters that are updated
+ *for each frame of audio.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_frame_tag
+{
+ EAS_I32 gainTarget;
+ EAS_I32 phaseIncrement;
+
+#if defined(_FILTER_ENABLED)
+ EAS_I32 k;
+ EAS_I32 b1;
+ EAS_I32 b2;
+#endif
+} S_WT_FRAME;
+
+/*----------------------------------------------------------------------------
+ * S_WT_CONFIG
+ *
+ * This structure contains state data for the wavetable engine
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_config_tag
+{
+ EAS_U32 loopEnd; /* points to last PCM sample (not 1 beyond last) */
+ EAS_U32 loopStart; /* points to first sample at start of loop */
+ EAS_U32 phaseAccum; /* current sample, integer portion of phase */
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_I16 gainLeft; /* left channel gain */
+ EAS_I16 gainRight; /* right channel gain */
+#endif
+
+ EAS_I16 gain; /* current voice gain */
+} S_WT_CONFIG;
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/eas_wtengine.c b/arm-wt-22k/lib_src/eas_wtengine.c new file mode 100644 index 0000000..dd46f22 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wtengine.c @@ -0,0 +1,661 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_wtengine.c + * + * Contents and purpose: + * This file contains the critical synthesizer components that need to + * be optimized for best performance. + * + * Copyright Sonic Network Inc. 2004-2005 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 844 $ + * $Date: 2007-08-23 14:33:32 -0700 (Thu, 23 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +/*------------------------------------ + * includes + *------------------------------------ +*/ +#include "eas_types.h" +#include "eas_math.h" +#include "eas_audioconst.h" +#include "eas_sndlib.h" +#include "eas_wtengine.h" +#include "eas_mixer.h" + +/*---------------------------------------------------------------------------- + * prototypes + *---------------------------------------------------------------------------- +*/ +extern void WT_NoiseGenerator (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); +extern void WT_VoiceGain (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); + +#if defined(_OPTIMIZED_MONO) +extern void WT_InterpolateMono (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); +#else +extern void WT_InterpolateNoLoop (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); +extern void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame); +#endif + +#if defined(_FILTER_ENABLED) +extern void WT_VoiceFilter (S_FILTER_CONTROL*pFilter, S_WT_INT_FRAME *pWTIntFrame); +#endif + +#if defined(_OPTIMIZED_MONO) || !defined(NATIVE_EAS_KERNEL) +/*---------------------------------------------------------------------------- + * WT_VoiceGain + *---------------------------------------------------------------------------- + * Purpose: + * Output gain for individual voice + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pWTVoice) reserved for future use */ +void WT_VoiceGain (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + EAS_I32 *pMixBuffer; + EAS_PCM *pInputBuffer; + EAS_I32 gain; + EAS_I32 gainIncrement; + EAS_I32 tmp0; + EAS_I32 tmp1; + EAS_I32 tmp2; + EAS_I32 numSamples; + +#if (NUM_OUTPUT_CHANNELS == 2) + EAS_I32 gainLeft, gainRight; +#endif + + /* initialize some local variables */ + numSamples = pWTIntFrame->numSamples; + pMixBuffer = pWTIntFrame->pMixBuffer; + pInputBuffer = pWTIntFrame->pAudioBuffer; + + /*lint -e{703} <avoid multiply for performance>*/ + gainIncrement = (pWTIntFrame->frame.gainTarget - pWTIntFrame->prevGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); + if (gainIncrement < 0) + gainIncrement++; + /*lint -e{703} <avoid multiply for performance>*/ + gain = pWTIntFrame->prevGain << 16; + +#if (NUM_OUTPUT_CHANNELS == 2) + gainLeft = pWTVoice->gainLeft; + gainRight = pWTVoice->gainRight; +#endif + + while (numSamples--) { + + /* incremental gain step to prevent zipper noise */ + tmp0 = *pInputBuffer++; + gain += gainIncrement; + /*lint -e{704} <avoid divide>*/ + tmp2 = gain >> 16; + + /* scale sample by gain */ + tmp2 *= tmp0; + + + /* stereo output */ +#if (NUM_OUTPUT_CHANNELS == 2) + /*lint -e{704} <avoid divide>*/ + tmp2 = tmp2 >> 14; + + /* get the current sample in the final mix buffer */ + tmp1 = *pMixBuffer; + + /* left channel */ + tmp0 = tmp2 * gainLeft; + /*lint -e{704} <avoid divide>*/ + tmp0 = tmp0 >> NUM_MIXER_GUARD_BITS; + tmp1 += tmp0; + *pMixBuffer++ = tmp1; + + /* get the current sample in the final mix buffer */ + tmp1 = *pMixBuffer; + + /* right channel */ + tmp0 = tmp2 * gainRight; + /*lint -e{704} <avoid divide>*/ + tmp0 = tmp0 >> NUM_MIXER_GUARD_BITS; + tmp1 += tmp0; + *pMixBuffer++ = tmp1; + + /* mono output */ +#else + + /* get the current sample in the final mix buffer */ + tmp1 = *pMixBuffer; + /*lint -e{704} <avoid divide>*/ + tmp2 = tmp2 >> (NUM_MIXER_GUARD_BITS - 1); + tmp1 += tmp2; + *pMixBuffer++ = tmp1; +#endif + + } +} +#endif + +#ifndef NATIVE_EAS_KERNEL +/*---------------------------------------------------------------------------- + * WT_Interpolate + *---------------------------------------------------------------------------- + * Purpose: + * Interpolation engine for wavetable synth + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +void WT_Interpolate (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + EAS_PCM *pOutputBuffer; + EAS_I32 phaseInc; + EAS_I32 phaseFrac; + EAS_I32 acc0; + const EAS_SAMPLE *pSamples; + const EAS_SAMPLE *loopEnd; + EAS_I32 samp1; + EAS_I32 samp2; + EAS_I32 numSamples; + + /* initialize some local variables */ + numSamples = pWTIntFrame->numSamples; + pOutputBuffer = pWTIntFrame->pAudioBuffer; + + loopEnd = (const EAS_SAMPLE*) pWTVoice->loopEnd + 1; + pSamples = (const EAS_SAMPLE*) pWTVoice->phaseAccum; + /*lint -e{713} truncation is OK */ + phaseFrac = pWTVoice->phaseFrac; + phaseInc = pWTIntFrame->frame.phaseIncrement; + + /* fetch adjacent samples */ +#if defined(_8_BIT_SAMPLES) + /*lint -e{701} <avoid multiply for performance>*/ + samp1 = pSamples[0] << 8; + /*lint -e{701} <avoid multiply for performance>*/ + samp2 = pSamples[1] << 8; +#else + samp1 = pSamples[0]; + samp2 = pSamples[1]; +#endif + + while (numSamples--) { + + /* linear interpolation */ + acc0 = samp2 - samp1; + acc0 = acc0 * phaseFrac; + /*lint -e{704} <avoid divide>*/ + acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS); + + /* save new output sample in buffer */ + /*lint -e{704} <avoid divide>*/ + *pOutputBuffer++ = (EAS_I16)(acc0 >> 2); + + /* increment phase */ + phaseFrac += phaseInc; + /*lint -e{704} <avoid divide>*/ + acc0 = phaseFrac >> NUM_PHASE_FRAC_BITS; + + /* next sample */ + if (acc0 > 0) { + + /* advance sample pointer */ + pSamples += acc0; + phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK); + + /* check for loop end */ + acc0 = (EAS_I32) (pSamples - loopEnd); + if (acc0 >= 0) + pSamples = (const EAS_SAMPLE*) pWTVoice->loopStart + acc0; + + /* fetch new samples */ +#if defined(_8_BIT_SAMPLES) + /*lint -e{701} <avoid multiply for performance>*/ + samp1 = pSamples[0] << 8; + /*lint -e{701} <avoid multiply for performance>*/ + samp2 = pSamples[1] << 8; +#else + samp1 = pSamples[0]; + samp2 = pSamples[1]; +#endif + } + } + + /* save pointer and phase */ + pWTVoice->phaseAccum = (EAS_U32) pSamples; + pWTVoice->phaseFrac = (EAS_U32) phaseFrac; +} +#endif + +#ifndef NATIVE_EAS_KERNEL +/*---------------------------------------------------------------------------- + * WT_InterpolateNoLoop + *---------------------------------------------------------------------------- + * Purpose: + * Interpolation engine for wavetable synth + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +void WT_InterpolateNoLoop (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + EAS_PCM *pOutputBuffer; + EAS_I32 phaseInc; + EAS_I32 phaseFrac; + EAS_I32 acc0; + const EAS_SAMPLE *pSamples; + EAS_I32 samp1; + EAS_I32 samp2; + EAS_I32 numSamples; + + /* initialize some local variables */ + numSamples = pWTIntFrame->numSamples; + pOutputBuffer = pWTIntFrame->pAudioBuffer; + + phaseInc = pWTIntFrame->frame.phaseIncrement; + pSamples = (const EAS_SAMPLE*) pWTVoice->phaseAccum; + phaseFrac = (EAS_I32)pWTVoice->phaseFrac; + + /* fetch adjacent samples */ +#if defined(_8_BIT_SAMPLES) + /*lint -e{701} <avoid multiply for performance>*/ + samp1 = pSamples[0] << 8; + /*lint -e{701} <avoid multiply for performance>*/ + samp2 = pSamples[1] << 8; +#else + samp1 = pSamples[0]; + samp2 = pSamples[1]; +#endif + + while (numSamples--) { + + + /* linear interpolation */ + acc0 = samp2 - samp1; + acc0 = acc0 * phaseFrac; + /*lint -e{704} <avoid divide>*/ + acc0 = samp1 + (acc0 >> NUM_PHASE_FRAC_BITS); + + /* save new output sample in buffer */ + /*lint -e{704} <avoid divide>*/ + *pOutputBuffer++ = (EAS_I16)(acc0 >> 2); + + /* increment phase */ + phaseFrac += phaseInc; + /*lint -e{704} <avoid divide>*/ + acc0 = phaseFrac >> NUM_PHASE_FRAC_BITS; + + /* next sample */ + if (acc0 > 0) { + + /* advance sample pointer */ + pSamples += acc0; + phaseFrac = (EAS_I32)((EAS_U32)phaseFrac & PHASE_FRAC_MASK); + + /* fetch new samples */ +#if defined(_8_BIT_SAMPLES) + /*lint -e{701} <avoid multiply for performance>*/ + samp1 = pSamples[0] << 8; + /*lint -e{701} <avoid multiply for performance>*/ + samp2 = pSamples[1] << 8; +#else + samp1 = pSamples[0]; + samp2 = pSamples[1]; +#endif + } + } + + /* save pointer and phase */ + pWTVoice->phaseAccum = (EAS_U32) pSamples; + pWTVoice->phaseFrac = (EAS_U32) phaseFrac; +} +#endif + +#if defined(_FILTER_ENABLED) && !defined(NATIVE_EAS_KERNEL) +/*---------------------------------------------------------------------------- + * WT_VoiceFilter + *---------------------------------------------------------------------------- + * Purpose: + * Implements a 2-pole filter + * + * Inputs: + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +void WT_VoiceFilter (S_FILTER_CONTROL *pFilter, S_WT_INT_FRAME *pWTIntFrame) +{ + EAS_PCM *pAudioBuffer; + EAS_I32 k; + EAS_I32 b1; + EAS_I32 b2; + EAS_I32 z1; + EAS_I32 z2; + EAS_I32 acc0; + EAS_I32 acc1; + EAS_I32 numSamples; + + /* initialize some local variables */ + numSamples = pWTIntFrame->numSamples; + pAudioBuffer = pWTIntFrame->pAudioBuffer; + + z1 = pFilter->z1; + z2 = pFilter->z2; + b1 = -pWTIntFrame->frame.b1; + + /*lint -e{702} <avoid divide> */ + b2 = -pWTIntFrame->frame.b2 >> 1; + + /*lint -e{702} <avoid divide> */ + k = pWTIntFrame->frame.k >> 1; + + while (numSamples--) + { + + /* do filter calculations */ + acc0 = *pAudioBuffer; + acc1 = z1 * b1; + acc1 += z2 * b2; + acc0 = acc1 + k * acc0; + z2 = z1; + + /*lint -e{702} <avoid divide> */ + z1 = acc0 >> 14; + *pAudioBuffer++ = (EAS_I16) z1; + } + + /* save delay values */ + pFilter->z1 = (EAS_I16) z1; + pFilter->z2 = (EAS_I16) z2; +} +#endif + +/*---------------------------------------------------------------------------- + * WT_NoiseGenerator + *---------------------------------------------------------------------------- + * Purpose: + * Generate pseudo-white noise using PRNG and interpolation engine + * + * Inputs: + * + * Outputs: + * + * Notes: + * This output is scaled -12dB to prevent saturation in the filter. For a + * high quality synthesizer, the output can be set to full scale, however + * if the filter is used, it can overflow with certain coefficients. In this + * case, either a saturation operation should take in the filter before + * scaling back to 16 bits or the signal path should be increased to 18 bits + * or more. + *---------------------------------------------------------------------------- +*/ + void WT_NoiseGenerator (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) + { + EAS_PCM *pOutputBuffer; + EAS_I32 phaseInc; + EAS_I32 tmp0; + EAS_I32 tmp1; + EAS_I32 nInterpolatedSample; + EAS_I32 numSamples; + + /* initialize some local variables */ + numSamples = pWTIntFrame->numSamples; + pOutputBuffer = pWTIntFrame->pAudioBuffer; + phaseInc = pWTIntFrame->frame.phaseIncrement; + + /* get last two samples generated */ + /*lint -e{704} <avoid divide for performance>*/ + tmp0 = (EAS_I32) (pWTVoice->phaseAccum) >> 18; + /*lint -e{704} <avoid divide for performance>*/ + tmp1 = (EAS_I32) (pWTVoice->loopEnd) >> 18; + + /* generate a buffer of noise */ + while (numSamples--) { + nInterpolatedSample = MULT_AUDIO_COEF( tmp0, (PHASE_ONE - pWTVoice->phaseFrac)); + nInterpolatedSample += MULT_AUDIO_COEF( tmp1, pWTVoice->phaseFrac); + *pOutputBuffer++ = (EAS_PCM) nInterpolatedSample; + + /* update PRNG */ + pWTVoice->phaseFrac += (EAS_U32) phaseInc; + if (GET_PHASE_INT_PART(pWTVoice->phaseFrac)) { + tmp0 = tmp1; + pWTVoice->phaseAccum = pWTVoice->loopEnd; + pWTVoice->loopEnd = (5 * pWTVoice->loopEnd + 1); + tmp1 = (EAS_I32) (pWTVoice->loopEnd) >> 18; + pWTVoice->phaseFrac = GET_PHASE_FRAC_PART(pWTVoice->phaseFrac); + } + + } +} + +#ifndef _OPTIMIZED_MONO +/*---------------------------------------------------------------------------- + * WT_ProcessVoice + *---------------------------------------------------------------------------- + * Purpose: + * This routine does the block processing for one voice. It is isolated + * from the main synth code to allow for various implementation-specific + * optimizations. It calls the interpolator, filter, and gain routines + * appropriate for a particular configuration. + * + * Inputs: + * + * Outputs: + * + * Notes: + *---------------------------------------------------------------------------- +*/ +void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + + /* use noise generator */ + if (pWTVoice->loopStart == WT_NOISE_GENERATOR) + WT_NoiseGenerator(pWTVoice, pWTIntFrame); + + /* generate interpolated samples for looped waves */ + else if (pWTVoice->loopStart != pWTVoice->loopEnd) + WT_Interpolate(pWTVoice, pWTIntFrame); + + /* generate interpolated samples for unlooped waves */ + else + { + WT_InterpolateNoLoop(pWTVoice, pWTIntFrame); + } + +#ifdef _FILTER_ENABLED + if (pWTIntFrame->frame.k != 0) + WT_VoiceFilter(&pWTVoice->filter, pWTIntFrame); +#endif + +//2 TEST NEW MIXER FUNCTION +#ifdef UNIFIED_MIXER + { + EAS_I32 gainLeft, gainIncLeft; + +#if (NUM_OUTPUT_CHANNELS == 2) + EAS_I32 gainRight, gainIncRight; +#endif + + gainLeft = (pWTIntFrame->prevGain * pWTVoice->gainLeft) << 1; + gainIncLeft = (((pWTIntFrame->frame.gainTarget * pWTVoice->gainLeft) << 1) - gainLeft) >> SYNTH_UPDATE_PERIOD_IN_BITS; + +#if (NUM_OUTPUT_CHANNELS == 2) + gainRight = (pWTIntFrame->prevGain * pWTVoice->gainRight) << 1; + gainIncRight = (((pWTIntFrame->frame.gainTarget * pWTVoice->gainRight) << 1) - gainRight) >> SYNTH_UPDATE_PERIOD_IN_BITS; + EAS_MixStream( + pWTIntFrame->pAudioBuffer, + pWTIntFrame->pMixBuffer, + pWTIntFrame->numSamples, + gainLeft, + gainRight, + gainIncLeft, + gainIncRight, + MIX_FLAGS_STEREO_OUTPUT); + +#else + EAS_MixStream( + pWTIntFrame->pAudioBuffer, + pWTIntFrame->pMixBuffer, + pWTIntFrame->numSamples, + gainLeft, + 0, + gainIncLeft, + 0, + 0); +#endif + } + +#else + /* apply gain, and left and right gain */ + WT_VoiceGain(pWTVoice, pWTIntFrame); +#endif +} +#endif + +#if defined(_OPTIMIZED_MONO) && !defined(NATIVE_EAS_KERNEL) +/*---------------------------------------------------------------------------- + * WT_InterpolateMono + *---------------------------------------------------------------------------- + * Purpose: + * A C version of the sample interpolation + gain routine, optimized for mono. + * It's not pretty, but it matches the assembly code exactly. + * + * Inputs: + * + * Outputs: + * + * Notes: + *---------------------------------------------------------------------------- +*/ +void WT_InterpolateMono (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + EAS_I32 *pMixBuffer; + const EAS_I8 *pLoopEnd; + const EAS_I8 *pCurrentPhaseInt; + EAS_I32 numSamples; + EAS_I32 gain; + EAS_I32 gainIncrement; + EAS_I32 currentPhaseFrac; + EAS_I32 phaseInc; + EAS_I32 tmp0; + EAS_I32 tmp1; + EAS_I32 tmp2; + EAS_I8 *pLoopStart; + + numSamples = pWTIntFrame->numSamples; + pMixBuffer = pWTIntFrame->pMixBuffer; + + /* calculate gain increment */ + gainIncrement = (pWTIntFrame->gainTarget - pWTIntFrame->prevGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS); + if (gainIncrement < 0) + gainIncrement++; + gain = pWTIntFrame->prevGain << 16; + + pCurrentPhaseInt = pWTVoice->pPhaseAccum; + currentPhaseFrac = pWTVoice->phaseFrac; + phaseInc = pWTIntFrame->phaseIncrement; + + pLoopStart = pWTVoice->pLoopStart; + pLoopEnd = pWTVoice->pLoopEnd + 1; + +InterpolationLoop: + tmp0 = (EAS_I32)(pCurrentPhaseInt - pLoopEnd); + if (tmp0 >= 0) + pCurrentPhaseInt = pLoopStart + tmp0; + + tmp0 = *pCurrentPhaseInt; + tmp1 = *(pCurrentPhaseInt + 1); + + tmp2 = phaseInc + currentPhaseFrac; + + tmp1 = tmp1 - tmp0; + tmp1 = tmp1 * currentPhaseFrac; + + tmp1 = tmp0 + (tmp1 >> NUM_EG1_FRAC_BITS); + + pCurrentPhaseInt += (tmp2 >> NUM_PHASE_FRAC_BITS); + currentPhaseFrac = tmp2 & PHASE_FRAC_MASK; + + gain += gainIncrement; + tmp2 = (gain >> SYNTH_UPDATE_PERIOD_IN_BITS); + + tmp0 = *pMixBuffer; + tmp2 = tmp1 * tmp2; + tmp2 = (tmp2 >> 9); + tmp0 = tmp2 + tmp0; + *pMixBuffer++ = tmp0; + + numSamples--; + if (numSamples > 0) + goto InterpolationLoop; + + pWTVoice->pPhaseAccum = pCurrentPhaseInt; + pWTVoice->phaseFrac = currentPhaseFrac; + /*lint -e{702} <avoid divide>*/ + pWTVoice->gain = (EAS_I16)(gain >> SYNTH_UPDATE_PERIOD_IN_BITS); +} +#endif + +#ifdef _OPTIMIZED_MONO +/*---------------------------------------------------------------------------- + * WT_ProcessVoice + *---------------------------------------------------------------------------- + * Purpose: + * This routine does the block processing for one voice. It is isolated + * from the main synth code to allow for various implementation-specific + * optimizations. It calls the interpolator, filter, and gain routines + * appropriate for a particular configuration. + * + * Inputs: + * + * Outputs: + * + * Notes: + * This special version works handles an optimized mono-only signal + * without filters + *---------------------------------------------------------------------------- +*/ +void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame) +{ + + /* use noise generator */ + if (pWTVoice->loopStart== WT_NOISE_GENERATOR) + { + WT_NoiseGenerator(pWTVoice, pWTIntFrame); + WT_VoiceGain(pWTVoice, pWTIntFrame); + } + + /* or generate interpolated samples */ + else + { + WT_InterpolateMono(pWTVoice, pWTIntFrame); + } +} +#endif + diff --git a/arm-wt-22k/lib_src/eas_wtengine.h b/arm-wt-22k/lib_src/eas_wtengine.h new file mode 100644 index 0000000..6401322 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wtengine.h @@ -0,0 +1,171 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wtengine.h
+ *
+ * Contents and purpose:
+ * This file defines the interface for wavetable synthesis engine
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 818 $
+ * $Date: 2007-08-02 15:19:41 -0700 (Thu, 02 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_WTENGINE_H
+#define _EAS_WTENGINE_H
+
+/* option sanity check */
+#if defined(_OPTIMIZED_MONO) && defined(_FILTER_ENABLED)
+#error "Incompatible build settings: _OPTIMIZED_MONO cannot be used with _FILTER_ENABLED"
+#endif
+
+#if defined(_OPTIMIZED_MONO) && (NUM_OUTPUT_CHANNELS != 1)
+#error "Incompatible build settings: _OPTIMIZED_MONO can only be used with NUM_OUTPUT_CHANNELS = 1"
+#endif
+
+#include "eas_wt_IPC_frame.h"
+
+/*----------------------------------------------------------------------------
+ * defines
+ *----------------------------------------------------------------------------
+*/
+#define WT_NOISE_GENERATOR 0xffffffff
+
+/*----------------------------------------------------------------------------
+ * typedefs
+ *----------------------------------------------------------------------------
+*/
+
+/*----------------------------------------------------------------------------
+ * S_WT_INT_FRAME
+ *
+ * This structure includes S_WT_FRAME plus the bus mixing
+ * parameters for the internal voices.
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_int_frame_tag
+{
+ S_WT_FRAME frame;
+ EAS_PCM *pAudioBuffer;
+ EAS_I32 *pMixBuffer;
+ EAS_I32 numSamples;
+ EAS_I32 prevGain;
+} S_WT_INT_FRAME;
+
+#if defined(_FILTER_ENABLED)
+/*----------------------------------------------------------------------------
+ * S_FILTER_CONTROL data structure
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_filter_control_tag
+{
+ EAS_I16 z1; /* 1 sample delay state variable */
+ EAS_I16 z2; /* 2 sample delay state variable */
+} S_FILTER_CONTROL;
+#endif
+
+/*------------------------------------
+ * S_LFO_CONTROL data structure
+ *------------------------------------
+*/
+typedef struct s_lfo_control_tag
+{
+ EAS_I16 lfoValue; /* LFO current output value */
+ EAS_I16 lfoPhase; /* LFO current phase */
+} S_LFO_CONTROL;
+
+/* bit definitions for S_WT_VOICE:flags */
+#define WT_FLAGS_ADPCM_NIBBLE 1 /* high/low nibble flag */
+#define WT_FLAGS_ADPCM_READY 2 /* first 2 samples are decoded */
+#define WT_FLAGS_USE_ADPCM 4 /* sample is ADPCM encoded */
+
+/* eg1State and eg2State */
+typedef enum {
+ eEnvelopeStateInit = 0,
+ eEnvelopeStateDelay,
+ eEnvelopeStateAttack,
+ eEnvelopeStateHold,
+ eEnvelopeStateDecay,
+ eEnvelopeStateSustain,
+ eEnvelopeStateRelease,
+ eEnvelopeStateMuting,
+ eEnvelopeStateMuted,
+ eEnvelopeStateInvalid /* should never be in this state! */
+} E_ENVELOPE_STATE;
+
+#define DEFAULT_EG1_STATE eEnvelopeStateAttack
+#define DEFAULT_EG1_VALUE 0
+#define DEFAULT_EG1_INCREMENT 0
+#define DEFAULT_EG2_STATE eEnvelopeStateAttack
+#define DEFAULT_EG2_VALUE 0
+#define DEFAULT_EG2_INCREMENT 0
+
+/*----------------------------------------------------------------------------
+ * S_WT_VOICE
+ *
+ * This structure contains state data for the wavetable engine
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_wt_voice_tag
+{
+ EAS_U32 loopEnd; /* points to last PCM sample (not 1 beyond last) */
+ EAS_U32 loopStart; /* points to first sample at start of loop */
+ EAS_U32 phaseAccum; /* current sample, integer portion of phase */
+ EAS_U32 phaseFrac; /* fractional portion of phase */
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_I16 gainLeft; /* current gain, left ch */
+ EAS_I16 gainRight; /* current gain, right ch */
+#endif
+
+#if defined(_FILTER_ENABLED)
+ S_FILTER_CONTROL filter; /* low pass filter */
+#endif
+
+ S_LFO_CONTROL modLFO; /* modulator LFO */
+
+#ifdef DLS_SYNTHESIZER
+ S_LFO_CONTROL vibLFO; /* vibrato LFO */
+#endif
+
+ /* envelope control */
+ EAS_I16 eg1Value;
+ EAS_I16 eg2Value;
+ EAS_I16 eg1Increment;
+ EAS_I16 eg2Increment;
+ EAS_U8 eg1State;
+ EAS_U8 eg2State;
+
+ EAS_U16 artIndex; /* index to articulation params */
+
+} S_WT_VOICE;
+
+/*----------------------------------------------------------------------------
+ * prototypes
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update);
+void WT_ProcessVoice (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame);
+
+#ifdef EAS_SPLIT_WT_SYNTH
+void WTE_ConfigVoice (EAS_I32 voiceNum, S_WT_CONFIG *pWTConfig, EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+void WTE_ProcessVoice (EAS_I32 voiceNum, S_WT_FRAME *pWTParams, EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+#endif
+
+#endif
diff --git a/arm-wt-22k/lib_src/eas_wtsynth.c b/arm-wt-22k/lib_src/eas_wtsynth.c new file mode 100644 index 0000000..3eadf2d --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wtsynth.c @@ -0,0 +1,1257 @@ +/*---------------------------------------------------------------------------- + * + * File: + * eas_wtsynth.c + * + * Contents and purpose: + * Implements the synthesizer functions. + * + * Copyright Sonic Network Inc. 2004 + + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + * + *---------------------------------------------------------------------------- + * Revision Control: + * $Revision: 795 $ + * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $ + *---------------------------------------------------------------------------- +*/ + +// includes +#include "eas_data.h" +#include "eas_report.h" +#include "eas_host.h" +#include "eas_math.h" +#include "eas_synth_protos.h" +#include "eas_wtsynth.h" +#include "eas_pan.h" + +#ifdef DLS_SYNTHESIZER +#include "eas_dlssynth.h" +#endif + +#ifdef _METRICS_ENABLED +#include "eas_perf.h" +#endif + +/* local prototypes */ +static EAS_RESULT WT_Initialize(S_VOICE_MGR *pVoiceMgr); +static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); +static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum); +static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum); +static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex); +static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples); +static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel); +static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents); +static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain); +static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv); +static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv); + +#ifdef EAS_SPLIT_WT_SYNTH +extern EAS_BOOL WTE_StartFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer); +extern EAS_BOOL WTE_EndFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer, EAS_I32 *pMixBuffer, EAS_I16 masterGain); +#endif + +#ifdef _FILTER_ENABLED +static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt); +#endif + +#ifdef _STATS +extern double statsPhaseIncrement; +extern double statsMaxPhaseIncrement; +extern long statsPhaseSampleCount; +extern double statsSampleSize; +extern long statsSampleCount; +#endif + +/*---------------------------------------------------------------------------- + * Synthesizer interface + *---------------------------------------------------------------------------- +*/ + +const S_SYNTH_INTERFACE wtSynth = +{ + WT_Initialize, + WT_StartVoice, + WT_UpdateVoice, + WT_ReleaseVoice, + WT_MuteVoice, + WT_SustainPedal, + WT_UpdateChannel +}; + +#ifdef EAS_SPLIT_WT_SYNTH +const S_FRAME_INTERFACE wtFrameInterface = +{ + WTE_StartFrame, + WTE_EndFrame +}; +#endif + +/*---------------------------------------------------------------------------- + * WT_Initialize() + *---------------------------------------------------------------------------- + * Purpose: + * + * Inputs: + * pVoice - pointer to voice to initialize + * + * Outputs: + * + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT WT_Initialize (S_VOICE_MGR *pVoiceMgr) +{ + EAS_INT i; + + for (i = 0; i < NUM_WT_VOICES; i++) + { + + pVoiceMgr->wtVoices[i].artIndex = DEFAULT_ARTICULATION_INDEX; + + pVoiceMgr->wtVoices[i].eg1State = DEFAULT_EG1_STATE; + pVoiceMgr->wtVoices[i].eg1Value = DEFAULT_EG1_VALUE; + pVoiceMgr->wtVoices[i].eg1Increment = DEFAULT_EG1_INCREMENT; + + pVoiceMgr->wtVoices[i].eg2State = DEFAULT_EG2_STATE; + pVoiceMgr->wtVoices[i].eg2Value = DEFAULT_EG2_VALUE; + pVoiceMgr->wtVoices[i].eg2Increment = DEFAULT_EG2_INCREMENT; + + /* left and right gain values are needed only if stereo output */ +#if (NUM_OUTPUT_CHANNELS == 2) + pVoiceMgr->wtVoices[i].gainLeft = DEFAULT_VOICE_GAIN; + pVoiceMgr->wtVoices[i].gainRight = DEFAULT_VOICE_GAIN; +#endif + + pVoiceMgr->wtVoices[i].phaseFrac = DEFAULT_PHASE_FRAC; + pVoiceMgr->wtVoices[i].phaseAccum = DEFAULT_PHASE_INT; + +#ifdef _FILTER_ENABLED + pVoiceMgr->wtVoices[i].filter.z1 = DEFAULT_FILTER_ZERO; + pVoiceMgr->wtVoices[i].filter.z2 = DEFAULT_FILTER_ZERO; +#endif + } + + return EAS_TRUE; +} + +/*---------------------------------------------------------------------------- + * WT_ReleaseVoice() + *---------------------------------------------------------------------------- + * Purpose: + * The selected voice is being released. + * + * Inputs: + * pEASData - pointer to S_EAS_DATA + * pVoice - pointer to voice to release + * + * Outputs: + * None + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pVoice) used in some implementations */ +static void WT_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) +{ + S_WT_VOICE *pWTVoice; + const S_ARTICULATION *pArticulation; + +#ifdef DLS_SYNTHESIZER + if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) + { + DLS_ReleaseVoice(pVoiceMgr, pSynth, pVoice, voiceNum); + return; + } +#endif + + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pArticulation = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; + + /* release EG1 */ + pWTVoice->eg1State = eEnvelopeStateRelease; + pWTVoice->eg1Increment = pArticulation->eg1.releaseTime; + + /* + The spec says we should release EG2, but doing so with the current + voicing is causing clicks. This fix will need to be coordinated with + a new sound library release + */ + + /* release EG2 */ + pWTVoice->eg2State = eEnvelopeStateRelease; + pWTVoice->eg2Increment = pArticulation->eg2.releaseTime; +} + +/*---------------------------------------------------------------------------- + * WT_MuteVoice() + *---------------------------------------------------------------------------- + * Purpose: + * The selected voice is being muted. + * + * Inputs: + * pVoice - pointer to voice to release + * + * Outputs: + * None + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pSynth) used in some implementations */ +static void WT_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum) +{ + +#ifdef DLS_SYNTHESIZER + if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) + { + DLS_MuteVoice(pVoiceMgr, pSynth, pVoice, voiceNum); + return; + } +#endif + + /* clear deferred action flags */ + pVoice->voiceFlags &= + ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF | + VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF | + VOICE_FLAG_DEFER_MUTE); + + /* set the envelope state */ + pVoiceMgr->wtVoices[voiceNum].eg1State = eEnvelopeStateMuted; + pVoiceMgr->wtVoices[voiceNum].eg2State = eEnvelopeStateMuted; +} + +/*---------------------------------------------------------------------------- + * WT_SustainPedal() + *---------------------------------------------------------------------------- + * Purpose: + * The selected voice is held due to sustain pedal + * + * Inputs: + * pVoice - pointer to voice to sustain + * + * Outputs: + * None + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pChannel) used in some implementations */ +static void WT_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum) +{ + S_WT_VOICE *pWTVoice; + +#ifdef DLS_SYNTHESIZER + if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) + { + DLS_SustainPedal(pVoiceMgr, pSynth, pVoice, pChannel, voiceNum); + return; + } +#endif + + /* don't catch the voice if below the sustain level */ + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + if (pWTVoice->eg1Value < pSynth->pEAS->pArticulations[pWTVoice->artIndex].eg1.sustainLevel) + return; + + /* sustain flag is set, damper pedal is on */ + /* defer releasing this note until the damper pedal is off */ + pWTVoice->eg1State = eEnvelopeStateDecay; + pVoice->voiceState = eVoiceStatePlay; + + /* + because sustain pedal is on, this voice + should defer releasing its note + */ + pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF; + +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_SustainPedal: defer note off because sustain pedal is on\n"); */ } +#endif +} + +/*---------------------------------------------------------------------------- + * WT_StartVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Assign the region for the given instrument using the midi key number + * and the RPN2 (coarse tuning) value. By using RPN2 as part of the + * region selection process, we reduce the amount a given sample has + * to be transposed by selecting the closest recorded root instead. + * + * This routine is the second half of SynthAssignRegion(). + * If the region was successfully found by SynthFindRegionIndex(), + * then assign the region's parameters to the voice. + * + * Setup and initialize the following voice parameters: + * m_nRegionIndex + * + * Inputs: + * pVoice - ptr to the voice we have assigned for this channel + * nRegionIndex - index of the region + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * success - could find and assign the region for this voice's note otherwise + * failure - could not find nor assign the region for this voice's note + * + * Side Effects: + * psSynthObject->m_sVoice[].m_nRegionIndex is assigned + * psSynthObject->m_sVoice[] parameters are assigned + *---------------------------------------------------------------------------- +*/ +static EAS_RESULT WT_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex) +{ + S_WT_VOICE *pWTVoice; + const S_WT_REGION *pRegion; + const S_ARTICULATION *pArt; + S_SYNTH_CHANNEL *pChannel; + +#if (NUM_OUTPUT_CHANNELS == 2) + EAS_INT pan; +#endif + +#ifdef EAS_SPLIT_WT_SYNTH + S_WT_CONFIG wtConfig; +#endif + + /* no samples have been synthesized for this note yet */ + pVoice->regionIndex = regionIndex; + pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; + + /* get the articulation index for this region */ + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pChannel = &pSynth->channels[pVoice->channel & 15]; + + /* update static channel parameters */ + if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS) + WT_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15); + +#ifdef DLS_SYNTHESIZER + if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) + return DLS_StartVoice(pVoiceMgr, pSynth, pVoice, voiceNum, regionIndex); +#endif + + pRegion = &(pSynth->pEAS->pWTRegions[regionIndex]); + pWTVoice->artIndex = pRegion->artIndex; + +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_StartVoice: Voice %ld; Region %d\n", (EAS_I32) (pVoice - pVoiceMgr->voices), regionIndex); */ } +#endif + + pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; + + /* MIDI note on puts this voice into attack state */ + pWTVoice->eg1State = eEnvelopeStateAttack; + pWTVoice->eg1Value = 0; + pWTVoice->eg1Increment = pArt->eg1.attackTime; + pWTVoice->eg2State = eEnvelopeStateAttack; + pWTVoice->eg2Value = 0; + pWTVoice->eg2Increment = pArt->eg2.attackTime; + + /* init the LFO */ + pWTVoice->modLFO.lfoValue = 0; + pWTVoice->modLFO.lfoPhase = -pArt->lfoDelay; + + pVoice->gain = 0; + +#if (NUM_OUTPUT_CHANNELS == 2) + /* + Get the Midi CC10 pan value for this voice's channel + convert the pan value to an "angle" representation suitable for + our sin, cos calculator. This representation is NOT necessarily the same + as the transform in the GM manuals because of our sin, cos calculator. + "angle" = (CC10 - 64)/128 + */ + pan = (EAS_INT) pSynth->channels[pVoice->channel & 15].pan - 64; + pan += pArt->pan; + EAS_CalcPanControl(pan, &pWTVoice->gainLeft, &pWTVoice->gainRight); +#endif + +#ifdef _FILTER_ENABLED + /* clear out the filter states */ + pWTVoice->filter.z1 = 0; + pWTVoice->filter.z2 = 0; +#endif + + /* if this wave is to be generated using noise generator */ + if (pRegion->region.keyGroupAndFlags & REGION_FLAG_USE_WAVE_GENERATOR) + { + pWTVoice->phaseAccum = 4574296; + pWTVoice->loopStart = WT_NOISE_GENERATOR; + pWTVoice->loopEnd = 4574295; + } + + /* normal sample */ + else + { + +#ifdef EAS_SPLIT_WT_SYNTH + if (voiceNum < NUM_PRIMARY_VOICES) + pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; + else + pWTVoice->phaseAccum = pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; +#else + pWTVoice->phaseAccum = (EAS_U32) pSynth->pEAS->pSamples + pSynth->pEAS->pSampleOffsets[pRegion->waveIndex]; +#endif + + if (pRegion->region.keyGroupAndFlags & REGION_FLAG_IS_LOOPED) + { + pWTVoice->loopStart = pWTVoice->phaseAccum + pRegion->loopStart; + pWTVoice->loopEnd = pWTVoice->phaseAccum + pRegion->loopEnd - 1; + } + else + pWTVoice->loopStart = pWTVoice->loopEnd = pWTVoice->phaseAccum + pSynth->pEAS->pSampleLen[pRegion->waveIndex] - 1; + } + +#ifdef EAS_SPLIT_WT_SYNTH + /* configure off-chip voices */ + if (voiceNum >= NUM_PRIMARY_VOICES) + { + wtConfig.phaseAccum = pWTVoice->phaseAccum; + wtConfig.loopStart = pWTVoice->loopStart; + wtConfig.loopEnd = pWTVoice->loopEnd; + wtConfig.gain = pVoice->gain; + +#if (NUM_OUTPUT_CHANNELS == 2) + wtConfig.gainLeft = pWTVoice->gainLeft; + wtConfig.gainRight = pWTVoice->gainRight; +#endif + + WTE_ConfigVoice(voiceNum - NUM_PRIMARY_VOICES, &wtConfig, pVoiceMgr->pFrameBuffer); + } +#endif + + return EAS_SUCCESS; +} + +/*---------------------------------------------------------------------------- + * WT_CheckSampleEnd + *---------------------------------------------------------------------------- + * Purpose: + * Check for end of sample and calculate number of samples to synthesize + * + * Inputs: + * + * Outputs: + * + * Notes: + * + *---------------------------------------------------------------------------- +*/ +EAS_BOOL WT_CheckSampleEnd (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pWTIntFrame, EAS_BOOL update) +{ + EAS_U32 endPhaseAccum; + EAS_U32 endPhaseFrac; + EAS_I32 numSamples; + EAS_BOOL done = EAS_FALSE; + + /* check to see if we hit the end of the waveform this time */ + /*lint -e{703} use shift for performance */ + endPhaseFrac = pWTVoice->phaseFrac + (pWTIntFrame->frame.phaseIncrement << SYNTH_UPDATE_PERIOD_IN_BITS); + endPhaseAccum = pWTVoice->phaseAccum + GET_PHASE_INT_PART(endPhaseFrac); + if (endPhaseAccum >= pWTVoice->loopEnd) + { + /* calculate how far current ptr is from end */ + numSamples = (EAS_I32) (pWTVoice->loopEnd - pWTVoice->phaseAccum); + + /* now account for the fractional portion */ + /*lint -e{703} use shift for performance */ + numSamples = (EAS_I32) ((numSamples << NUM_PHASE_FRAC_BITS) - pWTVoice->phaseFrac); + if (pWTIntFrame->frame.phaseIncrement) { + pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement); + } else { + pWTIntFrame->numSamples = numSamples; + } + + /* sound will be done this frame */ + done = EAS_TRUE; + } + + /* update data for off-chip synth */ + if (update) + { + pWTVoice->phaseFrac = endPhaseFrac; + pWTVoice->phaseAccum = endPhaseAccum; + } + + return done; +} + +/*---------------------------------------------------------------------------- + * WT_UpdateVoice() + *---------------------------------------------------------------------------- + * Purpose: + * Synthesize a block of samples for the given voice. + * Use linear interpolation. + * + * Inputs: + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * number of samples actually written to buffer + * + * Side Effects: + * - samples are added to the presently free buffer + * + *---------------------------------------------------------------------------- +*/ +static EAS_BOOL WT_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples) +{ + S_WT_VOICE *pWTVoice; + S_WT_INT_FRAME intFrame; + S_SYNTH_CHANNEL *pChannel; + const S_WT_REGION *pWTRegion; + const S_ARTICULATION *pArt; + EAS_I32 temp; + EAS_BOOL done; + +#ifdef DLS_SYNTHESIZER + if (pVoice->regionIndex & FLAG_RGN_IDX_DLS_SYNTH) + return DLS_UpdateVoice(pVoiceMgr, pSynth, pVoice, voiceNum, pMixBuffer, numSamples); +#endif + + /* establish pointers to critical data */ + pWTVoice = &pVoiceMgr->wtVoices[voiceNum]; + pWTRegion = &pSynth->pEAS->pWTRegions[pVoice->regionIndex & REGION_INDEX_MASK]; + pArt = &pSynth->pEAS->pArticulations[pWTVoice->artIndex]; + pChannel = &pSynth->channels[pVoice->channel & 15]; + intFrame.prevGain = pVoice->gain; + + /* update the envelopes */ + WT_UpdateEG1(pWTVoice, &pArt->eg1); + WT_UpdateEG2(pWTVoice, &pArt->eg2); + + /* update the LFO */ + WT_UpdateLFO(&pWTVoice->modLFO, pArt->lfoFreq); + +#ifdef _FILTER_ENABLED + /* calculate filter if library uses filter */ + if (pSynth->pEAS->libAttr & LIB_FORMAT_FILTER_ENABLED) + WT_UpdateFilter(pWTVoice, &intFrame, pArt); + else + intFrame.frame.k = 0; +#endif + + /* update the gain */ + intFrame.frame.gainTarget = WT_UpdateGain(pVoice, pWTVoice, pArt, pChannel, pWTRegion->gain); + + /* calculate base pitch*/ + temp = pChannel->staticPitch + pWTRegion->tuning; + + /* include global transpose */ + if (pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL) + temp += pVoice->note * 100; + else + temp += (pVoice->note + pSynth->globalTranspose) * 100; + intFrame.frame.phaseIncrement = WT_UpdatePhaseInc(pWTVoice, pArt, pChannel, temp); + + /* call into engine to generate samples */ + intFrame.pAudioBuffer = pVoiceMgr->voiceBuffer; + intFrame.pMixBuffer = pMixBuffer; + intFrame.numSamples = numSamples; + + /* check for end of sample */ + if ((pWTVoice->loopStart != WT_NOISE_GENERATOR) && (pWTVoice->loopStart == pWTVoice->loopEnd)) + done = WT_CheckSampleEnd(pWTVoice, &intFrame, (EAS_BOOL) (voiceNum >= NUM_PRIMARY_VOICES)); + else + done = EAS_FALSE; + + if (intFrame.numSamples < 0) intFrame.numSamples = 0; + +#ifdef EAS_SPLIT_WT_SYNTH + if (voiceNum < NUM_PRIMARY_VOICES) + { +#ifndef _SPLIT_WT_TEST_HARNESS + WT_ProcessVoice(pWTVoice, &intFrame); +#endif + } + else + WTE_ProcessVoice(voiceNum - NUM_PRIMARY_VOICES, &intFrame.frame, pVoiceMgr->pFrameBuffer); +#else + WT_ProcessVoice(pWTVoice, &intFrame); +#endif + + /* clear flag */ + pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET; + + /* if voice has finished, set flag for voice manager */ + if ((pVoice->voiceState != eVoiceStateStolen) && (pWTVoice->eg1State == eEnvelopeStateMuted)) + done = EAS_TRUE; + + /* if the update interval has elapsed, then force the current gain to the next + * gain since we never actually reach the next gain when ramping -- we just get + * very close to the target gain. + */ + pVoice->gain = (EAS_I16) intFrame.frame.gainTarget; + + return done; +} + +/*---------------------------------------------------------------------------- + * WT_UpdatePhaseInc() + *---------------------------------------------------------------------------- + * Purpose: + * Calculate the phase increment + * + * Inputs: + * pVoice - pointer to the voice being updated + * psRegion - pointer to the region + * psArticulation - pointer to the articulation + * nChannelPitchForThisVoice - the portion of the pitch that is fixed for this + * voice during the duration of this synthesis + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * set the phase increment for this voice + *---------------------------------------------------------------------------- +*/ +static EAS_I32 WT_UpdatePhaseInc (S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 pitchCents) +{ + EAS_I32 temp; + + /*pitchCents due to CC1 = LFO * (CC1 / 128) * DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS */ + temp = MULT_EG1_EG1(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS, + ((pChannel->modWheel) << (NUM_EG1_FRAC_BITS -7))); + + /* pitchCents due to channel pressure = LFO * (channel pressure / 128) * DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS */ + temp += MULT_EG1_EG1(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS, + ((pChannel->channelPressure) << (NUM_EG1_FRAC_BITS -7))); + + /* now multiply the (channel pressure + CC1) pitch values by the LFO value */ + temp = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, temp); + + /* + add in the LFO pitch due to + channel pressure and CC1 along with + the LFO pitch, the EG2 pitch, and the + "static" pitch for this voice on this channel + */ + temp += pitchCents + + (MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToPitch)) + + (MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToPitch)); + + /* convert from cents to linear phase increment */ + return EAS_Calculate2toX(temp); +} + +/*---------------------------------------------------------------------------- + * WT_UpdateChannel() + *---------------------------------------------------------------------------- + * Purpose: + * Calculate and assign static channel parameters + * These values only need to be updated if one of the controller values + * for this channel changes + * + * Inputs: + * nChannel - channel to update + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * - the given channel's static gain and static pitch are updated + *---------------------------------------------------------------------------- +*/ +/*lint -esym(715, pVoiceMgr) reserved for future use */ +static void WT_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel) +{ + EAS_I32 staticGain; + EAS_I32 pitchBend; + S_SYNTH_CHANNEL *pChannel; + + pChannel = &pSynth->channels[channel]; + + /* + nChannelGain = (CC7 * CC11)^2 * master volume + where CC7 == 100 by default, CC11 == 127, master volume == 32767 + */ + staticGain = MULT_EG1_EG1((pChannel->volume) << (NUM_EG1_FRAC_BITS - 7), + (pChannel->expression) << (NUM_EG1_FRAC_BITS - 7)); + + /* staticGain has to be squared */ + staticGain = MULT_EG1_EG1(staticGain, staticGain); + + pChannel->staticGain = (EAS_I16) MULT_EG1_EG1(staticGain, pSynth->masterVolume); + + /* + calculate pitch bend: RPN0 * ((2*pitch wheel)/16384 -1) + However, if we use the EG1 macros, remember that EG1 has a full + scale value of 32768 (instead of 16384). So instead of multiplying + by 2, multiply by 4 (left shift by 2), and subtract by 32768 instead + of 16384. This utilizes the fact that the EG1 macro places a binary + point 15 places to the left instead of 14 places. + */ + /*lint -e{703} <avoid multiply for performance>*/ + pitchBend = + (((EAS_I32)(pChannel->pitchBend) << 2) + - 32768); + + pChannel->staticPitch = + MULT_EG1_EG1(pitchBend, pChannel->pitchBendSensitivity); + + /* if this is not a drum channel, then add in the per-channel tuning */ + if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL)) + pChannel->staticPitch += pChannel->finePitch + (pChannel->coarsePitch * 100); + + /* clear update flag */ + pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS; + return; +} + +/*---------------------------------------------------------------------------- + * WT_UpdateGain() + *---------------------------------------------------------------------------- + * Purpose: + * Calculate and assign static voice parameters as part of WT_UpdateVoice() + * + * Inputs: + * pVoice - ptr to the synth voice that we want to synthesize + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * - various voice parameters are calculated and assigned + * + *---------------------------------------------------------------------------- +*/ +static EAS_I32 WT_UpdateGain (S_SYNTH_VOICE *pVoice, S_WT_VOICE *pWTVoice, const S_ARTICULATION *pArt, S_SYNTH_CHANNEL *pChannel, EAS_I32 gain) +{ + EAS_I32 lfoGain; + EAS_I32 temp; + + /* + If this voice was stolen, then the velocity is actually + for the new note, not the note that we are currently ramping down. + So we really shouldn't use this velocity. However, that would require + more memory to store the velocity value, and the improvement may + not be sufficient to warrant the added memory. + */ + /* velocity is fixed at note start for a given voice and must be squared */ + temp = (pVoice->velocity) << (NUM_EG1_FRAC_BITS - 7); + temp = MULT_EG1_EG1(temp, temp); + + /* region gain is fixed as part of the articulation */ + temp = MULT_EG1_EG1(temp, gain); + + /* include the channel gain */ + temp = MULT_EG1_EG1(temp, pChannel->staticGain); + + /* calculate LFO gain using an approximation for 10^x */ + lfoGain = MULT_EG1_EG1(pWTVoice->modLFO.lfoValue, pArt->lfoToGain); + lfoGain = MULT_EG1_EG1(lfoGain, LFO_GAIN_TO_CENTS); + + /* convert from a dB-like value to linear gain */ + lfoGain = EAS_Calculate2toX(lfoGain); + temp = MULT_EG1_EG1(temp, lfoGain); + + /* calculate the voice's gain */ + temp = (EAS_I16)MULT_EG1_EG1(temp, pWTVoice->eg1Value); + + return temp; +} + +/*---------------------------------------------------------------------------- + * WT_UpdateEG1() + *---------------------------------------------------------------------------- + * Purpose: + * Calculate the EG1 envelope for the given voice (but do not update any + * state) + * + * Inputs: + * pVoice - ptr to the voice whose envelope we want to update + * nVoice - this voice's number - used only for debug + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * nValue - the envelope value + * + * Side Effects: + * - updates EG1 state value for the given voice + *---------------------------------------------------------------------------- +*/ +static void WT_UpdateEG1 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv) +{ + EAS_I32 temp; + + switch (pWTVoice->eg1State) + { + case eEnvelopeStateAttack: + temp = pWTVoice->eg1Value + pWTVoice->eg1Increment; + + /* check if we have reached peak amplitude */ + if (temp >= SYNTH_FULL_SCALE_EG1_GAIN) + { + /* limit the volume */ + temp = SYNTH_FULL_SCALE_EG1_GAIN; + + /* prepare to move to decay state */ + pWTVoice->eg1State = eEnvelopeStateDecay; + pWTVoice->eg1Increment = pEnv->decayTime; + } + + break; + + /* exponential decay */ + case eEnvelopeStateDecay: + temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment); + + /* check if we have reached sustain level */ + if (temp <= pEnv->sustainLevel) + { + /* enforce the sustain level */ + temp = pEnv->sustainLevel; + + /* if sustain level is zero, skip sustain & release the voice */ + if (temp > 0) + pWTVoice->eg1State = eEnvelopeStateSustain; + + /* move to sustain state */ + else + pWTVoice->eg1State = eEnvelopeStateMuted; + } + + break; + + case eEnvelopeStateSustain: + return; + + case eEnvelopeStateRelease: + temp = MULT_EG1_EG1(pWTVoice->eg1Value, pWTVoice->eg1Increment); + + /* if we hit zero, this voice isn't contributing any audio */ + if (temp <= 0) + { + temp = 0; + pWTVoice->eg1State = eEnvelopeStateMuted; + } + break; + + /* voice is muted, set target to zero */ + case eEnvelopeStateMuted: + temp = 0; + break; + + case eEnvelopeStateInvalid: + default: + temp = 0; +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG1: error, %d is an unrecognized state\n", + pWTVoice->eg1State); */ } +#endif + break; + + } + + pWTVoice->eg1Value = (EAS_I16) temp; +} + +/*---------------------------------------------------------------------------- + * WT_UpdateEG2() + *---------------------------------------------------------------------------- + * Purpose: + * Update the EG2 envelope for the given voice + * + * Inputs: + * pVoice - ptr to the voice whose envelope we want to update + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * - updates EG2 values for the given voice + *---------------------------------------------------------------------------- +*/ + +static void WT_UpdateEG2 (S_WT_VOICE *pWTVoice, const S_ENVELOPE *pEnv) +{ + EAS_I32 temp; + + switch (pWTVoice->eg2State) + { + case eEnvelopeStateAttack: + temp = pWTVoice->eg2Value + pWTVoice->eg2Increment; + + /* check if we have reached peak amplitude */ + if (temp >= SYNTH_FULL_SCALE_EG1_GAIN) + { + /* limit the volume */ + temp = SYNTH_FULL_SCALE_EG1_GAIN; + + /* prepare to move to decay state */ + pWTVoice->eg2State = eEnvelopeStateDecay; + + pWTVoice->eg2Increment = pEnv->decayTime; + } + + break; + + /* implement linear pitch decay in cents */ + case eEnvelopeStateDecay: + temp = pWTVoice->eg2Value -pWTVoice->eg2Increment; + + /* check if we have reached sustain level */ + if (temp <= pEnv->sustainLevel) + { + /* enforce the sustain level */ + temp = pEnv->sustainLevel; + + /* prepare to move to sustain state */ + pWTVoice->eg2State = eEnvelopeStateSustain; + } + break; + + case eEnvelopeStateSustain: + return; + + case eEnvelopeStateRelease: + temp = pWTVoice->eg2Value - pWTVoice->eg2Increment; + + if (temp <= 0) + { + temp = 0; + pWTVoice->eg2State = eEnvelopeStateMuted; + } + + break; + + /* voice is muted, set target to zero */ + case eEnvelopeStateMuted: + temp = 0; + break; + + case eEnvelopeStateInvalid: + default: + temp = 0; +#ifdef _DEBUG_SYNTH + { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "WT_UpdateEG2: error, %d is an unrecognized state\n", + pWTVoice->eg2State); */ } +#endif + break; + } + + pWTVoice->eg2Value = (EAS_I16) temp; +} + +/*---------------------------------------------------------------------------- + * WT_UpdateLFO () + *---------------------------------------------------------------------------- + * Purpose: + * Calculate the LFO for the given voice + * + * Inputs: + * pLFO - ptr to the LFO data + * phaseInc - phase increment + * + * Outputs: + * + * Side Effects: + * - updates LFO values for the given voice + *---------------------------------------------------------------------------- +*/ +void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc) +{ + + /* To save memory, if m_nPhaseValue is negative, we are in the + * delay phase, and m_nPhaseValue represents the time left + * in the delay. + */ + if (pLFO->lfoPhase < 0) + { + pLFO->lfoPhase++; + return; + } + + /* calculate LFO output from phase value */ + /*lint -e{701} Use shift for performance */ + pLFO->lfoValue = (EAS_I16) (pLFO->lfoPhase << 2); + /*lint -e{502} <shortcut to turn sawtooth into triangle wave> */ + if ((pLFO->lfoPhase > 0x1fff) && (pLFO->lfoPhase < 0x6000)) + pLFO->lfoValue = ~pLFO->lfoValue; + + /* update LFO phase */ + pLFO->lfoPhase = (pLFO->lfoPhase + phaseInc) & 0x7fff; +} + +#ifdef _FILTER_ENABLED +/*---------------------------------------------------------------------------- + * WT_UpdateFilter() + *---------------------------------------------------------------------------- + * Purpose: + * Update the Filter parameters + * + * Inputs: + * pVoice - ptr to the voice whose filter we want to update + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * - updates Filter values for the given voice + *---------------------------------------------------------------------------- +*/ +static void WT_UpdateFilter (S_WT_VOICE *pWTVoice, S_WT_INT_FRAME *pIntFrame, const S_ARTICULATION *pArt) +{ + EAS_I32 cutoff; + + /* no need to calculate filter coefficients if it is bypassed */ + if (pArt->filterCutoff == DEFAULT_EAS_FILTER_CUTOFF_FREQUENCY) + { + pIntFrame->frame.k = 0; + return; + } + + /* determine the dynamic cutoff frequency */ + cutoff = MULT_EG1_EG1(pWTVoice->eg2Value, pArt->eg2ToFc); + cutoff += pArt->filterCutoff; + + /* subtract the A5 offset and the sampling frequency */ + cutoff -= FILTER_CUTOFF_FREQ_ADJUST + A5_PITCH_OFFSET_IN_CENTS; + + /* limit the cutoff frequency */ + if (cutoff > FILTER_CUTOFF_MAX_PITCH_CENTS) + cutoff = FILTER_CUTOFF_MAX_PITCH_CENTS; + else if (cutoff < FILTER_CUTOFF_MIN_PITCH_CENTS) + cutoff = FILTER_CUTOFF_MIN_PITCH_CENTS; + + WT_SetFilterCoeffs(pIntFrame, cutoff, pArt->filterQ); +} +#endif + +#if defined(_FILTER_ENABLED) || defined(DLS_SYNTHESIZER) +/*---------------------------------------------------------------------------- + * coef + *---------------------------------------------------------------------------- + * Table of filter coefficients for low-pass filter + *---------------------------------------------------------------------------- + * + * polynomial coefficients are based on 8kHz sampling frequency + * filter coef b2 = k2 = k2g0*k^0 + k2g1*k^1*(2^x) + k2g2*k^2*(2^x) + * + *where k2g0, k2g1, k2g2 are from the truncated power series expansion on theta + *(k*2^x = theta, but we incorporate the k along with the k2g0, k2g1, k2g2) + *note: this is a power series in 2^x, not k*2^x + *where k = (2*pi*440)/8kHz == convert octaves to radians + * + * so actually, the following coefs listed as k2g0, k2g1, k2g2 are really + * k2g0*k^0 = k2g0 + * k2g1*k^1 + * k2g2*k^2 + * + * + * filter coef n1 = numerator = n1g0*k^0 + n1g1*k^1*(2^x) + n1g2*k^2*(2^x) + n1g3*k^3*(2^x) + * + *where n1g0, n1g1, n1g2, n1g3 are from the truncated power series expansion on theta + *(k*2^x = theta, but we incorporate the k along with the n1g0, n1g1, n1g2, n2g3) + *note: this is a power series in 2^x, not k*2^x + *where k = (2*pi*440)/8kHz == convert octaves to radians + *we also include the optimization factor of 0.81 + * + * so actually, the following coefs listed as n1g0, n1g1, n1g2, n2g3 are really + * n1g0*k^0 = n1g0 + * n1g1*k^1 + * n1g2*k^2 + * n1g3*k^3 + * + * NOTE that n1g0 == n1g1 == 0, always, so we only need to store n1g2 and n1g3 + *---------------------------------------------------------------------------- +*/ + +static const EAS_I16 nk1g0 = -32768; +static const EAS_I16 nk1g2 = 1580; +static const EAS_I16 k2g0 = 32767; + +static const EAS_I16 k2g1[] = +{ + -11324, /* k2g1[0] = -0.3455751918948761 */ + -10387, /* k2g1[1] = -0.3169878073928751 */ + -9528, /* k2g1[2] = -0.29076528753345476 */ + -8740, /* k2g1[3] = -0.2667120011011279 */ + -8017, /* k2g1[4] = -0.24464850028971705 */ + -7353, /* k2g1[5] = -0.22441018194495696 */ + -6745, /* k2g1[6] = -0.20584605955455101 */ + -6187, /* k2g1[7] = -0.18881763682420102 */ + -5675, /* k2g1[8] = -0.1731978744360067 */ + -5206, /* k2g1[9] = -0.15887024228080968 */ + -4775, /* k2g1[10] = -0.14572785009373057 */ + -4380, /* k2g1[11] = -0.13367265000706827 */ + -4018, /* k2g1[12] = -0.1226147050712642 */ + -3685, /* k2g1[13] = -0.11247151828678581 */ + -3381, /* k2g1[14] = -0.10316741714122014 */ + -3101, /* k2g1[15] = -0.0946329890599603 */ + -2844, /* k2g1[16] = -0.08680456355870586 */ + -2609, /* k2g1[17] = -0.07962373723441349 */ + -2393, /* k2g1[18] = -0.07303693805092666 */ + -2195, /* k2g1[19] = -0.06699502566866912 */ + -2014, /* k2g1[20] = -0.06145292483669077 */ + -1847, /* k2g1[21] = -0.056369289112013346 */ + -1694, /* k2g1[22] = -0.05170619239747895 */ + -1554, /* k2g1[23] = -0.04742884599684141 */ + -1426, /* k2g1[24] = -0.043505339076210514 */ + -1308, /* k2g1[25] = -0.03990640059558053 */ + -1199, /* k2g1[26] = -0.03660518093435039 */ + -1100, /* k2g1[27] = -0.03357705158166837 */ + -1009, /* k2g1[28] = -0.030799421397205727 */ + -926, /* k2g1[29] = -0.028251568071585884 */ + -849 /* k2g1[30] = -0.025914483529091967 */ +}; + +static const EAS_I16 k2g2[] = +{ + 1957, /* k2g2[0] = 0.059711106626580836 */ + 1646, /* k2g2[1] = 0.05024063501786333 */ + 1385, /* k2g2[2] = 0.042272226217199664 */ + 1165, /* k2g2[3] = 0.03556764576567844 */ + 981, /* k2g2[4] = 0.029926444346999134 */ + 825, /* k2g2[5] = 0.025179964880280382 */ + 694, /* k2g2[6] = 0.02118630011706455 */ + 584, /* k2g2[7] = 0.01782604998793514 */ + 491, /* k2g2[8] = 0.014998751854573014 */ + 414, /* k2g2[9] = 0.012619876941179595 */ + 348, /* k2g2[10] = 0.010618303146468736 */ + 293, /* k2g2[11] = 0.008934188679954682 */ + 246, /* k2g2[12] = 0.007517182949855368 */ + 207, /* k2g2[13] = 0.006324921212866403 */ + 174, /* k2g2[14] = 0.005321757979794424 */ + 147, /* k2g2[15] = 0.004477701309210577 */ + 123, /* k2g2[16] = 0.00376751612730811 */ + 104, /* k2g2[17] = 0.0031699697655869644 */ + 87, /* k2g2[18] = 0.00266719715992703 */ + 74, /* k2g2[19] = 0.0022441667321724647 */ + 62, /* k2g2[20] = 0.0018882309854916855 */ + 52, /* k2g2[21] = 0.0015887483774966232 */ + 44, /* k2g2[22] = 0.0013367651661223448 */ + 37, /* k2g2[23] = 0.0011247477162958733 */ + 31, /* k2g2[24] = 0.0009463572640678758 */ + 26, /* k2g2[25] = 0.0007962604042473498 */ + 22, /* k2g2[26] = 0.0006699696356181593 */ + 18, /* k2g2[27] = 0.0005637091964589207 */ + 16, /* k2g2[28] = 0.00047430217920125243 */ + 13, /* k2g2[29] = 0.00039907554925166274 */ + 11 /* k2g2[30] = 0.00033578022828973666 */ +}; + +static const EAS_I16 n1g2[] = +{ + 3170, /* n1g2[0] = 0.0967319927350769 */ + 3036, /* n1g2[1] = 0.0926446051254155 */ + 2908, /* n1g2[2] = 0.08872992911818503 */ + 2785, /* n1g2[3] = 0.08498066682523227 */ + 2667, /* n1g2[4] = 0.08138982872895201 */ + 2554, /* n1g2[5] = 0.07795072065216213 */ + 2446, /* n1g2[6] = 0.0746569312785634 */ + 2343, /* n1g2[7] = 0.07150232020051943 */ + 2244, /* n1g2[8] = 0.06848100647187474 */ + 2149, /* n1g2[9] = 0.06558735764447099 */ + 2058, /* n1g2[10] = 0.06281597926792246 */ + 1971, /* n1g2[11] = 0.06016170483307614 */ + 1888, /* n1g2[12] = 0.05761958614040857 */ + 1808, /* n1g2[13] = 0.05518488407540374 */ + 1732, /* n1g2[14] = 0.052853059773715245 */ + 1659, /* n1g2[15] = 0.05061976615964251 */ + 1589, /* n1g2[16] = 0.04848083984214659 */ + 1521, /* n1g2[17] = 0.046432293353298 */ + 1457, /* n1g2[18] = 0.04447030771468711 */ + 1396, /* n1g2[19] = 0.04259122531793907 */ + 1337, /* n1g2[20] = 0.040791543106060944 */ + 1280, /* n1g2[21] = 0.03906790604290942 */ + 1226, /* n1g2[22] = 0.037417100858604564 */ + 1174, /* n1g2[23] = 0.035836050059229754 */ + 1125, /* n1g2[24] = 0.03432180618965023 */ + 1077, /* n1g2[25] = 0.03287154633875494 */ + 1032, /* n1g2[26] = 0.03148256687687814 */ + 988, /* n1g2[27] = 0.030152278415589925 */ + 946, /* n1g2[28] = 0.028878200980459685 */ + 906, /* n1g2[29] = 0.02765795938779331 */ + 868 /* n1g2[30] = 0.02648927881672521 */ +}; + +static const EAS_I16 n1g3[] = +{ + -548, /* n1g3[0] = -0.016714088475899017 */ + -481, /* n1g3[1] = -0.014683605122742116 */ + -423, /* n1g3[2] = -0.012899791676436092 */ + -371, /* n1g3[3] = -0.01133268185193299 */ + -326, /* n1g3[4] = -0.00995594976868754 */ + -287, /* n1g3[5] = -0.008746467702146129 */ + -252, /* n1g3[6] = -0.00768391756106361 */ + -221, /* n1g3[7] = -0.006750449563854721 */ + -194, /* n1g3[8] = -0.005930382380083576 */ + -171, /* n1g3[9] = -0.005209939699767622 */ + -150, /* n1g3[10] = -0.004577018805123356 */ + -132, /* n1g3[11] = -0.004020987256990177 */ + -116, /* n1g3[12] = -0.003532504280467257 */ + -102, /* n1g3[13] = -0.00310336384922047 */ + -89, /* n1g3[14] = -0.002726356832432369 */ + -78, /* n1g3[15] = -0.002395149888601605 */ + -69, /* n1g3[16] = -0.0021041790717285314 */ + -61, /* n1g3[17] = -0.0018485563625771063 */ + -53, /* n1g3[18] = -0.001623987554831628 */ + -47, /* n1g3[19] = -0.0014267001167177025 */ + -41, /* n1g3[20] = -0.0012533798162347005 */ + -36, /* n1g3[21] = -0.0011011150453668693 */ + -32, /* n1g3[22] = -0.0009673479079754438 */ + -28, /* n1g3[23] = -0.0008498312496971563 */ + -24, /* n1g3[24] = -0.0007465909079943587 */ + -21, /* n1g3[25] = -0.0006558925481952733 */ + -19, /* n1g3[26] = -0.0005762125284029567 */ + -17, /* n1g3[27] = -0.0005062123038325457 */ + -15, /* n1g3[28] = -0.0004447159405951901 */ + -13, /* n1g3[29] = -0.00039069036118270117 */ + -11 /* n1g3[30] = -0.00034322798979677605 */ +}; + +/*---------------------------------------------------------------------------- + * WT_SetFilterCoeffs() + *---------------------------------------------------------------------------- + * Purpose: + * Update the Filter parameters + * + * Inputs: + * pVoice - ptr to the voice whose filter we want to update + * pEASData - pointer to overall EAS data structure + * + * Outputs: + * + * Side Effects: + * - updates Filter values for the given voice + *---------------------------------------------------------------------------- +*/ +void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance) +{ + EAS_I32 temp; + + /* + Convert the cutoff, which has had A5 subtracted, using the 2^x approx + Note, this cutoff is related to theta cutoff by + theta = k * 2^x + We use 2^x and incorporate k in the power series coefs instead + */ + cutoff = EAS_Calculate2toX(cutoff); + + /* calculate b2 coef */ + temp = k2g1[resonance] + MULT_AUDIO_COEF(cutoff, k2g2[resonance]); + temp = k2g0 + MULT_AUDIO_COEF(cutoff, temp); + pIntFrame->frame.b2 = temp; + + /* calculate b1 coef */ + temp = MULT_AUDIO_COEF(cutoff, nk1g2); + temp = nk1g0 + MULT_AUDIO_COEF(cutoff, temp); + temp += MULT_AUDIO_COEF(temp, pIntFrame->frame.b2); + pIntFrame->frame.b1 = temp >> 1; + + /* calculate K coef */ + temp = n1g2[resonance] + MULT_AUDIO_COEF(cutoff, n1g3[resonance]); + temp = MULT_AUDIO_COEF(cutoff, temp); + temp = MULT_AUDIO_COEF(cutoff, temp); + pIntFrame->frame.k = temp; +} +#endif + diff --git a/arm-wt-22k/lib_src/eas_wtsynth.h b/arm-wt-22k/lib_src/eas_wtsynth.h new file mode 100644 index 0000000..106ab40 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_wtsynth.h @@ -0,0 +1,66 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_wtsynth.h
+ *
+ * Contents and purpose:
+ * This file defines the interface for synthesizer engine
+ *
+ * Copyright Sonic Network Inc. 2004
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_WTSYNTH_H
+#define _EAS_WTSYNTH_H
+
+#include "eas_sndlib.h"
+#include "eas_wtengine.h"
+
+/* adjust the filter cutoff frequency to the sample rate */
+#if defined (_SAMPLE_RATE_8000)
+#define FILTER_CUTOFF_FREQ_ADJUST 0
+#elif defined (_SAMPLE_RATE_16000)
+#define FILTER_CUTOFF_FREQ_ADJUST 1200
+#elif defined (_SAMPLE_RATE_20000)
+#define FILTER_CUTOFF_FREQ_ADJUST 1586
+#elif defined (_SAMPLE_RATE_22050)
+#define FILTER_CUTOFF_FREQ_ADJUST 1756
+#elif defined (_SAMPLE_RATE_24000)
+#define FILTER_CUTOFF_FREQ_ADJUST 1902
+#elif defined (_SAMPLE_RATE_32000)
+#define FILTER_CUTOFF_FREQ_ADJUST 2400
+#elif defined (_SAMPLE_RATE_44100)
+#define FILTER_CUTOFF_FREQ_ADJUST 2956
+#elif defined (_SAMPLE_RATE_48000)
+#define FILTER_CUTOFF_FREQ_ADJUST 3102
+#else
+#error "_SAMPLE_RATE_XXXXX must be defined to valid rate"
+#endif
+
+/* function prototypes */
+void WT_UpdateLFO (S_LFO_CONTROL *pLFO, EAS_I16 phaseInc);
+
+#if defined(_FILTER_ENABLED) || defined(DLS_SYNTHESIZER)
+void WT_SetFilterCoeffs (S_WT_INT_FRAME *pIntFrame, EAS_I32 cutoff, EAS_I32 resonance);
+#endif
+
+#endif
+
+
diff --git a/arm-wt-22k/lib_src/eas_xmf.c b/arm-wt-22k/lib_src/eas_xmf.c new file mode 100644 index 0000000..b498b42 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_xmf.c @@ -0,0 +1,850 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_xmf.c
+ * 5
+ * Contents and purpose:
+ * XMF File Parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 501 $
+ * $Date: 2006-12-11 17:53:36 -0800 (Mon, 11 Dec 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_data.h"
+#include "eas_miditypes.h"
+#include "eas_parser.h"
+#include "eas_report.h"
+#include "eas_host.h"
+#include "eas_midi.h"
+#include "eas_xmf.h"
+#include "eas_xmfdata.h"
+#include "eas_config.h"
+#include "eas_vm_protos.h"
+#include "eas_mdls.h"
+#include "eas_smf.h"
+
+
+/* XMF header file type */
+#define XMF_IDENTIFIER 0x584d465f
+#define XMF_VERSION_2_00 0x322e3030
+#define XMF_FILE_TYPE 0x00000002
+#define XMF_SPEC_LEVEL 0x00000001
+#define XMF_RIFF_CHUNK 0x52494646
+#define XMF_RIFF_DLS 0x444c5320
+#define XMF_SMF_CHUNK 0x4d546864
+
+/* local prototypes */
+static EAS_RESULT XMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset);
+static EAS_RESULT XMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT XMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime);
+static EAS_RESULT XMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode);
+static EAS_RESULT XMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_STATE *pState);
+static EAS_RESULT XMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT XMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT XMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT XMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData);
+static EAS_RESULT XMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value);
+static EAS_RESULT XMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue);
+static EAS_RESULT XMF_FindFileContents (EAS_HW_DATA_HANDLE hwInstData, S_XMF_DATA *pXMFData);
+static EAS_RESULT XMF_ReadNode (EAS_HW_DATA_HANDLE hwInstData, S_XMF_DATA *pXMFData, EAS_I32 nodeOffset, EAS_I32 *pLength);
+static EAS_RESULT XMF_ReadVLQ (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 *value);
+
+
+/*----------------------------------------------------------------------------
+ *
+ * XMF_Parser
+ *
+ * This structure contains the functional interface for the XMF parser
+ *----------------------------------------------------------------------------
+*/
+const S_FILE_PARSER_INTERFACE EAS_XMF_Parser =
+{
+ XMF_CheckFileType,
+ XMF_Prepare,
+ XMF_Time,
+ XMF_Event,
+ XMF_State,
+ XMF_Close,
+ XMF_Reset,
+ XMF_Pause,
+ XMF_Resume,
+ NULL,
+ XMF_SetData,
+ XMF_GetData,
+ NULL
+};
+
+/*----------------------------------------------------------------------------
+ * XMF_CheckFileType()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Check the file type to see if we can parse it
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_CheckFileType (S_EAS_DATA *pEASData, EAS_FILE_HANDLE fileHandle, EAS_VOID_PTR *ppHandle, EAS_I32 offset)
+{
+ S_XMF_DATA *pXMFData;
+ EAS_RESULT result;
+ EAS_U32 temp;
+
+ /* assume we don't recognize it initially */
+ *ppHandle = NULL;
+
+ /* read the file identifier */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+ if (temp != XMF_IDENTIFIER)
+ return EAS_SUCCESS;
+
+ /* read the version */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+ if (temp != XMF_VERSION_2_00)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "XMF file version was 0x%08x, expected 0x%08x\n", temp, XMF_VERSION_2_00); */ }
+ return EAS_SUCCESS;
+ }
+
+ /* read the file type */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+ if (temp != XMF_FILE_TYPE)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "XMF file type was 0x%08x, expected 0x%08x\n", temp, XMF_FILE_TYPE); */ }
+ return EAS_SUCCESS;
+ }
+
+ /* read the spec level */
+ if ((result = EAS_HWGetDWord(pEASData->hwInstData, fileHandle, &temp, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+ if (temp != XMF_SPEC_LEVEL)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "XMF file spec was 0x%08x, expected 0x%08x\n", temp, XMF_SPEC_LEVEL); */ }
+ return EAS_SUCCESS;
+ }
+
+ /* check for static memory allocation */
+ if (pEASData->staticMemoryModel)
+ pXMFData = EAS_CMEnumData(EAS_CM_XMF_DATA);
+ else
+ pXMFData = EAS_HWMalloc(pEASData->hwInstData, sizeof(S_XMF_DATA));
+ if (!pXMFData)
+ return EAS_ERROR_MALLOC_FAILED;
+
+ /* zero the memory to insure complete initialization */
+ EAS_HWMemSet((void *)pXMFData,0, sizeof(S_XMF_DATA));
+
+ pXMFData->fileHandle = fileHandle;
+ pXMFData->fileOffset = offset;
+ *ppHandle = pXMFData;
+
+ /* locate the SMF and DLS contents */
+ if ((result = XMF_FindFileContents(pEASData->hwInstData, pXMFData)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No SMF data found in XMF file\n"); */ }
+ return result;
+ }
+
+ /* let the SMF parser take over */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, pXMFData->midiOffset)) != EAS_SUCCESS)
+ return result;
+ return SMF_CheckFileType(pEASData, fileHandle, &pXMFData->pSMFData, pXMFData->midiOffset);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Prepare()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Prepare to parse the file. Allocates instance data (or uses static allocation for
+ * static memory model).
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Prepare (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_XMF_DATA* pXMFData;
+ EAS_RESULT result;
+
+ /* parse DLS collection */
+ pXMFData = (S_XMF_DATA*) pInstData;
+ if (pXMFData->dlsOffset != 0)
+ {
+ if ((result = DLSParser(pEASData->hwInstData, pXMFData->fileHandle, pXMFData->dlsOffset, &pXMFData->pDLS)) != EAS_SUCCESS)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Error converting XMF DLS data\n"); */ }
+ return result;
+ }
+ }
+
+ /* Prepare the SMF parser */
+ if ((result = SMF_Prepare(pEASData, pXMFData->pSMFData)) != EAS_SUCCESS)
+ return result;
+
+ /* if no DLS file, skip this step */
+ if (pXMFData->pDLS == NULL)
+ return EAS_SUCCESS;
+
+ /* tell the synth to use the DLS collection */
+ result = VMSetDLSLib(((S_SMF_DATA*) pXMFData->pSMFData)->pSynth, pXMFData->pDLS);
+ if (result == EAS_SUCCESS)
+ {
+ DLSAddRef(pXMFData->pDLS);
+ VMInitializeAllChannels(pEASData->pVoiceMgr, ((S_SMF_DATA*) pXMFData->pSMFData)->pSynth);
+ }
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Time()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the time of the next event in msecs
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pTime - pointer to variable to hold time of next event (in msecs)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Time (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_U32 *pTime)
+{
+ return SMF_Time(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData, pTime);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Event()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Parse the next event in the file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Event (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_INT parserMode)
+{
+ return SMF_Event(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData, parserMode);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_State()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns the current state of the stream
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * pState - pointer to variable to store state
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_State (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 *pState)
+{
+ return SMF_State(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData, pState);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Close()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Close the file and clean up
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Close (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ S_XMF_DATA* pXMFData;
+ EAS_RESULT result;
+
+ pXMFData = (S_XMF_DATA *)pInstData;
+
+ /* close the SMF stream, it will close the file handle */
+ if ((result = SMF_Close(pEASData, pXMFData->pSMFData)) != EAS_SUCCESS)
+ return result;
+
+ if (pXMFData->pDLS)
+ DLSCleanup(pEASData->hwInstData, pXMFData->pDLS);
+
+ /* if using dynamic memory, free it */
+ if (!pEASData->staticMemoryModel)
+ {
+ /* free the instance data */
+ EAS_HWFree(pEASData->hwInstData, pXMFData);
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Reset()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reset the sequencer. Used for locating backwards in the file.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Reset (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ return SMF_Reset(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Pause()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Pauses the sequencer. Mutes all voices and sets state to pause.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Pause (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ return SMF_Pause(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_Resume()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Resume playing after a pause, sets state back to playing.
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_Resume (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData)
+{
+ return SMF_Resume(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_SetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Sets the playback rate of the underlying SMF file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * rate - rate (28-bit fraction)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_SetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 value)
+{
+ return SMF_SetData(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData, param, value);
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_GetData()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Gets the file type
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * handle - pointer to file handle
+ * rate - rate (28-bit fraction)
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_GetData (S_EAS_DATA *pEASData, EAS_VOID_PTR pInstData, EAS_I32 param, EAS_I32 *pValue)
+{
+ EAS_RESULT result;
+
+ /* call SMF parser to get value */
+ if ((result = SMF_GetData(pEASData, ((S_XMF_DATA*) pInstData)->pSMFData, param, pValue)) != EAS_SUCCESS)
+ return result;
+
+ /* special case for file type */
+ if (param == PARSER_DATA_FILE_TYPE)
+ {
+ if (*pValue == EAS_FILE_SMF0)
+ *pValue = EAS_FILE_XMF0;
+ else if (*pValue == EAS_FILE_SMF1)
+ *pValue = EAS_FILE_XMF1;
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_FindFileContents()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Finds SMF data and DLS data in XMF file, and remembers offset for each.
+ * If more than one is found, uses the first one found of each.
+ * Makes assumptions about the format of a mobile XMF file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pXMFData - pointer to XMF parser instance data
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_FindFileContents (EAS_HW_DATA_HANDLE hwInstData, S_XMF_DATA *pXMFData)
+{
+ EAS_RESULT result;
+ EAS_I32 value;
+ EAS_I32 length;
+
+ /* initialize offsets */
+ pXMFData->dlsOffset = pXMFData->midiOffset = 0;
+
+ /* read file length, ignore it for now */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* read MetaDataTypesTable length and skip over it */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+ if ((result = EAS_HWFileSeekOfs(hwInstData, pXMFData->fileHandle, value)) != EAS_SUCCESS)
+ return result;
+
+ /* get TreeStart offset and jump to it */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+ if ((result = XMF_ReadNode(hwInstData, pXMFData, value, &length)) != EAS_SUCCESS)
+ return result;
+
+ /* check for SMF data */
+ if (pXMFData->midiOffset == 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "No SMF data found in XMF file\n"); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* check for SFM in wrong order */
+ if ((pXMFData->dlsOffset > 0) && (pXMFData->midiOffset < pXMFData->dlsOffset))
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "DLS data must precede SMF data in Mobile XMF file\n"); */ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * XMF_ReadNode()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_ReadNode (EAS_HW_DATA_HANDLE hwInstData, S_XMF_DATA *pXMFData, EAS_I32 nodeOffset, EAS_I32 *pLength)
+{
+ EAS_RESULT result;
+ EAS_I32 refType;
+ EAS_I32 numItems;
+ EAS_I32 offset;
+ EAS_I32 length;
+ EAS_I32 headerLength;
+ EAS_U32 chunkType;
+
+ /* seek to start of node */
+ if ((result = EAS_HWFileSeek(hwInstData, pXMFData->fileHandle, nodeOffset)) != EAS_SUCCESS)
+ return result;
+
+ /* get node length */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, pLength)) != EAS_SUCCESS)
+ return result;
+
+ /* get number of contained items */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &numItems)) != EAS_SUCCESS)
+ return result;
+
+ /* get node header length */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &headerLength)) != EAS_SUCCESS)
+ return result;
+
+ /* get metadata length */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &length)) != EAS_SUCCESS)
+ return result;
+
+ /* get the current location */
+ if ((result = EAS_HWFilePos(hwInstData, pXMFData->fileHandle, &offset)) != EAS_SUCCESS)
+ return result;
+
+ /* skip to node contents */
+ if ((result = EAS_HWFileSeek(hwInstData, pXMFData->fileHandle, nodeOffset + headerLength)) != EAS_SUCCESS)
+ return result;
+
+ /* get reference type */
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &refType)) != EAS_SUCCESS)
+ return result;
+
+ /* get the current location */
+ if ((result = EAS_HWFilePos(hwInstData, pXMFData->fileHandle, &offset)) != EAS_SUCCESS)
+ return result;
+
+ /* process file node */
+ if (numItems == 0)
+
+ {
+ /* if in-file resource, find out where it is and jump to it */
+ if (refType == 2)
+ {
+ if ((result = XMF_ReadVLQ(hwInstData, pXMFData->fileHandle, &offset)) != EAS_SUCCESS)
+ return result;
+ offset += pXMFData->fileOffset;
+ if ((result = EAS_HWFileSeek(hwInstData, pXMFData->fileHandle, offset)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* or else it must be an inline resource */
+ else if (refType != 1)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Unexpected reference type %d\n", refType); */ }
+ return EAS_ERROR_FILE_FORMAT;
+ }
+
+ /* get the chunk type */
+ if ((result = EAS_HWGetDWord(hwInstData, pXMFData->fileHandle, &chunkType, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+
+ /* found a RIFF chunk, check for DLS type */
+ if (chunkType == XMF_RIFF_CHUNK)
+ {
+ /* skip length */
+ if ((result = EAS_HWFileSeekOfs(hwInstData, pXMFData->fileHandle, sizeof(EAS_I32))) != EAS_SUCCESS)
+ return result;
+
+ /* get RIFF file type */
+ if ((result = EAS_HWGetDWord(hwInstData, pXMFData->fileHandle, &chunkType, EAS_TRUE)) != EAS_SUCCESS)
+ return result;
+ if (chunkType == XMF_RIFF_DLS)
+ pXMFData->dlsOffset = offset;
+ }
+
+ /* found an SMF chunk */
+ else if (chunkType == XMF_SMF_CHUNK)
+ pXMFData->midiOffset = offset;
+ }
+
+ /* folder node, process the items in the list */
+ else
+ {
+ for ( ; numItems > 0; numItems--)
+ {
+ /* process this item */
+ if ((result = XMF_ReadNode(hwInstData, pXMFData, offset, &length)) != EAS_SUCCESS)
+ return result;
+
+ /* seek to start of next item */
+ offset += length;
+ if ((result = EAS_HWFileSeek(hwInstData, pXMFData->fileHandle, offset)) != EAS_SUCCESS)
+ return result;
+ }
+ }
+
+ return EAS_SUCCESS;
+}
+
+#if 0
+/*----------------------------------------------------------------------------
+ * XMF_FindFileContents()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Finds SMF data and DLS data in XMF file, and remembers offset for each.
+ * If more than one is found, uses the first one found of each.
+ * Makes assumptions about the format of a mobile XMF file
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * pXMFData - pointer to XMF parser instance data
+ * handle - pointer to file handle
+ *
+ * Outputs:
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_FindFileContents(S_EAS_DATA *pEASData, S_XMF_DATA *pXMFData, EAS_FILE_HANDLE fileHandle)
+{
+ EAS_RESULT result;
+ EAS_I32 offset;
+ EAS_I32 value;
+ EAS_I32 numItems;
+ EAS_I32 length;
+ EAS_CHAR id[4];
+ EAS_I32 location;
+
+ /* init dls offset, so that we know we haven't found a dls chunk yet */
+ pXMFData->dlsOffset = 0;
+
+ /* read file length, ignore it for now */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* read MetaDataTypesTable length and skip over it */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+ if ((result = EAS_HWFileSeekOfs(pEASData, fileHandle, value)) != EAS_SUCCESS)
+ return result;
+
+ /* get TreeStart offset and jump to it */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &offset)) != EAS_SUCCESS)
+ return result;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset)) != EAS_SUCCESS)
+ return result;
+
+ /* read node length, ignore it for now */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* read number of contained items */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &numItems)) != EAS_SUCCESS)
+ return result;
+
+ /*read node header length */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /*go to the node offset */
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset + value)) != EAS_SUCCESS)
+ return result;
+
+ /* read Reference Type */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* make sure it is an in-line resource, for now */
+ if (value != 1)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Problem parsing XMF file tree\n"); */ }
+ return EAS_FAILURE;
+ }
+
+ /* parse through the list of items */
+ while (numItems > 0)
+ {
+ /*get current offset */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &offset)) != EAS_SUCCESS)
+ return result;
+
+ /*read node length */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &length)) != EAS_SUCCESS)
+ return result;
+
+ /* read number of items */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* make sure not a folder */
+ if (value != 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Problem parsing XMF file node\n"); */ }
+ return EAS_FAILURE;
+ }
+
+ /* read offset to resource and jump to it */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset + value)) != EAS_SUCCESS)
+ return result;
+
+ /* read Reference Type */
+ if ((result = XMF_ReadVLQ(pEASData, fileHandle, &value)) != EAS_SUCCESS)
+ return result;
+
+ /* make sure it is an in-line resource */
+ if (value != 1)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "Problem parsing XMF file node\n"); */ }
+ return EAS_FAILURE;
+ }
+
+ /* get current offset as a possible location for SMF file or DLS file */
+ if ((result = EAS_HWFilePos(pEASData->hwInstData, fileHandle, &location)) != EAS_SUCCESS)
+ return result;
+
+ /* read four bytes */
+ if ((result = EAS_HWReadFile(pEASData->hwInstData, fileHandle, id, sizeof(id), &value)) != EAS_SUCCESS)
+ return result;
+
+ /* check if DLS */
+ if (pXMFData->dlsOffset == 0 && id[0] == 'R' && id[1] == 'I' && id[2] == 'F' && id[3] == 'F')
+ {
+ //remember offset
+ pXMFData->dlsOffset = location;
+ }
+
+ /* else check if SMF */
+ else if (id[0] == 'M' && id[1] == 'T' && id[2] == 'h' && id[3] == 'd')
+ {
+ //remember offset
+ pXMFData->midiOffset = location;
+
+ //we are done
+ return EAS_SUCCESS;
+ }
+
+ //one less item
+ numItems--;
+
+ //if more data, go to the next item
+ if (numItems >0)
+ {
+ if ((result = EAS_HWFileSeek(pEASData->hwInstData, fileHandle, offset + length)) != EAS_SUCCESS)
+ return result;
+ }
+ }
+
+ return EAS_FAILURE;
+
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * XMF_ReadVLQ()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Reads a VLQ encoded value from the file referenced by fileHandle
+ *
+ * Inputs:
+ * pEASData - pointer to overall EAS data structure
+ * fileHandle - pointer to file handle
+ *
+ * Outputs:
+ * value - pointer to the value decoded from the VLQ data
+ *
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT XMF_ReadVLQ (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 *value)
+{
+ EAS_RESULT result;
+ EAS_U8 c;
+
+ *value = 0;
+
+ if ((result = EAS_HWGetByte(hwInstData, fileHandle, &c)) != EAS_SUCCESS)
+ return result;
+
+ while (c > 0x7F)
+ {
+ /*lint -e{703} shift for performance */
+ *value = (*value << 7) | (c & 0x7F);
+
+ if ((result = EAS_HWGetByte(hwInstData, fileHandle, &c)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /*lint -e{703} shift for performance */
+ *value = (*value << 7) | c;
+
+ return EAS_SUCCESS;
+}
+
diff --git a/arm-wt-22k/lib_src/eas_xmf.h b/arm-wt-22k/lib_src/eas_xmf.h new file mode 100644 index 0000000..970d00c --- /dev/null +++ b/arm-wt-22k/lib_src/eas_xmf.h @@ -0,0 +1,60 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_xmf.h
+ *
+ * Contents and purpose:
+ * XMF Type 0 and 1 File Parser
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_XMF_H
+#define _EAS_XMF_H
+
+#ifndef MAX_XMF_STREAMS
+#define MAX_XMF_STREAMS 16
+#endif
+
+/* offsets in to the XMF file */
+#define XMF_OFS_HEADER_SIZE 4
+#define XMF_OFS_FILE_TYPE 8
+#define XMF_OFS_NUM_TRACKS 10
+
+/* size of chunk info (chunk ID + chunk size) */
+#define XMF_CHUNK_INFO_SIZE 8
+
+/* 'MTrk' track chunk ID */
+#define XMF_CHUNK_TYPE_TRACK 0x4d54726bL
+
+/* some useful meta-events */
+#define XMF_META_END_OF_TRACK 0x2f
+#define XMF_META_TEMPO 0x51
+
+/* default timebase (120BPM) */
+#define XMF_DEFAULT_TIMEBASE 500000L
+
+/* value for pXMFStream->ticks to signify end of track */
+#define XMF_END_OF_TRACK 0xffffffff
+
+#endif /* end _EAS_XMF_H */
+
+
diff --git a/arm-wt-22k/lib_src/eas_xmfdata.c b/arm-wt-22k/lib_src/eas_xmfdata.c new file mode 100644 index 0000000..ed2e84a --- /dev/null +++ b/arm-wt-22k/lib_src/eas_xmfdata.c @@ -0,0 +1,44 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_xmfdata.c
+ *
+ * Contents and purpose:
+ * XMF File Parser
+ *
+ * This file contains data definitions for the XMF parser.
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 547 $
+ * $Date: 2007-01-31 16:30:17 -0800 (Wed, 31 Jan 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_miditypes.h"
+#include "eas_xmf.h"
+#include "eas_xmfdata.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * eas_XMFData
+ *
+ * Static memory allocation for XMF parser
+ *----------------------------------------------------------------------------
+*/
+S_XMF_DATA eas_XMFData;
+
diff --git a/arm-wt-22k/lib_src/eas_xmfdata.h b/arm-wt-22k/lib_src/eas_xmfdata.h new file mode 100644 index 0000000..c535d55 --- /dev/null +++ b/arm-wt-22k/lib_src/eas_xmfdata.h @@ -0,0 +1,55 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_xmfdata.h
+ *
+ * Contents and purpose:
+ * Contains declarations for the XMF file parser.
+ *
+ *
+ * Copyright Sonic Network Inc. 2005
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 82 $
+ * $Date: 2006-07-10 11:45:19 -0700 (Mon, 10 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _EAS_XMFDATA_H
+#define _EAS_XMFDATA_H
+
+#include "eas_data.h"
+
+/*----------------------------------------------------------------------------
+ *
+ * S_XMF_DATA
+ *
+ * This structure contains the instance data required to parse an XMF file.
+ *
+ *----------------------------------------------------------------------------
+*/
+
+typedef struct
+{
+ EAS_FILE_HANDLE fileHandle;
+ EAS_I32 fileOffset;
+ EAS_VOID_PTR pSMFData;
+ EAS_I32 midiOffset;
+ EAS_I32 dlsOffset;
+ S_DLS *pDLS;
+} S_XMF_DATA;
+
+#endif
diff --git a/arm-wt-22k/lib_src/jet.c b/arm-wt-22k/lib_src/jet.c new file mode 100644 index 0000000..bcb4e5e --- /dev/null +++ b/arm-wt-22k/lib_src/jet.c @@ -0,0 +1,1125 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * jet.c
+ *
+ * Contents and purpose:
+ * Implementation for JET sound engine
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 563 $
+ * $Date: 2007-02-13 20:26:23 -0800 (Tue, 13 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+//FIXME: remove debug flag for release
+#define LOG_NDEBUG 0
+#define LOG_TAG "JET_C"
+
+#define DEBUG_JET
+
+#include "eas_data.h"
+#include "eas_smf.h"
+#include "jet_data.h"
+#include "eas_host.h"
+#include "eas_report.h"
+
+#include <cutils/log.h>
+
+/* default configuration */
+static const S_JET_CONFIG jetDefaultConfig =
+{
+ JET_EVENT_APP_LOW,
+ JET_EVENT_APP_HIGH
+};
+
+/* function prototypes */
+extern EAS_RESULT EAS_IntSetStrmParam (S_EAS_DATA *pEASData, EAS_HANDLE pStream, EAS_INT param, EAS_I32 value);
+extern EAS_RESULT EAS_OpenJETStream (EAS_DATA_HANDLE pEASData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, EAS_HANDLE *ppStream);
+extern EAS_RESULT DLSParser (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE fileHandle, EAS_I32 offset, EAS_DLSLIB_HANDLE *ppDLS);
+
+/*----------------------------------------------------------------------------
+ * JET_ParseEvent()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC void JET_ParseEvent (EAS_U32 event, S_JET_EVENT *pEvent)
+{
+ pEvent->segment = (EAS_U8) ((event & JET_EVENT_SEG_MASK) >> JET_EVENT_SEG_SHIFT);
+ pEvent->track = (EAS_U8) ((event & JET_EVENT_TRACK_MASK) >> JET_EVENT_TRACK_SHIFT);
+ pEvent->channel = (EAS_U8) ((event & JET_EVENT_CHAN_MASK) >> JET_EVENT_CHAN_SHIFT);
+ pEvent->controller = (EAS_U8) ((event & JET_EVENT_CTRL_MASK) >> JET_EVENT_CTRL_SHIFT);
+ pEvent->value = (EAS_U8) (event & JET_EVENT_VAL_MASK);
+}
+
+#ifdef DEBUG_JET
+/*----------------------------------------------------------------------------
+ * JET_DumpEvent
+ *----------------------------------------------------------------------------
+ * Advances queue read/write index
+ *----------------------------------------------------------------------------
+*/
+static void JET_DumpEvent (const char *procName, EAS_U32 event)
+{
+ S_JET_EVENT sEvent;
+ JET_ParseEvent(event, &sEvent);
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "%s: SegID=%d, TrkID=%d, channel=%d, ctrl=%d, val=%d\n",
+ procName, sEvent.segment, sEvent.track, sEvent.channel, sEvent.controller, sEvent.value); */ }
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * JET_IncQueueIndex
+ *----------------------------------------------------------------------------
+ * Advances queue read/write index
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_U8 JET_IncQueueIndex (EAS_U8 index, EAS_U8 queueSize)
+{
+ if (++index == queueSize)
+ index = 0;
+ return index;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_WriteQueue
+ *----------------------------------------------------------------------------
+ * Save event to queue
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE void JET_WriteQueue (EAS_U32 *pEventQueue, EAS_U8 *pWriteIndex, EAS_U8 readIndex, EAS_U8 queueSize, EAS_U32 event)
+{
+ EAS_U8 temp;
+
+ /* check for queue overflow */
+ temp = JET_IncQueueIndex(*pWriteIndex, queueSize);
+ if (temp == readIndex)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "JET_Event: Event queue overflow --- event ignored!\n"); */ }
+ return;
+ }
+
+ /* save in queue and advance write index */
+ pEventQueue[*pWriteIndex] = event;
+ *pWriteIndex = temp;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_ReadQueue
+ *----------------------------------------------------------------------------
+ * Read event to queue
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_BOOL JET_ReadQueue (EAS_U32 *pEventQueue, EAS_U8 *pReadIndex, EAS_U8 writeIndex, EAS_U8 queueSize, EAS_U32 *pEvent)
+{
+
+ /* check for empty queue */
+ if (*pReadIndex == writeIndex)
+ return EAS_FALSE;
+
+ /* save in queue and advance write index */
+ *pEvent = pEventQueue[*pReadIndex];
+ *pReadIndex = JET_IncQueueIndex(*pReadIndex, queueSize);
+ return EAS_TRUE;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_NextSegment
+ *----------------------------------------------------------------------------
+ * Advances segment number
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_INT JET_NextSegment (EAS_INT seg_num)
+{
+ if (++seg_num == SEG_QUEUE_DEPTH)
+ seg_num = 0;
+ return seg_num;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_PrepareSegment()
+ *----------------------------------------------------------------------------
+ * Prepare a segment for playback
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT JET_PrepareSegment (EAS_DATA_HANDLE easHandle, EAS_I32 queueNum)
+{
+ EAS_RESULT result;
+ S_JET_SEGMENT *p;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_PrepareSegment: %d\n", queueNum); */ }
+
+ p = &easHandle->jetHandle->segQueue[queueNum];
+ result = EAS_Prepare(easHandle, p->streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ /* pause segment - must be triggered by play or end of previous segment */
+ result = EAS_Pause(easHandle, p->streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+ p->state = JET_STATE_READY;
+
+ /* set calback data */
+ result = EAS_IntSetStrmParam(easHandle, p->streamHandle, PARSER_DATA_JET_CB, queueNum);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ /* set DLS collection */
+ if (p->libNum >= 0)
+ {
+ result = EAS_IntSetStrmParam(easHandle, p->streamHandle,
+ PARSER_DATA_DLS_COLLECTION, (EAS_I32) easHandle->jetHandle->libHandles[p->libNum]);
+ if (result != EAS_SUCCESS)
+ return result;
+ }
+
+ /* set transposition */
+ if (p->transpose)
+ {
+ result = EAS_SetTransposition(easHandle, p->streamHandle, p->transpose);
+ if (result != EAS_SUCCESS)
+ return result;
+ }
+
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_StartPlayback()
+ *----------------------------------------------------------------------------
+ * Start segment playback
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT JET_StartPlayback (EAS_DATA_HANDLE easHandle, EAS_I32 queueNum)
+{
+ EAS_RESULT result = EAS_SUCCESS;
+ S_JET_SEGMENT *pSeg;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_StartPlayback %d\n", queueNum); */ }
+
+ /* if next segment is queued, start playback */
+ pSeg = &easHandle->jetHandle->segQueue[queueNum];
+ if (pSeg->streamHandle != NULL)
+ {
+ result = EAS_Resume(easHandle, pSeg->streamHandle);
+ easHandle->jetHandle->segQueue[queueNum].state = JET_STATE_PLAYING;
+
+ /* set mute flags */
+ if ((result == EAS_SUCCESS) && (pSeg->muteFlags != 0))
+ result = EAS_IntSetStrmParam(easHandle, pSeg->streamHandle, PARSER_DATA_MUTE_FLAGS, (EAS_I32) pSeg->muteFlags);
+ }
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_CloseSegment
+ *----------------------------------------------------------------------------
+ * Closes stream associated with a segment
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_INT JET_CloseSegment (EAS_DATA_HANDLE easHandle, EAS_INT queueNum)
+{
+ EAS_RESULT result;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_CloseSegment %d\n", queueNum); */ }
+
+ /* close the segment */
+ result = EAS_CloseFile(easHandle, easHandle->jetHandle->segQueue[queueNum].streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ easHandle->jetHandle->segQueue[queueNum].streamHandle = NULL;
+ easHandle->jetHandle->segQueue[queueNum].state = JET_STATE_CLOSED;
+ easHandle->jetHandle->numQueuedSegments--;
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JetParseInfoChunk()
+ *----------------------------------------------------------------------------
+ * Parses the JET info chunk
+ *----------------------------------------------------------------------------
+*/
+static EAS_RESULT JetParseInfoChunk (EAS_DATA_HANDLE easHandle, EAS_I32 pos, EAS_I32 chunkSize)
+{
+ EAS_RESULT result;
+ EAS_U32 infoType;
+ EAS_U32 temp;
+
+ /* offset to data */
+ result = EAS_HWFileSeek(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, pos);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ /* read the entire chunk */
+ result = EAS_SUCCESS;
+ while ((result == EAS_SUCCESS) && (chunkSize > 0))
+ {
+
+ /* get info infoType */
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &infoType, EAS_TRUE);
+ if (result != EAS_SUCCESS)
+ break;
+
+ /* get info field */
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &temp, EAS_FALSE);
+ if (result == EAS_SUCCESS)
+
+ switch (infoType)
+ {
+ case INFO_NUM_SMF_CHUNKS:
+ easHandle->jetHandle->numSegments = (EAS_U8) temp;
+ break;
+
+ case INFO_NUM_DLS_CHUNKS:
+ easHandle->jetHandle->numLibraries = (EAS_U8) temp;
+ break;
+
+ case INFO_JET_VERSION:
+ /* check major version number */
+ if ((temp & 0xff000000) != (JET_VERSION & 0xff000000))
+ return EAS_ERROR_INCOMPATIBLE_VERSION;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unrecognized JET info type 0x%08x", infoType); */ }
+ break;
+ }
+
+ chunkSize -= 8;
+ }
+
+ /* allocate pointers for chunks to follow */
+
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_OpenFile()
+ *----------------------------------------------------------------------------
+ * Opens a JET content file for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_OpenFile (EAS_DATA_HANDLE easHandle, EAS_FILE_LOCATOR locator)
+{
+ EAS_RESULT result;
+ EAS_U32 chunkType;
+ EAS_I32 pos;
+ EAS_I32 chunkSize;
+ EAS_INT smfChunkNum;
+ EAS_INT dlsChunkNum;
+ EAS_I32 dataSize = 0; /* make lint happy */
+
+ /* make sure that we don't have an open file */
+ if (easHandle->jetHandle->jetFileHandle != NULL)
+ return EAS_ERROR_FILE_ALREADY_OPEN;
+
+ /* open the media file */
+ result = EAS_HWOpenFile(easHandle->hwInstData, locator, &easHandle->jetHandle->jetFileHandle, EAS_FILE_READ);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ /* check header */
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &chunkType, EAS_TRUE);
+ if (result == EAS_SUCCESS)
+ {
+ if (chunkType != JET_HEADER_TAG)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "File is not JET format\n"); */ }
+ result = EAS_ERROR_UNRECOGNIZED_FORMAT;
+ }
+ }
+ /* get the file data size */
+ if (result == EAS_SUCCESS)
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &dataSize, EAS_FALSE);
+
+ /* parse through the file to find contents */
+ smfChunkNum = dlsChunkNum = 0;
+ pos = chunkSize = 8;
+ while ((result == EAS_SUCCESS) && (pos < dataSize))
+ {
+
+ /* offset to chunk data */
+ result = EAS_HWFileSeek(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, pos);
+ if (result != EAS_SUCCESS)
+ break;
+
+ /* get chunk size and type */
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &chunkType, EAS_TRUE);
+ if (result != EAS_SUCCESS)
+ break;
+
+ result = EAS_HWGetDWord(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &chunkSize, EAS_FALSE);
+ if (result != EAS_SUCCESS)
+ break;
+ pos += 8;
+
+ switch (chunkType)
+ {
+ case JET_INFO_CHUNK:
+ result = JetParseInfoChunk(easHandle, pos, chunkSize);
+ break;
+
+ case JET_SMF_CHUNK:
+ if (smfChunkNum < easHandle->jetHandle->numSegments)
+ easHandle->jetHandle->segmentOffsets[smfChunkNum++] = pos;
+ else
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring extraneous SMF chunk"); */ }
+ break;
+
+ case JET_DLS_CHUNK:
+ if (dlsChunkNum < easHandle->jetHandle->numLibraries)
+ result = DLSParser(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, pos, &easHandle->jetHandle->libHandles[dlsChunkNum++]);
+ else
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring extraneous DLS chunk"); */ }
+ break;
+
+ case JET_APP_DATA_CHUNK:
+ easHandle->jetHandle->appDataOffset = pos;
+ easHandle->jetHandle->appDataSize = chunkSize;
+ break;
+
+ case INFO_JET_COPYRIGHT:
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_WARNING, "Ignoring unrecognized JET chunk type 0x%08x", chunkType); */ }
+ break;
+ }
+
+ /* offset to next chunk */
+ pos += chunkSize;
+ }
+
+ /* close file if something went wrong */
+ if (result != EAS_SUCCESS)
+ EAS_HWCloseFile(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle);
+
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_GetAppData()
+ *----------------------------------------------------------------------------
+ * Returns location and size of application data in the JET file
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT JET_GetAppData (EAS_DATA_HANDLE easHandle, EAS_I32 *pAppDataOffset, EAS_I32 *pAppDataSize)
+{
+
+ /* check for app chunk */
+ if (easHandle->jetHandle->appDataSize == 0)
+ {
+ *pAppDataOffset = *pAppDataSize = 0;
+ return EAS_FAILURE;
+ }
+
+ /* return app data */
+ *pAppDataOffset = easHandle->jetHandle->appDataOffset;
+ *pAppDataSize = easHandle->jetHandle->appDataSize;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_CloseFile()
+ *----------------------------------------------------------------------------
+ * Closes a JET content file and releases associated resources
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_CloseFile (EAS_DATA_HANDLE easHandle)
+{
+ EAS_INT index;
+ EAS_RESULT result = EAS_SUCCESS;
+
+ /* close open streams */
+ for (index = 0; index < SEG_QUEUE_DEPTH; index++)
+ {
+ if (easHandle->jetHandle->segQueue[index].streamHandle != NULL)
+ {
+ result = JET_CloseSegment(easHandle, index);
+ if (result != EAS_SUCCESS)
+ break;
+ }
+ }
+
+ /* close the main file handle */
+ if ((result == EAS_SUCCESS) && (easHandle->jetHandle->jetFileHandle != NULL))
+ {
+ result = EAS_HWCloseFile(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle);
+ if (result == EAS_SUCCESS)
+ easHandle->jetHandle->jetFileHandle = NULL;
+ }
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Init()
+ *----------------------------------------------------------------------------
+ * Initializes the JET library, allocates memory, etc. Call
+ * JET_Shutdown to de-allocate memory.
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Init (EAS_DATA_HANDLE easHandle, const S_JET_CONFIG *pConfig, EAS_INT configSize)
+{
+ S_JET_DATA *pJet;
+ EAS_U8 flags = 0;
+
+ /* sanity check */
+ if (easHandle == NULL)
+ return EAS_ERROR_HANDLE_INTEGRITY;
+ if (easHandle->jetHandle != NULL)
+ return EAS_ERROR_FEATURE_ALREADY_ACTIVE;
+ if (pConfig == NULL)
+ pConfig = &jetDefaultConfig;
+
+ /* allocate the JET data object */
+ pJet = EAS_HWMalloc(easHandle->hwInstData, sizeof(S_JET_DATA));
+ if (pJet == NULL)
+ return EAS_ERROR_MALLOC_FAILED;
+
+ /* initialize JET data structure */
+ EAS_HWMemSet(pJet, 0, sizeof(S_JET_DATA));
+ easHandle->jetHandle = pJet;
+ pJet->flags = flags;
+
+ /* copy config data */
+ if (configSize > (EAS_INT) sizeof(S_JET_CONFIG))
+ configSize = sizeof(S_JET_CONFIG);
+ EAS_HWMemCpy(&pJet->config, pConfig, configSize);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Shutdown()
+ *----------------------------------------------------------------------------
+ * Frees any memory used by the JET library
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Shutdown (EAS_DATA_HANDLE easHandle)
+{
+ EAS_RESULT result;
+
+ /* close any open files */
+ result = JET_CloseFile(easHandle);
+
+ /* free allocated data */
+ EAS_HWFree(easHandle->hwInstData, easHandle->jetHandle);
+ easHandle->jetHandle = NULL;
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Status()
+ *----------------------------------------------------------------------------
+ * Returns current status
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Status (EAS_DATA_HANDLE easHandle, S_JET_STATUS *pStatus)
+{
+ S_JET_SEGMENT *pSeg;
+
+ pSeg = &easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment];
+ if (pSeg->streamHandle != NULL)
+ {
+ pStatus->currentUserID = pSeg->userID;
+ pStatus->segmentRepeatCount = pSeg->repeatCount;
+ }
+ else
+ {
+ pStatus->currentUserID = -1;
+ pStatus->segmentRepeatCount = 0;
+ }
+
+ pStatus->paused = !(easHandle->jetHandle->flags & JET_FLAGS_PLAYING);
+ pStatus->numQueuedSegments = easHandle->jetHandle->numQueuedSegments;
+ pStatus->currentPlayingSegment = easHandle->jetHandle->playSegment;
+ pStatus->currentQueuedSegment = easHandle->jetHandle->queueSegment;
+ if (pSeg->streamHandle != NULL)
+ {
+ EAS_RESULT result;
+ EAS_I32 location ;
+ if ((result = EAS_GetLocation(easHandle, pSeg->streamHandle, &location)) == EAS_SUCCESS)
+ if(location != 0)
+ {
+ pStatus->location = location;
+ }
+ }
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_GetEvent()
+ *----------------------------------------------------------------------------
+ * Checks for application events
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL JET_GetEvent (EAS_DATA_HANDLE easHandle, EAS_U32 *pEventRaw, S_JET_EVENT *pEvent)
+{
+ EAS_U32 jetEvent;
+ EAS_BOOL gotEvent;
+
+ /* process event queue */
+ gotEvent = JET_ReadQueue(easHandle->jetHandle->appEventQueue,
+ &easHandle->jetHandle->appEventQueueRead,
+ easHandle->jetHandle->appEventQueueWrite,
+ APP_EVENT_QUEUE_SIZE, &jetEvent);
+
+ if (gotEvent)
+ {
+ if (pEventRaw != NULL)
+ *pEventRaw = jetEvent;
+
+ if (pEvent != NULL)
+ JET_ParseEvent(jetEvent, pEvent);
+ }
+
+ return gotEvent;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_QueueSegment()
+ *----------------------------------------------------------------------------
+ * Queue a segment for playback
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_QueueSegment (EAS_DATA_HANDLE easHandle, EAS_INT segmentNum, EAS_INT libNum, EAS_INT repeatCount, EAS_INT transpose, EAS_U32 muteFlags, EAS_U8 userID)
+{
+ EAS_FILE_HANDLE fileHandle;
+ EAS_RESULT result;
+ S_JET_SEGMENT *p;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_QueueSegment segNum=%d, queue=%d\n", segmentNum, easHandle->jetHandle->queueSegment); */ }
+
+ /* make sure it's a valid segment */
+ if (segmentNum >= easHandle->jetHandle->numSegments)
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* make sure it's a valid DLS */
+ if (libNum >= easHandle->jetHandle->numLibraries)
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* check to see if queue is full */
+ p = &easHandle->jetHandle->segQueue[easHandle->jetHandle->queueSegment];
+ if (p->streamHandle != NULL)
+ return EAS_ERROR_QUEUE_IS_FULL;
+
+ /* initialize data */
+ p->userID = userID;
+ p->repeatCount = (EAS_I16) repeatCount;
+ p->transpose = (EAS_I8) transpose;
+ p->libNum = (EAS_I8) libNum;
+ p->muteFlags = muteFlags;
+ p->state = JET_STATE_CLOSED;
+
+ /* open the file */
+ result = EAS_OpenJETStream(easHandle, easHandle->jetHandle->jetFileHandle, easHandle->jetHandle->segmentOffsets[segmentNum], &p->streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+ p->state = JET_STATE_OPEN;
+
+ /* if less than SEG_QUEUE_DEPTH segments queued up, prepare file for playback */
+ if (++easHandle->jetHandle->numQueuedSegments < SEG_QUEUE_DEPTH)
+ {
+ result = JET_PrepareSegment(easHandle, easHandle->jetHandle->queueSegment);
+ if (result != EAS_SUCCESS)
+ return result;
+ }
+
+ /* create duplicate file handle */
+ result = EAS_HWDupHandle(easHandle->hwInstData, easHandle->jetHandle->jetFileHandle, &fileHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ easHandle->jetHandle->jetFileHandle = fileHandle;
+ easHandle->jetHandle->queueSegment = (EAS_U8) JET_NextSegment(easHandle->jetHandle->queueSegment);
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Play()
+ *----------------------------------------------------------------------------
+ * Starts playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Play (EAS_DATA_HANDLE easHandle)
+{
+ EAS_RESULT result;
+ EAS_INT index;
+ EAS_INT count = 0;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_Play\n"); */ }
+
+ /* sanity check */
+ if (easHandle->jetHandle->flags & JET_FLAGS_PLAYING)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* resume all paused streams */
+ for (index = 0; index < SEG_QUEUE_DEPTH; index++)
+ {
+ if (((index == easHandle->jetHandle->playSegment) && (easHandle->jetHandle->segQueue[index].state == JET_STATE_READY)) ||
+ (easHandle->jetHandle->segQueue[index].state == JET_STATE_PAUSED))
+ {
+ result = JET_StartPlayback(easHandle, index);
+ if (result != EAS_SUCCESS)
+ return result;
+ count++;
+ }
+ }
+
+ /* if no streams are playing, return error */
+ if (!count)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+
+ easHandle->jetHandle->flags |= JET_FLAGS_PLAYING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Pause()
+ *----------------------------------------------------------------------------
+ * Pauses playback of the file
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Pause (EAS_DATA_HANDLE easHandle)
+{
+ EAS_RESULT result;
+ EAS_INT index;
+ EAS_INT count = 0;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_Pause\n"); */ }
+
+ /* sanity check */
+ if ((easHandle->jetHandle->flags & JET_FLAGS_PLAYING) == 0)
+ return EAS_ERROR_NOT_VALID_IN_THIS_STATE;
+
+ /* pause all playing streams */
+ for (index = 0; index < SEG_QUEUE_DEPTH; index++)
+ {
+ if (easHandle->jetHandle->segQueue[index].state == JET_STATE_PLAYING)
+ {
+ result = EAS_Pause(easHandle, easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment].streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+ easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment].state = JET_STATE_PAUSED;
+ count++;
+ }
+ }
+
+ /* if no streams are paused, return error */
+ if (!count)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+
+ easHandle->jetHandle->flags &= ~JET_FLAGS_PLAYING;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlags()
+ *----------------------------------------------------------------------------
+ * Change the state of the mute flags
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlags (EAS_DATA_HANDLE easHandle, EAS_U32 muteFlags, EAS_BOOL sync)
+{
+ S_JET_SEGMENT *pSeg;
+
+ /* get pointer to current segment */
+ pSeg = &easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment];
+
+ /* unsynchronized mute, set flags and return */
+ if (!sync)
+ {
+ if (pSeg->streamHandle == NULL)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+ pSeg->muteFlags = muteFlags;
+ return EAS_IntSetStrmParam(easHandle, pSeg->streamHandle, PARSER_DATA_MUTE_FLAGS, (EAS_I32) muteFlags);
+ }
+
+
+ /* check for valid stream state */
+ if (pSeg->state == JET_STATE_CLOSED)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+
+ /* save mute flags */
+ pSeg->muteFlags = muteFlags;
+
+ /* if repeating segment, set mute update flag */
+ if (sync)
+ pSeg->flags |= JET_SEG_FLAG_MUTE_UPDATE;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_SetMuteFlag()
+ *----------------------------------------------------------------------------
+ * Change the state of a single mute flag
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_SetMuteFlag (EAS_DATA_HANDLE easHandle, EAS_INT trackNum, EAS_BOOL muteFlag, EAS_BOOL sync)
+{
+ S_JET_SEGMENT *pSeg;
+ EAS_U32 trackMuteFlag;
+
+
+ /* setup flag */
+ if ((trackNum < 0) || (trackNum > 31))
+ return EAS_ERROR_PARAMETER_RANGE;
+ trackMuteFlag = (1 << trackNum);
+
+ /* get pointer to current segment */
+ pSeg = &easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment];
+
+ /* unsynchronized mute, set flags and return */
+ if (!sync)
+ {
+ if (pSeg->streamHandle == NULL)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+ if (muteFlag)
+ pSeg->muteFlags |= trackMuteFlag;
+ else
+ pSeg->muteFlags &= ~trackMuteFlag;
+ return EAS_IntSetStrmParam(easHandle, pSeg->streamHandle, PARSER_DATA_MUTE_FLAGS, (EAS_I32) pSeg->muteFlags);
+ }
+
+
+ /* check for valid stream state */
+ if (pSeg->state == JET_STATE_CLOSED)
+ return EAS_ERROR_QUEUE_IS_EMPTY;
+
+ /* save mute flags and set mute update flag */
+ if (muteFlag)
+ pSeg->muteFlags |= trackMuteFlag;
+ else
+ pSeg->muteFlags &= ~trackMuteFlag;
+ pSeg->flags |= JET_SEG_FLAG_MUTE_UPDATE;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_TriggerClip()
+ *----------------------------------------------------------------------------
+ * Unmute a track and then mute it when it is complete. If a clip
+ * is already playing, change mute event to a trigger event. The
+ * JET_Event function will not mute the clip, but will allow it
+ * to continue playing through the next clip.
+ *
+ * NOTE: We use bit 7 to indicate an entry in the queue. For a
+ * small queue, it is cheaper in both memory and CPU cycles to
+ * scan the entire queue for non-zero events than keep enqueue
+ * and dequeue indices.
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_TriggerClip (EAS_DATA_HANDLE easHandle, EAS_INT clipID)
+{
+ EAS_INT i;
+ EAS_INT index = -1;
+
+ /* check for valid clipID */
+ if ((clipID < 0) || (clipID > 63))
+ return EAS_ERROR_PARAMETER_RANGE;
+
+ /* set active flag */
+ clipID |= JET_CLIP_ACTIVE_FLAG;
+
+ /* Reverse the search so that we get the first empty element */
+ for (i = JET_MUTE_QUEUE_SIZE-1; i >= 0 ; i--)
+ {
+ if (easHandle->jetHandle->muteQueue[i] == clipID)
+ {
+ index = i;
+ break;
+ }
+ if (easHandle->jetHandle->muteQueue[i] == 0)
+ index = i;
+ }
+ if (index < 0)
+ return EAS_ERROR_QUEUE_IS_FULL;
+
+ easHandle->jetHandle->muteQueue[index] = (EAS_U8) clipID | JET_CLIP_TRIGGER_FLAG;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Process()
+ *----------------------------------------------------------------------------
+ * Called during EAS_Render to process stream states
+ *----------------------------------------------------------------------------
+*/
+EAS_PUBLIC EAS_RESULT JET_Process (EAS_DATA_HANDLE easHandle)
+{
+ S_JET_SEGMENT *pSeg;
+ EAS_STATE state;
+ EAS_INT index;
+ EAS_INT playIndex;
+ EAS_RESULT result = EAS_SUCCESS;
+ EAS_BOOL endOfLoop = EAS_FALSE;
+ EAS_BOOL startNextSegment = EAS_FALSE;
+ EAS_BOOL prepareNextSegment = EAS_FALSE;
+ EAS_U32 jetEvent;
+
+ /* process event queue */
+ while (JET_ReadQueue(easHandle->jetHandle->jetEventQueue,
+ &easHandle->jetHandle->jetEventQueueRead,
+ easHandle->jetHandle->jetEventQueueWrite,
+ JET_EVENT_QUEUE_SIZE, &jetEvent))
+ {
+ S_JET_EVENT event;
+#ifdef DEBUG_JET
+ JET_DumpEvent("JET_Process", jetEvent);
+#endif
+ JET_ParseEvent(jetEvent, &event);
+
+ /* check for end of loop */
+ if ((event.controller == JET_EVENT_MARKER) &&
+ (event.value == JET_MARKER_LOOP_END) &&
+ (easHandle->jetHandle->segQueue[easHandle->jetHandle->playSegment].streamHandle != NULL))
+ endOfLoop = EAS_TRUE;
+ }
+
+ /* check state of all streams */
+ index = playIndex = easHandle->jetHandle->playSegment;
+ for (;;)
+ {
+ pSeg = &easHandle->jetHandle->segQueue[index];
+ if (pSeg->state != JET_STATE_CLOSED)
+ {
+
+ /* get playback state */
+ result = EAS_State(easHandle, pSeg->streamHandle, &state);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ /* process state */
+ switch (pSeg->state)
+ {
+ /* take action if this segment is stopping */
+ case JET_STATE_PLAYING:
+ if (endOfLoop || (state == EAS_STATE_STOPPING) || (state == EAS_STATE_STOPPED))
+ {
+ /* handle repeats */
+ if (pSeg->repeatCount != 0)
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_Render repeating segment %d\n", index); */ }
+ result = EAS_Locate(easHandle, pSeg->streamHandle, 0, EAS_FALSE);
+ if (result != EAS_SUCCESS)
+ return result;
+ if (pSeg->repeatCount > 0)
+ pSeg->repeatCount--;
+
+ /* update mute flags if necessary */
+ if (pSeg->flags & JET_SEG_FLAG_MUTE_UPDATE)
+ {
+ result = EAS_IntSetStrmParam(easHandle, pSeg->streamHandle, PARSER_DATA_MUTE_FLAGS, (EAS_I32) pSeg->muteFlags);
+ if (result != EAS_SUCCESS)
+ return result;
+ pSeg->flags &= ~JET_SEG_FLAG_MUTE_UPDATE;
+ }
+
+ }
+ /* no repeat, start next segment */
+ else
+ {
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_Render stopping queue %d\n", index); */ }
+ startNextSegment = EAS_TRUE;
+ pSeg->state = JET_STATE_STOPPING;
+ easHandle->jetHandle->playSegment = (EAS_U8) JET_NextSegment(index);
+ }
+ }
+ break;
+
+ /* if playback has stopped, close the segment */
+ case JET_STATE_STOPPING:
+ if (state == EAS_STATE_STOPPED)
+ {
+ result = JET_CloseSegment(easHandle, index);
+ if (result != EAS_SUCCESS)
+ return result;
+ }
+ break;
+
+ case JET_STATE_READY:
+ if (startNextSegment)
+ {
+ result = JET_StartPlayback(easHandle, index);
+ if (result != EAS_SUCCESS)
+ return result;
+ startNextSegment = EAS_FALSE;
+ prepareNextSegment = EAS_TRUE;
+ }
+ break;
+
+ case JET_STATE_OPEN:
+ if (prepareNextSegment)
+ {
+ result = JET_PrepareSegment(easHandle, index);
+ if (result != EAS_SUCCESS)
+ return result;
+ prepareNextSegment = EAS_FALSE;
+ }
+ break;
+
+ case JET_STATE_PAUSED:
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_ERROR, "JET_Render: Unexpected segment state %d\n", pSeg->state); */ }
+ break;
+ }
+ }
+
+ /* increment index */
+ index = JET_NextSegment(index);
+ if (index == playIndex)
+ break;
+ }
+
+ /* if out of segments, clear playing flag */
+ if (easHandle->jetHandle->numQueuedSegments == 0)
+ easHandle->jetHandle->flags &= ~JET_FLAGS_PLAYING;
+
+ return result;
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Event()
+ *----------------------------------------------------------------------------
+ * Called from MIDI parser when data of interest is received
+ *----------------------------------------------------------------------------
+*/
+void JET_Event (EAS_DATA_HANDLE easHandle, EAS_U32 segTrack, EAS_U8 channel, EAS_U8 controller, EAS_U8 value)
+{
+ EAS_U32 event;
+
+ if (easHandle->jetHandle == NULL)
+ return;
+
+ /* handle triggers */
+ if (controller == JET_EVENT_TRIGGER_CLIP)
+ {
+ S_JET_SEGMENT *pSeg;
+ EAS_INT i;
+ EAS_U32 muteFlag;
+
+ for (i = 0; i < JET_MUTE_QUEUE_SIZE; i++)
+ {
+ /* search for event in queue */
+ if ((easHandle->jetHandle->muteQueue[i] & JET_CLIP_ID_MASK) == (value & JET_CLIP_ID_MASK))
+ {
+ /* get segment pointer and mute flag */
+ pSeg = &easHandle->jetHandle->segQueue[segTrack >> JET_EVENT_SEG_SHIFT];
+ muteFlag = 1 << ((segTrack & JET_EVENT_TRACK_MASK) >> JET_EVENT_TRACK_SHIFT);
+
+ /* un-mute the track */
+ if ((easHandle->jetHandle->muteQueue[i] & JET_CLIP_TRIGGER_FLAG) && ((value & 0x40) > 0))
+ {
+ pSeg->muteFlags &= ~muteFlag;
+ easHandle->jetHandle->muteQueue[i] &= ~JET_CLIP_TRIGGER_FLAG;
+ }
+
+ /* mute the track */
+ else
+ {
+ EAS_U32 beforeMute ;
+ beforeMute = pSeg->muteFlags ;
+ pSeg->muteFlags |= muteFlag;
+ if (beforeMute != pSeg->muteFlags)
+ easHandle->jetHandle->muteQueue[i] = 0;
+ }
+ EAS_IntSetStrmParam(easHandle, pSeg->streamHandle, PARSER_DATA_MUTE_FLAGS, (EAS_I32) pSeg->muteFlags);
+ return;
+ }
+ }
+ return;
+ }
+
+ /* generic event stuff */
+ event = (channel << JET_EVENT_CHAN_SHIFT) | (controller << JET_EVENT_CTRL_SHIFT) | value;
+
+ /* write to app queue, translate queue index to segment number */
+ if ((controller >= easHandle->jetHandle->config.appEventRangeLow) && (controller <= easHandle->jetHandle->config.appEventRangeHigh))
+ {
+
+ event |= easHandle->jetHandle->segQueue[(segTrack & JET_EVENT_SEG_MASK) >> JET_EVENT_SEG_SHIFT].userID << JET_EVENT_SEG_SHIFT;
+#ifdef DEBUG_JET
+ JET_DumpEvent("JET_Event[app]", event);
+#endif
+ JET_WriteQueue(easHandle->jetHandle->appEventQueue,
+ &easHandle->jetHandle->appEventQueueWrite,
+ easHandle->jetHandle->appEventQueueRead,
+ APP_EVENT_QUEUE_SIZE,
+ event);
+ }
+
+ /* write to JET queue */
+ else if ((controller >= JET_EVENT_LOW) && (controller <= JET_EVENT_HIGH))
+ {
+ event |= segTrack;
+#ifdef DEBUG_JET
+ JET_DumpEvent("JET_Event[jet]", event);
+#endif
+ JET_WriteQueue(easHandle->jetHandle->jetEventQueue,
+ &easHandle->jetHandle->jetEventQueueWrite,
+ easHandle->jetHandle->jetEventQueueRead,
+ JET_EVENT_QUEUE_SIZE,
+ event);
+ }
+}
+
+/*----------------------------------------------------------------------------
+ * JET_Clear_Queue()
+ *----------------------------------------------------------------------------
+ * Clears the queue and stops play without a complete shutdown
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT JET_Clear_Queue(EAS_DATA_HANDLE easHandle)
+{
+ EAS_INT index;
+ EAS_RESULT result = EAS_SUCCESS;
+
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_INFO, "JET_Clear_Queue\n"); */ }
+
+ /* pause all playing streams */
+ for (index = 0; index < SEG_QUEUE_DEPTH; index++)
+ {
+ if (easHandle->jetHandle->segQueue[index].state == JET_STATE_PLAYING)
+ {
+ result = EAS_Pause(easHandle, easHandle->jetHandle->segQueue[index].streamHandle);
+ if (result != EAS_SUCCESS)
+ return result;
+
+ easHandle->jetHandle->segQueue[index].state = JET_STATE_PAUSED;
+ }
+ }
+
+ /* close all streams */
+ for (index = 0; index < SEG_QUEUE_DEPTH; index++)
+ {
+ if (easHandle->jetHandle->segQueue[index].streamHandle != NULL)
+ {
+ result = JET_CloseSegment(easHandle, index);
+ if (result != EAS_SUCCESS)
+ return result;
+ }
+ }
+
+ easHandle->jetHandle->flags &= ~JET_FLAGS_PLAYING;
+ easHandle->jetHandle->playSegment = easHandle->jetHandle->queueSegment = 0;
+ return result;
+}
+
diff --git a/arm-wt-22k/lib_src/jet_data.h b/arm-wt-22k/lib_src/jet_data.h new file mode 100644 index 0000000..27b9cb1 --- /dev/null +++ b/arm-wt-22k/lib_src/jet_data.h @@ -0,0 +1,183 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * jet_data.h
+ *
+ * Contents and purpose:
+ * Internal data structures and interfaces for JET
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 554 $
+ * $Date: 2007-02-02 11:06:10 -0800 (Fri, 02 Feb 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef _JET_DATA_H
+#define _JET_DATA_H
+
+#include "eas.h"
+#include "jet.h"
+
+/* maximum number of segments allowed in a JET file */
+#ifndef JET_MAX_SEGMENTS
+#define JET_MAX_SEGMENTS 32
+#endif
+
+/* maximum number of DLS collections allowed in a JET file */
+#ifndef JET_MAX_DLS_COLLECTIONS
+#define JET_MAX_DLS_COLLECTIONS 4
+#endif
+
+/* maximum number of JET events in internal queue */
+#ifndef JET_EVENT_QUEUE_SIZE
+#define JET_EVENT_QUEUE_SIZE 32
+#endif
+
+/* maximum number of JET events in application queue */
+#ifndef APP_EVENT_QUEUE_SIZE
+#define APP_EVENT_QUEUE_SIZE 32
+#endif
+
+/* maximum number of active mute events */
+#ifndef JET_MUTE_QUEUE_SIZE
+#define JET_MUTE_QUEUE_SIZE 8
+#endif
+
+/*----------------------------------------------------------------------------
+ * JET event definitions
+ *----------------------------------------------------------------------------
+*/
+#define JET_EVENT_APP_LOW 80
+#define JET_EVENT_APP_HIGH 83
+#define JET_EVENT_LOW 102
+#define JET_EVENT_HIGH 119
+#define JET_EVENT_MARKER 102
+#define JET_EVENT_TRIGGER_CLIP 103
+
+#define JET_MARKER_LOOP_END 0
+
+#define JET_CLIP_ACTIVE_FLAG 0x80
+#define JET_CLIP_TRIGGER_FLAG 0x40
+#define JET_CLIP_ID_MASK 0x3f
+
+/*----------------------------------------------------------------------------
+ * JET file definitions
+ *----------------------------------------------------------------------------
+*/
+#define JET_TAG(a,b,c,d) (\
+ ( ((EAS_U32)(a) & 0xFF) << 24 ) \
+ + ( ((EAS_U32)(b) & 0xFF) << 16 ) \
+ + ( ((EAS_U32)(c) & 0xFF) << 8 ) \
+ + ( ((EAS_U32)(d) & 0xFF)))
+
+#define JET_VERSION 0x01000000
+#define JET_HEADER_TAG JET_TAG('J','E','T',' ')
+#define JET_INFO_CHUNK JET_TAG('J','I','N','F')
+#define JET_SMF_CHUNK JET_TAG('J','S','M','F')
+#define JET_DLS_CHUNK JET_TAG('J','D','L','S')
+#define INFO_JET_COPYRIGHT JET_TAG('J','C','O','P')
+#define JET_APP_DATA_CHUNK JET_TAG('J','A','P','P')
+
+#define INFO_NUM_SMF_CHUNKS JET_TAG('S','M','F','#')
+#define INFO_NUM_DLS_CHUNKS JET_TAG('D','L','S','#')
+#define INFO_JET_VERSION JET_TAG('J','V','E','R')
+
+/*----------------------------------------------------------------------------
+ * S_JET_SEGMENT
+ *
+ * JET segment data
+ *----------------------------------------------------------------------------
+*/
+typedef struct s_jet_segment_tag
+{
+ EAS_HANDLE streamHandle;
+ EAS_U32 muteFlags;
+ EAS_I16 repeatCount;
+ EAS_U8 userID;
+ EAS_I8 transpose;
+ EAS_I8 libNum;
+ EAS_U8 state;
+ EAS_U8 flags;
+} S_JET_SEGMENT;
+
+/* S_JET_SEGMENT.state */
+typedef enum
+{
+ JET_STATE_CLOSED,
+ JET_STATE_OPEN,
+ JET_STATE_READY,
+ JET_STATE_PLAYING,
+ JET_STATE_PAUSED,
+ JET_STATE_STOPPING
+} E_JET_SEGMENT_STATE;
+
+/* S_JEG_SEGMENT.flags */
+#define JET_SEG_FLAG_MUTE_UPDATE 0x01
+
+/*----------------------------------------------------------------------------
+ * S_JET_DATA
+ *
+ * Main JET data structure
+ *----------------------------------------------------------------------------
+*/
+#define SEG_QUEUE_DEPTH 3
+typedef struct s_jet_data_tag
+{
+ EAS_FILE_HANDLE jetFileHandle;
+ S_JET_SEGMENT segQueue[SEG_QUEUE_DEPTH];
+ EAS_I32 segmentOffsets[JET_MAX_SEGMENTS];
+ EAS_I32 appDataOffset;
+ EAS_I32 appDataSize;
+ EAS_DLSLIB_HANDLE libHandles[JET_MAX_DLS_COLLECTIONS];
+ EAS_U32 jetEventQueue[JET_EVENT_QUEUE_SIZE];
+ EAS_U32 appEventQueue[APP_EVENT_QUEUE_SIZE];
+ S_JET_CONFIG config;
+ EAS_U32 segmentTime;
+ EAS_U8 muteQueue[JET_MUTE_QUEUE_SIZE];
+ EAS_U8 numSegments;
+ EAS_U8 numLibraries;
+ EAS_U8 flags;
+ EAS_U8 playSegment;
+ EAS_U8 queueSegment;
+ EAS_U8 numQueuedSegments;
+ EAS_U8 jetEventQueueRead;
+ EAS_U8 jetEventQueueWrite;
+ EAS_U8 appEventQueueRead;
+ EAS_U8 appEventQueueWrite;
+} S_JET_DATA;
+
+/* flags for S_JET_DATA.flags */
+#define JET_FLAGS_PLAYING 1
+
+#define JET_EVENT_VAL_MASK 0x0000007f /* mask for value */
+#define JET_EVENT_CTRL_MASK 0x00003f80 /* mask for controller */
+#define JET_EVENT_CHAN_MASK 0x0003c000 /* mask for channel */
+#define JET_EVENT_TRACK_MASK 0x00fc0000 /* mask for track number */
+#define JET_EVENT_SEG_MASK 0xff000000 /* mask for segment ID */
+#define JET_EVENT_CTRL_SHIFT 7 /* shift for controller number */
+#define JET_EVENT_CHAN_SHIFT 14 /* shift to for MIDI channel */
+#define JET_EVENT_TRACK_SHIFT 18 /* shift to get track ID to bit 0 */
+#define JET_EVENT_SEG_SHIFT 24 /* shift to get segment ID to bit 0 */
+
+/* prototype for callback function */
+extern void JET_Event (EAS_DATA_HANDLE easHandle, EAS_U32 segTrack, EAS_U8 channel, EAS_U8 controller, EAS_U8 value);
+
+/* prototype for JET render function */
+extern EAS_PUBLIC EAS_RESULT JET_Process (EAS_DATA_HANDLE easHandle);
+
+#endif
+
diff --git a/arm-wt-22k/lib_src/wt_22khz.c b/arm-wt-22k/lib_src/wt_22khz.c new file mode 100644 index 0000000..9a06051 --- /dev/null +++ b/arm-wt-22k/lib_src/wt_22khz.c @@ -0,0 +1,11423 @@ +/*----------------------------------------------------------------------------
+ *
+ * Filename: C:\Sonic\Trunk\EASLib\WTLibrary\wt_22khz.c
+ * Source: C:\Sonic\Trunk\Wavetables\agere_wt_rev6.dls
+ * CmdLine: -w C:\Sonic\Trunk\EASLib\WTLibrary\wt_22khz.c -l C:\Sonic\Trunk\EASLib\WTLibrary\wt_22khz.log -d 0 -c C:\Sonic\Trunk\Wavetables\agere_wt_rev6.dls -w -l -d -c C:\Sonic\Trunk\Wavetables\agere_wt_rev6.dls
+ * Purpose: Wavetable sound libary
+ *
+ * Copyright (c) 2006 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 311 $
+ * $Date: 2006-09-13 15:37:50 -0700 (Wed, 13 Sep 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#include "eas_sndlib.h"
+
+/*----------------------------------------------------------------------------
+ * Articulations
+ *----------------------------------------------------------------------------
+*/
+const S_ARTICULATION eas_articulations[] =
+{
+ { /* articulation 0 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 1 */
+ { 32767, 26863, 0, 26863 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 2 */
+ { 32767, 30484, 0, 30668 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 3 */
+ { 32767, 26439, 0, 26439 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 4 */
+ { 32767, 0, 32767, 32715 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 5 */
+ { 32767, 21333, 0, 21333 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 6 */
+ { 32767, 31882, 0, 31938 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 7 */
+ { 32767, 32663, 32767, 32663 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 8 */
+ { 32767, 0, 32767, 0 },
+ { 32767, 1902, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 9 */
+ { 32767, 32349, 0, 32349 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 10 */
+ { 32767, 0, 32767, 17213 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -1
+ },
+ { /* articulation 11 */
+ { 32767, 32072, 0, 32072 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -56
+ },
+ { /* articulation 12 */
+ { 32767, 23749, 0, 23749 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 56
+ },
+ { /* articulation 13 */
+ { 32767, 32010, 0, 32010 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -31
+ },
+ { /* articulation 14 */
+ { 9511, 21333, 0, 21333 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 56
+ },
+ { /* articulation 15 */
+ { 32767, 31844, 0, 31844 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -6
+ },
+ { /* articulation 16 */
+ { 32767, 32123, 0, 32194 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 56
+ },
+ { /* articulation 17 */
+ { 32767, 31730, 0, 31730 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 6
+ },
+ { /* articulation 18 */
+ { 32767, 31391, 0, 31391 },
+ { 32767, 951, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 31
+ },
+ { /* articulation 19 */
+ { 32767, 31964, 0, 31964 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 25
+ },
+ { /* articulation 20 */
+ { 32767, 31056, 0, 31056 },
+ { 32767, 951, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 63
+ },
+ { /* articulation 21 */
+ { 32767, 32289, 0, 32271 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -37
+ },
+ { /* articulation 22 */
+ { 19021, 31882, 0, 31911 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -37
+ },
+ { /* articulation 23 */
+ { 32767, 31988, 0, 32032 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -37
+ },
+ { /* articulation 24 */
+ { 32767, 0, 32767, 32663 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 12
+ },
+ { /* articulation 25 */
+ { 32767, 31352, 0, 31352 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -25
+ },
+ { /* articulation 26 */
+ { 32767, 0, 32767, 32663 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 25
+ },
+ { /* articulation 27 */
+ { 32767, 31817, 0, 31781 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -25
+ },
+ { /* articulation 28 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 95, 0, 0 },
+ 0, 0, 951, 240, 0, 0, 0, 0, -56
+ },
+ { /* articulation 29 */
+ { 32767, 32230, 0, 32218 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -37
+ },
+ { /* articulation 30 */
+ { 32767, 26439, 0, 26439 },
+ { 32767, 3804, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 50
+ },
+ { /* articulation 31 */
+ { 32767, 23749, 0, 23749 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -50
+ },
+ { /* articulation 32 */
+ { 32767, 29434, 0, 29434 },
+ { 32767, 3804, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -50
+ },
+ { /* articulation 33 */
+ { 32767, 30240, 0, 30234 },
+ { 32767, 3804, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -44
+ },
+ { /* articulation 34 */
+ { 32767, 32558, 0, 32558 },
+ { 32767, 254, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 25
+ },
+ { /* articulation 35 */
+ { 32767, 0, 32767, 32663 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -63
+ },
+ { /* articulation 36 */
+ { 3804, 23749, 0, 23749 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -63
+ },
+ { /* articulation 37 */
+ { 32767, 23749, 0, 23749 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -59
+ },
+ { /* articulation 38 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 50
+ },
+ { /* articulation 39 */
+ { 32767, 28809, 0, 28809 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 44
+ },
+ { /* articulation 40 */
+ { 1902, 30725, 0, 30725 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 951, -100, 0, 0, 0, 0, 44
+ },
+ { /* articulation 41 */
+ { 32767, 9042, 0, 9042 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 25
+ },
+ { /* articulation 42 */
+ { 32767, 29889, 0, 29889 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 63
+ },
+ { /* articulation 43 */
+ { 32767, 30240, 0, 30234 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 63
+ },
+ { /* articulation 44 */
+ { 19021, 19970, 0, 19970 },
+ { 951, 32767, 32767, 0 },
+ 0, 0, 951, 100, 0, 0, 0, 0, -25
+ },
+ { /* articulation 45 */
+ { 3804, 17213, 0, 17213 },
+ { 951, 32767, 32767, 0 },
+ 0, 0, 951, 500, 0, 0, 0, 0, -25
+ },
+ { /* articulation 46 */
+ { 32767, 17213, 0, 17213 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -56
+ },
+ { /* articulation 47 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, -56
+ },
+ { /* articulation 48 */
+ { 32767, 0, 32767, 0 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 49 */
+ { 32767, 31180, 0, 31180 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 50 */
+ { 19021, 31964, 0, 32071 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 51 */
+ { 32767, 29669, 0, 29669 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 52 */
+ { 32767, 31742, 0, 31352 },
+ { 32767, 294, 0, 0 },
+ 0, 0, 951, 0, 10000, 7121, 0, 0, 0
+ },
+ { /* articulation 53 */
+ { 32767, 32558, 0, 26439 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 54 */
+ { 32767, 32349, 0, 26439 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 55 */
+ { 32767, 32072, 0, 32072 },
+ { 32767, 95, 0, 0 },
+ 0, 34, 989, 0, 2400, 9521, 0, 0, 0
+ },
+ { /* articulation 56 */
+ { 32767, 30234, 0, 30234 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 11738, 0, 16, 0
+ },
+ { /* articulation 57 */
+ { 32767, 32349, 0, 30073 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 58 */
+ { 32767, 31730, 0, 31476 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 59 */
+ { 32767, 32418, 0, 25329 },
+ { 32767, 95, 0, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 60 */
+ { 32767, 32052, 0, 31964 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 61 */
+ { 32767, 31938, 0, 31938 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 62 */
+ { 9511, 32663, 18820, 23749 },
+ { 1902, 57, 13107, 0 },
+ 0, 0, 989, 0, 6000, 5535, 0, 4, 0
+ },
+ { /* articulation 63 */
+ { 32767, 31754, 0, 31730 },
+ { 32767, 1902, 0, 0 },
+ 0, 52, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 64 */
+ { 127, 32686, 3811, 32349 },
+ { 95, 38, 32767, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 65 */
+ { 4755, 32663, 3566, 28809 },
+ { 3804, 32767, 32767, 0 },
+ 0, 0, 989, 100, 0, 11919, 0, 0, 0
+ },
+ { /* articulation 66 */
+ { 32767, 31935, 0, 31935 },
+ { 32767, 335, 0, 0 },
+ 0, 17, 989, 0, 7000, 9023, 0, 0, 0
+ },
+ { /* articulation 67 */
+ { 32767, 31391, 0, 31391 },
+ { 32767, 335, 0, 0 },
+ 0, 2, 951, 0, 7000, 9023, 0, 0, 0
+ },
+ { /* articulation 68 */
+ { 32767, 32628, 6208, 31935 },
+ { 380, 95, 0, 0 },
+ 0, 0, 989, 0, 3840, 8302, 0, 8, 0
+ },
+ { /* articulation 69 */
+ { 32767, 32072, 0, 32171 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 989, 0, 5000, 8321, 0, 0, 0
+ },
+ { /* articulation 70 */
+ { 32767, 31935, 0, 31935 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 951, 0, 5000, 7934, 0, 0, 0
+ },
+ { /* articulation 71 */
+ { 32767, 32117, 0, 30685 },
+ { 32767, 63, 0, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 72 */
+ { 32767, 32245, 0, 23749 },
+ { 32767, 1902, 0, 0 },
+ 0, 172, 989, 0, 1000, 11107, 0, 0, 0
+ },
+ { /* articulation 73 */
+ { 32767, 32663, 6208, 31935 },
+ { 95, 95, 0, 0 },
+ 0, 34, 1622, 0, 3560, 8834, 1, 8, 0
+ },
+ { /* articulation 74 */
+ { 32767, 32362, 0, 26439 },
+ { 32767, 190, 0, 0 },
+ 0, 17, 989, 0, 6000, 9907, 0, 0, 0
+ },
+ { /* articulation 75 */
+ { 32767, 32245, 0, 23749 },
+ { 32767, 63, 0, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 76 */
+ { 32767, 31730, 18820, 9042 },
+ { 32767, 32767, 32767, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 77 */
+ { 32767, 32715, 128, 32168 },
+ { 32767, 127, 0, 0 },
+ 0, 0, 989, 0, 0, 11920, 0, 8, 0
+ },
+ { /* articulation 78 */
+ { 32767, 32072, 0, 32072 },
+ { 32767, 67, 0, 0 },
+ 3, 0, 572, 0, 5000, 5535, 0, 0, 0
+ },
+ { /* articulation 79 */
+ { 3804, 32663, 18820, 23749 },
+ { 32767, 2024, 0, 0 },
+ 10, 34, 1008, -30, 0, 0, 0, 0, 0
+ },
+ { /* articulation 80 */
+ { 19021, 32663, 18820, 23749 },
+ { 761, 95, 0, 0 },
+ 0, 34, 989, 0, 4473, 7131, 0, 8, 0
+ },
+ { /* articulation 81 */
+ { 1902, 32628, 6208, 32171 },
+ { 634, 38, 16384, 0 },
+ 0, 0, 989, 0, 2987, 7877, 0, 12, 0
+ },
+ { /* articulation 82 */
+ { 32767, 32593, 0, 31935 },
+ { 32767, 95, 0, 0 },
+ 0, 0, 1162, 0, 4053, 7930, 2, 12, 0
+ },
+ { /* articulation 83 */
+ { 380, 32684, 6208, 31935 },
+ { 32767, 112, 0, 0 },
+ 0, 0, 989, 0, 0, 8887, 0, 0, 0
+ },
+ { /* articulation 84 */
+ { 19021, 32663, 18820, 23749 },
+ { 1268, 95, 0, 0 },
+ 0, 34, 989, 0, 5113, 7981, 0, 4, 0
+ },
+ { /* articulation 85 */
+ { 1902, 32663, 6208, 30725 },
+ { 1902, 127, 0, 0 },
+ 0, 34, 989, 0, 3500, 7877, 0, 5, 0
+ },
+ { /* articulation 86 */
+ { 1902, 32663, 6208, 30725 },
+ { 1268, 95, 0, 0 },
+ 0, 34, 951, 0, 4773, 8355, 0, 5, 0
+ },
+ { /* articulation 87 */
+ { 476, 32663, 10809, 31935 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 88 */
+ { 3804, 32663, 18820, 30234 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 89 */
+ { 7608, 32663, 18820, 17213 },
+ { 2536, 261, 0, 0 },
+ 0, 34, 989, 0, 1200, 11690, 0, 4, 0
+ },
+ { /* articulation 90 */
+ { 32767, 32468, 15076, 30234 },
+ { 32767, 32767, 32767, 0 },
+ 0, 36, 2183, 0, 0, 11919, 1, 0, 0
+ },
+ { /* articulation 91 */
+ { 32767, 0, 32767, 32663 },
+ { 380, 32767, 32767, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 92 */
+ { 32767, 31391, 0, 31391 },
+ { 32767, 634, 0, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 93 */
+ { 32767, 0, 32767, 31742 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 989, 0, -2375, 11920, 0, 0, 0
+ },
+ { /* articulation 94 */
+ { 3804, 0, 32767, 31730 },
+ { 1902, 127, 0, 0 },
+ 0, 0, 989, 0, 1200, 9121, 0, 8, 0
+ },
+ { /* articulation 95 */
+ { 32767, 32663, 15076, 30725 },
+ { 1902, 634, 0, 0 },
+ 0, 0, 989, 0, 4133, 10925, 0, 0, 0
+ },
+ { /* articulation 96 */
+ { 32767, 32558, 128, 30725 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 989, 0, 6000, 8321, 0, 0, 0
+ },
+ { /* articulation 97 */
+ { 3804, 0, 32767, 30725 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 98 */
+ { 32767, 32558, 0, 23749 },
+ { 1268, 190, 13107, 0 },
+ 0, 34, 989, 0, 3200, 8321, 0, 0, 0
+ },
+ { /* articulation 99 */
+ { 32767, 0, 32767, 0 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 100 */
+ { 32767, 32072, 0, 23749 },
+ { 32767, 1087, 0, 0 },
+ 0, 34, 989, 0, 8187, 5535, 0, 5, 0
+ },
+ { /* articulation 101 */
+ { 32767, 32558, 0, 29434 },
+ { 32767, 190, 7667, 0 },
+ 5, 0, 989, 0, 6053, 5535, 0, 5, 0
+ },
+ { /* articulation 102 */
+ { 32767, 32663, 18820, 23749 },
+ { 1902, 95, 0, 0 },
+ 0, 0, 989, 0, 2700, 9852, 0, 0, 0
+ },
+ { /* articulation 103 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 1268, 0, 0 },
+ 0, 52, 951, 0, 2500, 10490, 1, 8, 0
+ },
+ { /* articulation 104 */
+ { 32767, 32663, 23493, 23749 },
+ { 32767, 380, 0, 0 },
+ 0, 34, 988, 0, 4000, 10223, 1, 4, 0
+ },
+ { /* articulation 105 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 126, 7667, 0 },
+ 0, 0, 989, 0, 1813, 9154, 0, 0, 0
+ },
+ { /* articulation 106 */
+ { 32767, 31730, 0, 31730 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 107 */
+ { 32767, 31180, 0, 30484 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 11690, 0, 0, 0
+ },
+ { /* articulation 108 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 109 */
+ { 32767, 32349, 18820, 27897 },
+ { 32767, 95, 0, 0 },
+ 12, 34, 951, 0, 3000, 10223, 0, 0, 0
+ },
+ { /* articulation 110 */
+ { 32767, 32349, 18820, 27897 },
+ { 32767, 63, 0, 0 },
+ 12, 34, 951, 0, 1900, 10031, 0, 0, 0
+ },
+ { /* articulation 111 */
+ { 32767, 32663, 18820, 26439 },
+ { 32767, 63, 0, 0 },
+ 12, 34, 988, 0, 1000, 11107, 0, 0, 0
+ },
+ { /* articulation 112 */
+ { 32767, 32663, 18820, 26439 },
+ { 32767, 63, 0, 0 },
+ 12, 34, 988, 0, 2000, 11107, 0, 0, 0
+ },
+ { /* articulation 113 */
+ { 32767, 32505, 0, 26439 },
+ { 32767, 190, 0, 0 },
+ 0, 17, 989, 0, 4000, 8321, 0, 0, 0
+ },
+ { /* articulation 114 */
+ { 32767, 31832, 19893, 9042 },
+ { 32767, 476, 0, 0 },
+ 0, 34, 1452, 0, 0, 11919, 0, 0, 0
+ },
+ { /* articulation 115 */
+ { 19021, 32072, 23493, 9042 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 1355, 0, 0, 11877, 1, 0, 0
+ },
+ { /* articulation 116 */
+ { 32767, 32468, 0, 23749 },
+ { 32767, 190, 0, 0 },
+ 0, 34, 989, 0, 3500, 9023, 0, 0, 0
+ },
+ { /* articulation 117 */
+ { 32767, 17213, 23493, 0 },
+ { 32767, 32767, 32767, 0 },
+ 0, 17, 1521, 0, 0, 10925, 1, 0, 0
+ },
+ { /* articulation 118 */
+ { 32767, 32505, 0, 26439 },
+ { 32767, 190, 0, 0 },
+ 0, 52, 989, 0, 3200, 8721, 0, 4, 0
+ },
+ { /* articulation 119 */
+ { 3804, 32663, 18820, 23749 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 0, 1, 0, 0
+ },
+ { /* articulation 120 */
+ { 9511, 32663, 18820, 25329 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 11877, 0, 8, 0
+ },
+ { /* articulation 121 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 32, 0, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 122 */
+ { 32767, 32558, 0, 23749 },
+ { 32767, 380, 0, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 123 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 24, 0, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 124 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 989, 0, 3000, 10223, 0, 8, 0
+ },
+ { /* articulation 125 */
+ { 127, 0, 32767, 32349 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 1522, 0, 0, 11423, 4, 0, 0
+ },
+ { /* articulation 126 */
+ { 951, 32422, 0, 32387 },
+ { 32767, 19, 0, 0 },
+ 0, 0, 989, 0, 0, 11423, 0, 0, 0
+ },
+ { /* articulation 127 */
+ { 391, 0, 0, 31180 },
+ { 190, 32767, 32767, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 128 */
+ { 32767, 30725, 0, 30725 },
+ { 32767, 761, 0, 0 },
+ 0, 0, 989, 1200, 0, 0, 0, 0, 0
+ },
+ { /* articulation 129 */
+ { 32767, 31730, 0, 31935 },
+ { 32767, 380, 0, 0 },
+ 0, 0, 989, 50, 0, 0, 0, 0, 0
+ },
+ { /* articulation 130 */
+ { 32767, 32072, 0, 32072 },
+ { 32767, 19021, 0, 0 },
+ 0, 0, 989, 0, 4700, 7769, 0, 0, 0
+ },
+ { /* articulation 131 */
+ { 32767, 30073, 0, 30073 },
+ { 32767, 32767, 0, 0 },
+ 0, 0, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 132 */
+ { 32767, 32558, 32767, 32558 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 133 */
+ { 32767, 32663, 18820, 17213 },
+ { 32767, 190, 0, 0 },
+ 10, 34, 951, 0, 2000, 10696, 0, 0, 0
+ },
+ { /* articulation 134 */
+ { 32767, 32663, 10809, 17213 },
+ { 32767, 190, 0, 0 },
+ 12, 34, 982, 0, 0, 10910, 0, 0, 0
+ },
+ { /* articulation 135 */
+ { 32767, 32663, 18820, 17213 },
+ { 32767, 190, 0, 0 },
+ 10, 34, 951, 0, 1200, 10218, 0, 0, 0
+ },
+ { /* articulation 136 */
+ { 32767, 32663, 18820, 17213 },
+ { 32767, 190, 0, 0 },
+ 10, 34, 951, 0, 1100, 9525, 0, 0, 0
+ },
+ { /* articulation 137 */
+ { 19021, 32558, 18820, 23749 },
+ { 32767, 19, 0, 0 },
+ 10, 34, 988, 0, 2000, 10962, 0, 0, 0
+ },
+ { /* articulation 138 */
+ { 32767, 32349, 18820, 23749 },
+ { 19021, 634, 0, 0 },
+ 10, 32, 1008, 0, 1200, 10090, 0, 0, 0
+ },
+ { /* articulation 139 */
+ { 2536, 0, 32767, 27897 },
+ { 1902, 380, 0, 0 },
+ 7, 34, 988, 0, 1620, 8933, 0, 0, 0
+ },
+ { /* articulation 140 */
+ { 32767, 32349, 10809, 23749 },
+ { 32767, 380, 0, 0 },
+ 7, 34, 988, 0, 2200, 8994, 0, 0, 0
+ },
+ { /* articulation 141 */
+ { 32767, 32663, 15076, 23749 },
+ { 32767, 1902, 0, 0 },
+ 10, 34, 982, 0, 2500, 9525, 0, 0, 0
+ },
+ { /* articulation 142 */
+ { 32767, 32663, 15076, 23749 },
+ { 32767, 190, 0, 0 },
+ 10, 34, 951, 0, 1500, 11423, 0, 0, 0
+ },
+ { /* articulation 143 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 1902, 0, 0 },
+ 9, 34, 982, 0, 1500, 9521, 0, 0, 0
+ },
+ { /* articulation 144 */
+ { 3804, 0, 32767, 28809 },
+ { 32767, 32767, 32767, 0 },
+ 0, 0, 1521, 0, 0, 9521, 0, 0, 0
+ },
+ { /* articulation 145 */
+ { 32767, 32558, 0, 23749 },
+ { 32767, 19021, 0, 0 },
+ 0, 17, 989, 0, 5000, 10223, 0, 0, 0
+ },
+ { /* articulation 146 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 63, 0, 0 },
+ 10, 34, 951, 0, 1500, 9907, 0, 0, 0
+ },
+ { /* articulation 147 */
+ { 32767, 32698, 11682, 23749 },
+ { 32767, 1902, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 148 */
+ { 32767, 32072, 0, 32072 },
+ { 32767, 380, 0, 0 },
+ 0, 17, 989, 0, 3440, 9260, 0, 0, 0
+ },
+ { /* articulation 149 */
+ { 32767, 30234, 0, 30725 },
+ { 32767, 1902, 0, 0 },
+ 0, 17, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 150 */
+ { 32767, 31730, 0, 30725 },
+ { 32767, 380, 0, 0 },
+ 0, 17, 989, 0, 4000, 7823, 0, 0, 0
+ },
+ { /* articulation 151 */
+ { 32767, 32558, 3566, 23749 },
+ { 783, 32767, 32767, 0 },
+ 100, 0, 1522, 500, 0, 11877, 0, 0, 0
+ },
+ { /* articulation 152 */
+ { 32767, 32663, 18820, 17213 },
+ { 32767, 1902, 0, 0 },
+ 8, 34, 989, -22, 0, 0, 0, 0, 0
+ },
+ { /* articulation 153 */
+ { 19021, 29007, 6784, 23749 },
+ { 32767, 1902, 0, 0 },
+ 0, 34, 951, 0, 5000, 9521, 1, 0, 0
+ },
+ { /* articulation 154 */
+ { 32767, 32558, 0, 31935 },
+ { 1902, 254, 16384, 0 },
+ 0, 52, 989, 0, 3627, 10547, 0, 5, 0
+ },
+ { /* articulation 155 */
+ { 3804, 0, 32767, 23749 },
+ { 1902, 1902, 0, 0 },
+ 0, 34, 989, 27, 0, 11919, 0, 0, 0
+ },
+ { /* articulation 156 */
+ { 32767, 0, 32767, 31730 },
+ { 76, 66, 10092, 0 },
+ 5, 0, 989, 0, 8007, 5535, 0, 8, 0
+ },
+ { /* articulation 157 */
+ { 32767, 32468, 0, 29434 },
+ { 32767, 127, 0, 0 },
+ 0, 52, 989, 0, 2500, 9032, 0, 0, 0
+ },
+ { /* articulation 158 */
+ { 9511, 32663, 10809, 25329 },
+ { 32767, 32767, 32767, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 159 */
+ { 19021, 32558, 18820, 23749 },
+ { 32767, 190, 0, 0 },
+ 10, 34, 988, 0, 2600, 9513, 0, 0, 0
+ },
+ { /* articulation 160 */
+ { 32767, 32106, 9568, 23749 },
+ { 2348, 391, 0, 0 },
+ 10, 52, 980, 0, 6500, 9023, 0, 0, 0
+ },
+ { /* articulation 161 */
+ { 32767, 32558, 0, 26439 },
+ { 32767, 63, 0, 0 },
+ 0, 34, 989, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 162 */
+ { 3804, 0, 32767, 31730 },
+ { 32767, 76, 0, 0 },
+ 0, 0, 989, 0, 6000, 6503, 0, 4, 0
+ },
+ { /* articulation 163 */
+ { 3804, 0, 32767, 30725 },
+ { 32767, 63, 0, 0 },
+ 0, 34, 989, 0, 1500, 11107, 0, 4, 0
+ },
+ { /* articulation 164 */
+ { 32767, 32072, 15076, 17213 },
+ { 32767, 1268, 0, 0 },
+ 0, 0, 951, 0, 2000, 10223, 0, 0, 0
+ },
+ { /* articulation 165 */
+ { 32767, 32558, 0, 23749 },
+ { 32767, 380, 0, 0 },
+ 0, 34, 989, 0, 3000, 9366, 0, 0, 0
+ },
+ { /* articulation 166 */
+ { 32767, 32663, 18820, 23749 },
+ { 1902, 127, 10879, 0 },
+ 0, 0, 989, 0, 6000, 7121, 0, 4, 0
+ },
+ { /* articulation 167 */
+ { 32767, 32505, 0, 26439 },
+ { 32767, 21, 0, 0 },
+ 0, 52, 989, 0, 3500, 6236, 0, 5, 0
+ },
+ { /* articulation 168 */
+ { 32767, 32505, 0, 26439 },
+ { 32767, 190, 0, 0 },
+ 0, 52, 989, 0, 2800, 7121, 0, 0, 0
+ },
+ { /* articulation 169 */
+ { 32767, 32418, 0, 29434 },
+ { 32767, 127, 0, 0 },
+ 0, 52, 989, 0, 2100, 9626, 0, 0, 0
+ },
+ { /* articulation 170 */
+ { 32767, 32349, 0, 30234 },
+ { 32767, 127, 0, 0 },
+ 0, 52, 989, 0, 3000, 9626, 0, 0, 0
+ },
+ { /* articulation 171 */
+ { 32767, 32288, 0, 28400 },
+ { 32767, 127, 0, 0 },
+ 0, 52, 989, 0, 1000, 9032, 0, 0, 0
+ },
+ { /* articulation 172 */
+ { 32767, 32072, 0, 28809 },
+ { 32767, 127, 0, 0 },
+ 0, 52, 989, 0, 1000, 9032, 0, 0, 0
+ },
+ { /* articulation 173 */
+ { 3804, 32072, 15076, 17213 },
+ { 32767, 1268, 0, 0 },
+ 0, 52, 991, 0, 0, 11107, 0, 8, 0
+ },
+ { /* articulation 174 */
+ { 32767, 32349, 15076, 23749 },
+ { 7608, 147, 0, 0 },
+ 0, 0, 989, 0, 4500, 9521, 0, 8, 0
+ },
+ { /* articulation 175 */
+ { 32767, 32663, 18820, 23749 },
+ { 32767, 95, 0, 0 },
+ 0, 0, 989, 0, 2000, 8321, 0, 8, 0
+ },
+ { /* articulation 176 */
+ { 32767, 32715, 128, 29669 },
+ { 32767, 1729, 0, 0 },
+ 0, 0, 989, 0, 6000, 7823, 0, 8, 0
+ },
+ { /* articulation 177 */
+ { 19021, 32448, 0, 31882 },
+ { 32767, 95, 0, 0 },
+ 0, 0, 989, 0, 4500, 7121, 0, 8, 0
+ },
+ { /* articulation 178 */
+ { 32767, 32560, 3646, 32107 },
+ { 32767, 190, 0, 0 },
+ 0, 0, 989, 0, 4000, 8321, 0, 8, 0
+ },
+ { /* articulation 179 */
+ { 32767, 32602, 13644, 26439 },
+ { 32767, 63, 0, 0 },
+ 12, 34, 988, 0, 2000, 11107, 0, 0, 0
+ },
+ { /* articulation 180 */
+ { 19021, 30484, 0, 23749 },
+ { 32767, 1902, 0, 0 },
+ 0, 0, 989, 0, 5000, 8321, 1, 0, 0
+ },
+ { /* articulation 181 */
+ { 261, 32466, 0, 31938 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ },
+ { /* articulation 182 */
+ { 32767, 32418, 0, 31742 },
+ { 2348, 39, 0, 0 },
+ 0, 34, 989, 0, 3600, 7121, 0, 4, 0
+ },
+ { /* articulation 183 */
+ { 32767, 32090, 0, 32090 },
+ { 32767, 634, 0, 0 },
+ 0, 34, 951, 0, 0, 0, 0, 0, 0
+ }
+}; /*end Articulations */
+
+/*----------------------------------------------------------------------------
+ * Regions
+ *----------------------------------------------------------------------------
+*/
+const S_WT_REGION eas_regions[] =
+{
+ { { 0, 27, 27 }, -2868, 16422, 0, 0, 78, 0 }, /* region 0 */
+ { { 0, 28, 28 }, -3568, 32767, 0, 0, 36, 0 }, /* region 1 */
+ { { 0, 29, 29 }, -4553, 32767, 0, 0, 28, 1 }, /* region 2 */
+ { { 0, 30, 30 }, -4853, 32767, 0, 0, 28, 2 }, /* region 3 */
+ { { 0, 31, 31 }, -3868, 23197, 0, 0, 45, 3 }, /* region 4 */
+ { { 1536, 32, 32 }, -3200, 20675, 0, 0, 134, 4 }, /* region 5 */
+ { { 1537, 33, 33 }, -3703, 20675, 792, 879, 47, 5 }, /* region 6 */
+ { { 1537, 34, 34 }, -3803, 16422, 792, 879, 47, 6 }, /* region 7 */
+ { { 0, 35, 35 }, -4968, 32767, 0, 0, 79, 7 }, /* region 8 */
+ { { 0, 36, 36 }, -4968, 32767, 0, 0, 79, 7 }, /* region 9 */
+ { { 0, 37, 37 }, -4051, 18426, 0, 0, 50, 8 }, /* region 10 */
+ { { 0, 38, 38 }, -4151, 23197, 0, 0, 12, 9 }, /* region 11 */
+ { { 0, 39, 39 }, -4668, 23197, 0, 0, 37, 10 }, /* region 12 */
+ { { 0, 40, 40 }, -4151, 23197, 0, 0, 12, 4 }, /* region 13 */
+ { { 1, 41, 41 }, -5855, 26028, 798, 993, 42, 11 }, /* region 14 */
+ { { 257, 42, 42 }, -4200, 26028, 4288, 7488, 4, 12 }, /* region 15 */
+ { { 1, 43, 43 }, -5755, 26028, 798, 993, 42, 13 }, /* region 16 */
+ { { 257, 44, 44 }, -4400, 26028, 4288, 7488, 4, 14 }, /* region 17 */
+ { { 1, 45, 45 }, -5755, 26028, 798, 993, 42, 15 }, /* region 18 */
+ { { 257, 46, 46 }, -4600, 26028, 4288, 7488, 4, 16 }, /* region 19 */
+ { { 1, 47, 47 }, -5455, 26028, 798, 993, 42, 17 }, /* region 20 */
+ { { 1, 48, 48 }, -5355, 26028, 798, 993, 42, 18 }, /* region 21 */
+ { { 1, 49, 49 }, -5200, 16422, 1294, 5778, 5, 19 }, /* region 22 */
+ { { 1, 50, 50 }, -5255, 26028, 798, 993, 42, 20 }, /* region 23 */
+ { { 1, 51, 51 }, -5268, 16422, 6592, 9921, 3, 21 }, /* region 24 */
+ { { 1, 52, 52 }, -5600, 32767, 1294, 5778, 5, 22 }, /* region 25 */
+ { { 1, 53, 53 }, -5418, 14636, 6592, 9921, 3, 23 }, /* region 26 */
+ { { 0, 54, 54 }, -5751, 26028, 0, 0, 35, 24 }, /* region 27 */
+ { { 1, 55, 55 }, -5300, 32767, 1294, 5778, 5, 25 }, /* region 28 */
+ { { 0, 56, 56 }, -7255, 32767, 0, 0, 88, 26 }, /* region 29 */
+ { { 1, 57, 57 }, -5700, 32767, 1294, 5778, 5, 27 }, /* region 30 */
+ { { 1, 58, 58 }, -7053, 23197, 0, 166, 110, 28 }, /* region 31 */
+ { { 1, 59, 59 }, -5968, 16422, 6592, 9921, 3, 29 }, /* region 32 */
+ { { 1, 60, 60 }, -6453, 23197, 432, 582, 60, 30 }, /* region 33 */
+ { { 1, 61, 61 }, -6853, 16422, 432, 582, 60, 30 }, /* region 34 */
+ { { 1, 62, 62 }, -7253, 20675, 432, 582, 60, 31 }, /* region 35 */
+ { { 1, 63, 63 }, -7353, 23197, 432, 582, 60, 32 }, /* region 36 */
+ { { 1, 64, 64 }, -7953, 23197, 432, 582, 60, 33 }, /* region 37 */
+ { { 0, 65, 65 }, -7555, 32767, 0, 0, 10, 34 }, /* region 38 */
+ { { 0, 66, 66 }, -7955, 20675, 0, 0, 10, 34 }, /* region 39 */
+ { { 512, 67, 67 }, -7155, 18426, 0, 0, 88, 35 }, /* region 40 */
+ { { 512, 68, 68 }, -7755, 18426, 0, 0, 88, 35 }, /* region 41 */
+ { { 0, 69, 69 }, -7755, 32767, 0, 0, 83, 36 }, /* region 42 */
+ { { 0, 70, 70 }, -6855, 21900, 0, 0, 83, 37 }, /* region 43 */
+ { { 769, 71, 71 }, -6355, 23197, 0, 1226, 31, 38 }, /* region 44 */
+ { { 769, 72, 72 }, -6955, 26028, 0, 1226, 31, 38 }, /* region 45 */
+ { { 1024, 73, 73 }, -7955, 32767, 0, 0, 18, 39 }, /* region 46 */
+ { { 1024, 74, 74 }, -8455, 32767, 0, 0, 18, 40 }, /* region 47 */
+ { { 1, 75, 75 }, -7900, 23197, 0, 31, 136, 41 }, /* region 48 */
+ { { 0, 76, 76 }, -10455, 23197, 0, 0, 131, 42 }, /* region 49 */
+ { { 0, 77, 77 }, -10055, 23197, 0, 0, 131, 43 }, /* region 50 */
+ { { 0, 78, 78 }, -8853, 16422, 0, 0, 86, 44 }, /* region 51 */
+ { { 0, 79, 79 }, -10253, 16422, 0, 0, 86, 45 }, /* region 52 */
+ { { 1281, 80, 80 }, -6300, 13045, 209, 230, 100, 46 }, /* region 53 */
+ { { 1281, 81, 81 }, -6400, 16422, 209, 230, 100, 47 }, /* region 54 */
+ { { 0, 82, 82 }, -8455, 20675, 0, 0, 84, 48 }, /* region 55 */
+ { { 0, 83, 83 }, -8900, 32767, 0, 0, 9, 49 }, /* region 56 */
+ { { 1, 84, 84 }, -8400, 23197, 0, 10294, 2, 50 }, /* region 57 */
+ { { 0, 85, 85 }, -9655, 32767, 0, 0, 132, 4 }, /* region 58 */
+ { { 0, 86, 86 }, -9068, 16422, 0, 0, 20, 51 }, /* region 59 */
+ { { 32769, 87, 87 }, -9168, 32767, 1335, 1603, 20, 52 }, /* region 60 */
+ { { 1, 12, 43 }, -3055, 23197, 920, 1383, 26, 53 }, /* region 61 */
+ { { 32769, 44, 96 }, -5060, 18426, 885, 1176, 33, 53 }, /* region 62 */
+ { { 1, 12, 48 }, -3461, 18426, 1148, 1514, 22, 54 }, /* region 63 */
+ { { 32769, 49, 96 }, -6253, 16422, 1347, 1420, 25, 54 }, /* region 64 */
+ { { 1, 33, 56 }, -5600, 26028, 1064, 1170, 34, 55 }, /* region 65 */
+ { { 1, 57, 72 }, -6000, 26028, 930, 1014, 41, 55 }, /* region 66 */
+ { { 32769, 73, 108 }, -7600, 26028, 726, 826, 49, 55 }, /* region 67 */
+ { { 1, 36, 96 }, -7600, 20675, 635, 735, 55, 56 }, /* region 68 */
+ { { 32769, 97, 108 }, -10108, 13045, 0, 31, 136, 56 }, /* region 69 */
+ { { 1, 36, 96 }, -7600, 14636, 635, 735, 55, 0 }, /* region 70 */
+ { { 32769, 97, 108 }, -10108, 13045, 0, 31, 136, 0 }, /* region 71 */
+ { { 1, 36, 83 }, -6006, 13045, 838, 922, 44, 57 }, /* region 72 */
+ { { 1, 84, 93 }, -8406, 14636, 209, 230, 100, 57 }, /* region 73 */
+ { { 32769, 94, 108 }, -10108, 13045, 0, 31, 136, 57 }, /* region 74 */
+ { { 1, 36, 83 }, -6006, 13045, 838, 922, 44, 58 }, /* region 75 */
+ { { 1, 84, 93 }, -8406, 13045, 209, 230, 100, 58 }, /* region 76 */
+ { { 32769, 94, 108 }, -10108, 13045, 0, 31, 136, 58 }, /* region 77 */
+ { { 1, 21, 56 }, -5595, 23197, 1064, 1170, 34, 59 }, /* region 78 */
+ { { 1, 57, 72 }, -5995, 23197, 930, 1014, 41, 59 }, /* region 79 */
+ { { 32769, 73, 108 }, -7598, 23197, 726, 826, 49, 59 }, /* region 80 */
+ { { 1, 12, 83 }, -6006, 16422, 838, 922, 44, 60 }, /* region 81 */
+ { { 1, 84, 93 }, -8406, 16422, 209, 230, 100, 60 }, /* region 82 */
+ { { 32769, 94, 108 }, -10108, 16422, 0, 31, 136, 60 }, /* region 83 */
+ { { 1, 24, 83 }, -6006, 16422, 838, 922, 44, 61 }, /* region 84 */
+ { { 1, 84, 93 }, -8406, 16422, 209, 230, 100, 61 }, /* region 85 */
+ { { 32769, 94, 108 }, -10108, 16422, 0, 31, 136, 61 }, /* region 86 */
+ { { 1, 12, 83 }, -6020, 16422, 0, 83, 123, 62 }, /* region 87 */
+ { { 1, 84, 90 }, -8482, 16422, 0, 20, 142, 62 }, /* region 88 */
+ { { 32769, 91, 108 }, -9101, 16422, 6, 20, 143, 62 }, /* region 89 */
+ { { 1, 48, 75 }, -7241, 20675, 419, 460, 73, 63 }, /* region 90 */
+ { { 32769, 76, 108 }, -9690, 18426, 254, 264, 98, 63 }, /* region 91 */
+ { { 32769, 36, 84 }, -7755, 16422, 0, 2775, 13, 64 }, /* region 92 */
+ { { 32769, 12, 108 }, -6655, 23197, 30, 276, 97, 65 }, /* region 93 */
+ { { 0, 12, 60 }, -7914, 26028, 0, 0, 11, 66 }, /* region 94 */
+ { { 32768, 61, 96 }, -7914, 26028, 0, 0, 11, 67 }, /* region 95 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 68 }, /* region 96 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 68 }, /* region 97 */
+ { { 1, 12, 35 }, -5355, 16422, 2869, 3778, 8, 69 }, /* region 98 */
+ { { 1, 36, 48 }, -6555, 20675, 2869, 3778, 8, 69 }, /* region 99 */
+ { { 32769, 49, 72 }, -6555, 20675, 2869, 3778, 8, 70 }, /* region 100 */
+ { { 1, 16, 55 }, -6224, 20675, 1045, 1119, 38, 71 }, /* region 101 */
+ { { 32769, 56, 96 }, -6718, 20675, 907, 963, 43, 71 }, /* region 102 */
+ { { 1, 16, 53 }, -5994, 29204, 1140, 1479, 23, 72 }, /* region 103 */
+ { { 1, 54, 70 }, -7171, 29204, 726, 812, 52, 72 }, /* region 104 */
+ { { 32769, 71, 108 }, -7788, 29204, 718, 748, 53, 72 }, /* region 105 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 73 }, /* region 106 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 73 }, /* region 107 */
+ { { 1, 16, 54 }, -5727, 20675, 5362, 5461, 6, 74 }, /* region 108 */
+ { { 1, 55, 63 }, -5851, 26028, 1362, 1454, 24, 74 }, /* region 109 */
+ { { 32769, 64, 108 }, -6744, 16422, 311, 366, 85, 74 }, /* region 110 */
+ { { 1, 16, 48 }, -4798, 20675, 1132, 1301, 27, 75 }, /* region 111 */
+ { { 32769, 49, 108 }, -5988, 20675, 1099, 1184, 32, 75 }, /* region 112 */
+ { { 1, 21, 68 }, -8458, 20675, 87, 2170, 14, 76 }, /* region 113 */
+ { { 1, 69, 82 }, -8960, 20675, 120, 2167, 15, 76 }, /* region 114 */
+ { { 32769, 83, 108 }, -10160, 20675, 376, 2041, 16, 76 }, /* region 115 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 77 }, /* region 116 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 77 }, /* region 117 */
+ { { 32769, 55, 108 }, -7368, 20675, 0, 477, 72, 78 }, /* region 118 */
+ { { 32769, 36, 96 }, -6900, 14636, 101, 151, 113, 79 }, /* region 119 */
+ { { 1, 24, 83 }, -6020, 13045, 0, 83, 123, 80 }, /* region 120 */
+ { { 1, 84, 90 }, -8482, 13045, 0, 20, 142, 80 }, /* region 121 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 143, 80 }, /* region 122 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 81 }, /* region 123 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 81 }, /* region 124 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 82 }, /* region 125 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 82 }, /* region 126 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 83 }, /* region 127 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 83 }, /* region 128 */
+ { { 1, 24, 83 }, -6020, 13045, 0, 83, 123, 84 }, /* region 129 */
+ { { 1, 84, 90 }, -8482, 13045, 0, 20, 142, 84 }, /* region 130 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 143, 84 }, /* region 131 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 85 }, /* region 132 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 85 }, /* region 133 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 86 }, /* region 134 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 86 }, /* region 135 */
+ { { 1, 12, 62 }, -7053, 16422, 23, 10953, 1, 87 }, /* region 136 */
+ { { 32769, 63, 108 }, -7755, 20675, 11, 11753, 0, 87 }, /* region 137 */
+ { { 1, 12, 62 }, -7053, 16422, 23, 10953, 1, 88 }, /* region 138 */
+ { { 32769, 63, 108 }, -7755, 16422, 11, 11753, 0, 88 }, /* region 139 */
+ { { 1, 24, 79 }, -6020, 13045, 0, 83, 123, 89 }, /* region 140 */
+ { { 1, 80, 90 }, -8482, 13045, 0, 20, 142, 89 }, /* region 141 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 143, 89 }, /* region 142 */
+ { { 1, 12, 65 }, -7053, 13045, 23, 10953, 1, 90 }, /* region 143 */
+ { { 32769, 66, 108 }, -7755, 16422, 11, 11753, 0, 90 }, /* region 144 */
+ { { 32768, 36, 84 }, -7500, 20675, 0, 0, 21, 91 }, /* region 145 */
+ { { 32769, 36, 96 }, -8855, 20675, 1482, 1613, 19, 92 }, /* region 146 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 93 }, /* region 147 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 94 }, /* region 148 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 95 }, /* region 149 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 96 }, /* region 150 */
+ { { 32769, 12, 108 }, -7538, 14636, 0, 3911, 7, 97 }, /* region 151 */
+ { { 32769, 12, 96 }, -4366, 32767, 818, 1033, 39, 98 }, /* region 152 */
+ { { 32769, 36, 84 }, -8568, 18426, 0, 293, 95, 99 }, /* region 153 */
+ { { 32769, 12, 96 }, -6020, 26028, 0, 83, 122, 100 }, /* region 154 */
+ { { 32769, 12, 96 }, -6020, 20675, 0, 83, 122, 101 }, /* region 155 */
+ { { 1, 12, 83 }, -6020, 13045, 0, 83, 122, 102 }, /* region 156 */
+ { { 1, 84, 90 }, -8482, 13045, 0, 20, 144, 102 }, /* region 157 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 145, 102 }, /* region 158 */
+ { { 32769, 36, 108 }, -8570, 32767, 472, 491, 71, 103 }, /* region 159 */
+ { { 32769, 36, 108 }, -8570, 20675, 472, 491, 71, 104 }, /* region 160 */
+ { { 1, 12, 72 }, -6012, 7336, 2, 86, 121, 105 }, /* region 161 */
+ { { 1, 73, 101 }, -8500, 8231, 2, 22, 140, 105 }, /* region 162 */
+ { { 32769, 102, 108 }, -9683, 20675, 173, 183, 107, 105 }, /* region 163 */
+ { { 1, 21, 96 }, -7768, 13045, 477, 507, 70, 106 }, /* region 164 */
+ { { 32769, 97, 108 }, -9683, 13045, 173, 183, 107, 107 }, /* region 165 */
+ { { 32769, 12, 108 }, -7771, 16422, 477, 507, 70, 108 }, /* region 166 */
+ { { 1, 12, 53 }, -4971, 16422, 388, 541, 65, 109 }, /* region 167 */
+ { { 32769, 54, 60 }, -5949, 11626, 473, 560, 62, 109 }, /* region 168 */
+ { { 32769, 36, 72 }, -5949, 16422, 473, 560, 62, 110 }, /* region 169 */
+ { { 1, 48, 58 }, -7053, 16422, 356, 402, 80, 111 }, /* region 170 */
+ { { 1, 59, 65 }, -7574, 16422, 514, 548, 64, 111 }, /* region 171 */
+ { { 1, 66, 78 }, -8174, 16422, 505, 529, 68, 111 }, /* region 172 */
+ { { 32769, 79, 96 }, -9233, 16422, 178, 191, 106, 111 }, /* region 173 */
+ { { 1, 55, 60 }, -7053, 16422, 356, 402, 80, 112 }, /* region 174 */
+ { { 1, 61, 69 }, -7574, 16422, 514, 548, 64, 112 }, /* region 175 */
+ { { 1, 70, 79 }, -8174, 16422, 505, 529, 68, 112 }, /* region 176 */
+ { { 32769, 80, 108 }, -9233, 16422, 178, 191, 106, 112 }, /* region 177 */
+ { { 1, 16, 82 }, -8029, 23197, 180, 206, 103, 113 }, /* region 178 */
+ { { 32769, 83, 108 }, -7240, 18426, 3, 44, 128, 113 }, /* region 179 */
+ { { 32769, 21, 108 }, -8869, 20675, 483, 515, 69, 114 }, /* region 180 */
+ { { 1, 21, 89 }, -7205, 18426, 3, 45, 127, 115 }, /* region 181 */
+ { { 32769, 90, 108 }, -9101, 10362, 6, 20, 145, 115 }, /* region 182 */
+ { { 1, 21, 42 }, -4686, 20675, 0, 180, 108, 116 }, /* region 183 */
+ { { 1, 43, 51 }, -5286, 23197, 0, 127, 117, 116 }, /* region 184 */
+ { { 1, 52, 58 }, -6292, 26028, 0, 71, 124, 116 }, /* region 185 */
+ { { 1, 59, 68 }, -7468, 23197, 0, 36, 133, 116 }, /* region 186 */
+ { { 32769, 69, 108 }, -8574, 20675, 0, 19, 146, 116 }, /* region 187 */
+ { { 1, 21, 89 }, -7199, 20675, 3, 45, 127, 117 }, /* region 188 */
+ { { 32769, 90, 108 }, -9101, 14636, 6, 20, 145, 117 }, /* region 189 */
+ { { 1, 21, 46 }, -5651, 26028, 236, 340, 89, 118 }, /* region 190 */
+ { { 1, 47, 71 }, -6563, 20675, 824, 885, 46, 118 }, /* region 191 */
+ { { 1, 72, 88 }, -7907, 18426, 719, 747, 54, 118 }, /* region 192 */
+ { { 1, 89, 93 }, -8876, 16422, 83, 99, 119, 118 }, /* region 193 */
+ { { 32769, 94, 108 }, -9689, 16422, 173, 183, 107, 118 }, /* region 194 */
+ { { 1, 60, 71 }, -7205, 16422, 0, 42, 129, 119 }, /* region 195 */
+ { { 1, 72, 78 }, -7903, 16422, 0, 28, 138, 119 }, /* region 196 */
+ { { 32769, 79, 96 }, -8405, 16422, 0, 21, 141, 119 }, /* region 197 */
+ { { 1, 48, 65 }, -6316, 11626, 0, 70, 125, 120 }, /* region 198 */
+ { { 1, 66, 79 }, -7724, 14636, 0, 31, 135, 120 }, /* region 199 */
+ { { 32769, 80, 96 }, -8030, 11626, 0, 26, 139, 120 }, /* region 200 */
+ { { 1, 16, 44 }, -5868, 14636, 163, 254, 99, 121 }, /* region 201 */
+ { { 1, 45, 51 }, -6418, 16422, 261, 393, 82, 121 }, /* region 202 */
+ { { 1, 52, 58 }, -7333, 18426, 190, 229, 101, 121 }, /* region 203 */
+ { { 1, 59, 66 }, -8100, 18426, 168, 193, 105, 121 }, /* region 204 */
+ { { 1, 67, 70 }, -8576, 18426, 138, 157, 112, 121 }, /* region 205 */
+ { { 1, 71, 80 }, -9103, 18426, 166, 180, 109, 121 }, /* region 206 */
+ { { 32769, 81, 108 }, -10074, 18426, 135, 151, 114, 121 }, /* region 207 */
+ { { 32769, 12, 96 }, -5004, 23197, 570, 719, 56, 122 }, /* region 208 */
+ { { 1, 12, 48 }, -5868, 14636, 163, 254, 99, 123 }, /* region 209 */
+ { { 1, 49, 54 }, -6418, 16422, 261, 393, 82, 123 }, /* region 210 */
+ { { 1, 55, 63 }, -7333, 18426, 190, 229, 101, 123 }, /* region 211 */
+ { { 1, 64, 70 }, -8100, 18426, 168, 193, 105, 123 }, /* region 212 */
+ { { 1, 71, 75 }, -8576, 18426, 138, 157, 112, 123 }, /* region 213 */
+ { { 1, 76, 82 }, -9103, 18426, 166, 180, 109, 123 }, /* region 214 */
+ { { 32769, 83, 108 }, -10074, 18426, 135, 151, 114, 123 }, /* region 215 */
+ { { 32770, 36, 84 }, -7200, 29204, 0, 0, 0, 124 }, /* region 216 */
+ { { 32770, 36, 84 }, -7600, 8231, 0, 0, 0, 125 }, /* region 217 */
+ { { 32770, 36, 84 }, -7200, 20675, 0, 0, 0, 126 }, /* region 218 */
+ { { 32769, 36, 84 }, -6000, -24285, 1294, 5778, 5, 127 }, /* region 219 */
+ { { 32769, 36, 84 }, -6555, 29204, 798, 993, 42, 128 }, /* region 220 */
+ { { 32769, 36, 84 }, -6855, 20675, 798, 993, 42, 129 }, /* region 221 */
+ { { 32769, 36, 84 }, -7755, 29204, 798, 993, 42, 130 }, /* region 222 */
+ { { 32768, 36, 84 }, -8155, 32767, 0, 0, 130, 131 }, /* region 223 */
+ { { 32768, 36, 84 }, -6555, 20675, 0, 0, 87, 132 }, /* region 224 */
+ { { 1, 24, 62 }, -7000, 23197, 286, 333, 91, 133 }, /* region 225 */
+ { { 1, 63, 66 }, -7364, 26028, 297, 335, 90, 133 }, /* region 226 */
+ { { 1, 67, 72 }, -7722, 23197, 368, 399, 81, 133 }, /* region 227 */
+ { { 32769, 73, 96 }, -8310, 23197, 116, 138, 116, 133 }, /* region 228 */
+ { { 1, 24, 48 }, -5141, 23197, 309, 447, 74, 134 }, /* region 229 */
+ { { 1, 49, 56 }, -6266, 26028, 211, 283, 96, 134 }, /* region 230 */
+ { { 1, 57, 63 }, -7000, 26028, 286, 333, 91, 134 }, /* region 231 */
+ { { 32769, 64, 84 }, -7722, 23197, 368, 399, 81, 134 }, /* region 232 */
+ { { 1, 24, 56 }, -6266, 29204, 211, 283, 96, 135 }, /* region 233 */
+ { { 1, 57, 63 }, -7000, 29204, 286, 333, 91, 135 }, /* region 234 */
+ { { 1, 64, 69 }, -7722, 29204, 368, 399, 81, 135 }, /* region 235 */
+ { { 32769, 70, 96 }, -8310, 29204, 116, 138, 116, 135 }, /* region 236 */
+ { { 1, 24, 68 }, -7722, 18426, 368, 399, 81, 136 }, /* region 237 */
+ { { 1, 69, 76 }, -8310, 26028, 116, 138, 116, 136 }, /* region 238 */
+ { { 32769, 77, 108 }, -8758, 23197, 127, 144, 115, 136 }, /* region 239 */
+ { { 1, 24, 82 }, -7613, 23197, 389, 422, 77, 137 }, /* region 240 */
+ { { 32769, 83, 108 }, -8764, 26028, 146, 163, 111, 137 }, /* region 241 */
+ { { 1, 12, 58 }, -6898, 29204, 386, 436, 75, 138 }, /* region 242 */
+ { { 32769, 59, 96 }, -7371, 26028, 290, 328, 92, 138 }, /* region 243 */
+ { { 1, 12, 58 }, -6898, 16422, 386, 436, 75, 139 }, /* region 244 */
+ { { 32769, 59, 96 }, -7371, 18426, 290, 328, 92, 139 }, /* region 245 */
+ { { 1, 12, 48 }, -6898, -28771, 386, 436, 75, 140 }, /* region 246 */
+ { { 32769, 49, 84 }, -7371, 29204, 290, 328, 92, 140 }, /* region 247 */
+ { { 1, 12, 60 }, -5453, 20675, 314, 430, 76, 141 }, /* region 248 */
+ { { 32769, 61, 84 }, -6553, 18426, 263, 324, 93, 141 }, /* region 249 */
+ { { 1, 24, 60 }, -6553, 16422, 263, 324, 93, 142 }, /* region 250 */
+ { { 1, 61, 70 }, -7669, 20675, 279, 311, 94, 142 }, /* region 251 */
+ { { 32769, 71, 96 }, -8098, 23197, 179, 204, 104, 142 }, /* region 252 */
+ { { 1, 24, 84 }, -8483, 20675, 191, 211, 102, 143 }, /* region 253 */
+ { { 32769, 85, 108 }, -9683, 20675, 92, 102, 118, 143 }, /* region 254 */
+ { { 1, 21, 69 }, -6553, 13045, 263, 324, 93, 144 }, /* region 255 */
+ { { 1, 70, 94 }, -7669, 20675, 279, 311, 94, 144 }, /* region 256 */
+ { { 1, 95, 96 }, -8098, -24285, 179, 204, 104, 144 }, /* region 257 */
+ { { 32769, 97, 108 }, -9683, -24285, 173, 183, 107, 144 }, /* region 258 */
+ { { 1, 16, 55 }, -8100, 20675, 168, 193, 105, 145 }, /* region 259 */
+ { { 1, 56, 74 }, -8576, 26028, 138, 157, 112, 145 }, /* region 260 */
+ { { 32769, 75, 96 }, -10074, 26028, 135, 151, 114, 145 }, /* region 261 */
+ { { 1, 24, 72 }, -8098, 26028, 179, 204, 104, 146 }, /* region 262 */
+ { { 1, 73, 85 }, -8483, 20675, 191, 211, 102, 146 }, /* region 263 */
+ { { 32769, 86, 108 }, -9683, 18426, 92, 102, 118, 146 }, /* region 264 */
+ { { 32769, 36, 108 }, -7730, 18426, 1839, 1901, 17, 147 }, /* region 265 */
+ { { 32769, 24, 108 }, -7273, 20675, 494, 534, 66, 148 }, /* region 266 */
+ { { 32769, 12, 108 }, -7273, 20675, 494, 534, 66, 149 }, /* region 267 */
+ { { 32769, 24, 108 }, -7273, 20675, 494, 534, 66, 150 }, /* region 268 */
+ { { 1, 36, 60 }, -4900, 5193, 2, 22, 140, 151 }, /* region 269 */
+ { { 32769, 61, 84 }, -6083, 20675, 173, 183, 107, 151 }, /* region 270 */
+ { { 32769, 24, 96 }, -6553, 14636, 263, 324, 93, 152 }, /* region 271 */
+ { { 32769, 36, 96 }, -7730, 26028, 1839, 1901, 17, 153 }, /* region 272 */
+ { { 32769, 24, 108 }, -7273, 20675, 494, 534, 66, 154 }, /* region 273 */
+ { { 1, 24, 58 }, -7851, 14636, 0, 29, 137, 155 }, /* region 274 */
+ { { 32769, 59, 96 }, -7851, 14636, 0, 29, 137, 155 }, /* region 275 */
+ { { 1, 12, 83 }, -6020, 13045, 0, 83, 122, 156 }, /* region 276 */
+ { { 1, 84, 90 }, -8482, 13045, 0, 20, 144, 156 }, /* region 277 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 145, 156 }, /* region 278 */
+ { { 1, 21, 42 }, -4663, 26028, 1047, 1229, 30, 157 }, /* region 279 */
+ { { 1, 43, 48 }, -5456, 29204, 1138, 1253, 29, 157 }, /* region 280 */
+ { { 1, 49, 53 }, -5845, 26028, 559, 651, 57, 157 }, /* region 281 */
+ { { 1, 54, 60 }, -6732, 26028, 508, 563, 61, 157 }, /* region 282 */
+ { { 1, 61, 65 }, -7080, 32767, 819, 864, 48, 157 }, /* region 283 */
+ { { 1, 66, 70 }, -6866, 26942, 981, 1032, 40, 157 }, /* region 284 */
+ { { 1, 71, 76 }, -8166, 26028, 790, 814, 51, 157 }, /* region 285 */
+ { { 1, 77, 82 }, -8766, 26028, 592, 609, 58, 157 }, /* region 286 */
+ { { 1, 83, 87 }, -9517, 23197, 543, 554, 63, 157 }, /* region 287 */
+ { { 1, 88, 96 }, -10071, 18426, 601, 609, 59, 157 }, /* region 288 */
+ { { 32769, 97, 108 }, -10566, 18426, 523, 529, 67, 157 }, /* region 289 */
+ { { 1, 48, 69 }, -6313, 14636, 0, 70, 125, 158 }, /* region 290 */
+ { { 1, 70, 79 }, -7724, 18426, 0, 31, 135, 158 }, /* region 291 */
+ { { 32769, 80, 96 }, -8030, 14636, 0, 26, 139, 158 }, /* region 292 */
+ { { 1, 36, 72 }, -7134, 29204, 0, 87, 120, 159 }, /* region 293 */
+ { { 32769, 73, 96 }, -7960, 29204, 0, 54, 126, 159 }, /* region 294 */
+ { { 32769, 36, 96 }, -7730, 26028, 1839, 1901, 17, 160 }, /* region 295 */
+ { { 32769, 12, 96 }, -4372, 32767, 818, 1033, 39, 161 }, /* region 296 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 162 }, /* region 297 */
+ { { 32769, 12, 108 }, -7538, 20675, 0, 3911, 7, 163 }, /* region 298 */
+ { { 32769, 36, 108 }, -8570, 26028, 472, 491, 71, 164 }, /* region 299 */
+ { { 32769, 12, 96 }, -5004, 29204, 570, 719, 56, 165 }, /* region 300 */
+ { { 1, 12, 83 }, -6020, 13045, 0, 83, 122, 166 }, /* region 301 */
+ { { 1, 84, 90 }, -8482, 13045, 0, 20, 144, 166 }, /* region 302 */
+ { { 32769, 91, 108 }, -9101, 13045, 6, 20, 145, 166 }, /* region 303 */
+ { { 1, 21, 46 }, -5651, 32767, 236, 340, 89, 167 }, /* region 304 */
+ { { 1, 47, 75 }, -6563, 26028, 824, 885, 46, 167 }, /* region 305 */
+ { { 1, 76, 84 }, -7907, 23197, 719, 747, 54, 167 }, /* region 306 */
+ { { 1, 85, 93 }, -8876, 20675, 83, 99, 119, 167 }, /* region 307 */
+ { { 32769, 94, 108 }, -9689, 20675, 173, 183, 107, 167 }, /* region 308 */
+ { { 1, 21, 46 }, -5651, 26028, 236, 340, 89, 168 }, /* region 309 */
+ { { 1, 47, 71 }, -6563, 20675, 824, 885, 46, 168 }, /* region 310 */
+ { { 1, 72, 88 }, -7907, 18426, 719, 747, 54, 168 }, /* region 311 */
+ { { 1, 89, 93 }, -8876, 16422, 83, 99, 119, 168 }, /* region 312 */
+ { { 32769, 94, 108 }, -9689, 16422, 173, 183, 107, 168 }, /* region 313 */
+ { { 1, 21, 45 }, -4663, 26028, 1047, 1229, 30, 169 }, /* region 314 */
+ { { 1, 46, 51 }, -5456, 29204, 1138, 1253, 29, 169 }, /* region 315 */
+ { { 1, 52, 54 }, -5845, 26028, 559, 651, 57, 169 }, /* region 316 */
+ { { 1, 55, 63 }, -6732, 26028, 508, 563, 61, 169 }, /* region 317 */
+ { { 1, 64, 68 }, -7080, 32767, 819, 864, 48, 169 }, /* region 318 */
+ { { 1, 69, 73 }, -6866, 26942, 981, 1032, 40, 169 }, /* region 319 */
+ { { 1, 74, 79 }, -8166, 26028, 790, 814, 51, 169 }, /* region 320 */
+ { { 1, 80, 88 }, -8766, 23197, 592, 609, 58, 169 }, /* region 321 */
+ { { 1, 89, 99 }, -10071, 18426, 601, 609, 59, 169 }, /* region 322 */
+ { { 32769, 100, 108 }, -10566, 18426, 523, 529, 67, 169 }, /* region 323 */
+ { { 1, 21, 45 }, -4663, 26028, 1047, 1229, 30, 170 }, /* region 324 */
+ { { 1, 46, 51 }, -5456, 29204, 1138, 1253, 29, 170 }, /* region 325 */
+ { { 1, 52, 54 }, -5845, 26028, 559, 651, 57, 170 }, /* region 326 */
+ { { 1, 55, 63 }, -6732, 26028, 508, 563, 61, 170 }, /* region 327 */
+ { { 1, 64, 68 }, -7080, 32767, 819, 864, 48, 170 }, /* region 328 */
+ { { 1, 69, 73 }, -6866, 26942, 981, 1032, 40, 170 }, /* region 329 */
+ { { 1, 74, 79 }, -8166, 26028, 790, 814, 51, 170 }, /* region 330 */
+ { { 1, 80, 88 }, -8766, 23197, 592, 609, 58, 170 }, /* region 331 */
+ { { 1, 89, 99 }, -10071, 18426, 601, 609, 59, 171 }, /* region 332 */
+ { { 32769, 100, 108 }, -10566, 18426, 523, 529, 67, 172 }, /* region 333 */
+ { { 32769, 36, 108 }, -8570, 20675, 472, 491, 71, 173 }, /* region 334 */
+ { { 32769, 12, 108 }, -7730, 20675, 1839, 1901, 17, 174 }, /* region 335 */
+ { { 1, 12, 44 }, -5868, 18426, 163, 254, 99, 175 }, /* region 336 */
+ { { 1, 45, 51 }, -6418, 20675, 261, 393, 82, 175 }, /* region 337 */
+ { { 1, 52, 58 }, -7333, 23197, 190, 229, 101, 175 }, /* region 338 */
+ { { 1, 59, 66 }, -8100, 23197, 168, 193, 105, 175 }, /* region 339 */
+ { { 1, 67, 70 }, -8576, 23197, 138, 157, 112, 175 }, /* region 340 */
+ { { 1, 71, 80 }, -9103, 23197, 166, 180, 109, 175 }, /* region 341 */
+ { { 32769, 81, 108 }, -10074, 23197, 135, 151, 114, 175 }, /* region 342 */
+ { { 1, 12, 65 }, -7053, 16422, 23, 10953, 1, 176 }, /* region 343 */
+ { { 32769, 66, 108 }, -7755, 20675, 11, 11753, 0, 176 }, /* region 344 */
+ { { 1, 12, 48 }, -4798, 29204, 1132, 1301, 27, 177 }, /* region 345 */
+ { { 32769, 49, 108 }, -5988, 29204, 1099, 1184, 32, 177 }, /* region 346 */
+ { { 1, 12, 83 }, -7241, 20675, 419, 460, 73, 178 }, /* region 347 */
+ { { 32769, 84, 108 }, -10123, 20675, 0, 31, 136, 178 }, /* region 348 */
+ { { 1, 55, 60 }, -7053, 18426, 356, 402, 80, 179 }, /* region 349 */
+ { { 1, 61, 69 }, -7574, 18426, 514, 548, 64, 179 }, /* region 350 */
+ { { 1, 70, 79 }, -8174, 18426, 505, 529, 68, 179 }, /* region 351 */
+ { { 32769, 80, 108 }, -9233, 23197, 178, 191, 106, 179 }, /* region 352 */
+ { { 32769, 36, 96 }, -7730, -24285, 1839, 1901, 17, 180 }, /* region 353 */
+ { { 1, 12, 83 }, -6006, 16422, 838, 922, 44, 181 }, /* region 354 */
+ { { 1, 84, 93 }, -8406, 18426, 209, 230, 100, 181 }, /* region 355 */
+ { { 32769, 94, 108 }, -10108, 16422, 0, 31, 136, 181 }, /* region 356 */
+ { { 1, 12, 56 }, -5595, 23197, 1064, 1170, 34, 182 }, /* region 357 */
+ { { 1, 57, 72 }, -5995, 23197, 930, 1014, 41, 182 }, /* region 358 */
+ { { 32769, 73, 108 }, -7598, 23197, 726, 826, 49, 182 }, /* region 359 */
+ { { 32769, 24, 108 }, -7600, 23197, 635, 735, 55, 56 }, /* region 360 */
+ { { 1, 36, 83 }, -6006, 13045, 838, 922, 44, 183 }, /* region 361 */
+ { { 1, 84, 93 }, -8406, 13045, 209, 230, 100, 183 }, /* region 362 */
+ { { 32769, 94, 108 }, -10108, 13045, 0, 31, 136, 183 } /* region 363 */
+}; /* end Regions */
+
+/*----------------------------------------------------------------------------
+ * Programs
+ *----------------------------------------------------------------------------
+*/
+const S_PROGRAM eas_programs[] =
+{
+ { 7864320, 0 } /* program 0 */
+}; /* end Programs */
+
+/*----------------------------------------------------------------------------
+ * Banks
+ *----------------------------------------------------------------------------
+*/
+const S_BANK eas_banks[] =
+{
+ { /* bank 0 */
+ 30976,
+ {
+ 279, 314, 304, 324, 190, 309, 78, 183,
+ 90, 75, 361, 72, 70, 68, 118, 65,
+ 188, 180, 113, 255, 181, 290, 195, 198,
+ 111, 108, 178, 103, 101, 201, 209, 259,
+ 63, 61, 296, 152, 208, 300, 154, 155,
+ 174, 170, 169, 167, 143, 166, 164, 98,
+ 138, 136, 134, 132, 151, 150, 298, 94,
+ 240, 242, 246, 293, 244, 140, 129, 120,
+ 237, 225, 233, 229, 262, 250, 248, 253,
+ 160, 159, 299, 265, 272, 295, 119, 334,
+ 161, 156, 335, 343, 336, 149, 93, 301,
+ 347, 127, 87, 148, 354, 357, 297, 276,
+ 125, 123, 81, 345, 116, 106, 147, 96,
+ 273, 268, 267, 266, 360, 274, 349, 271,
+ 84, 224, 146, 223, 222, 221, 220, 219,
+ 145, 353, 218, 269, 153, 217, 92, 216
+ }
+ }
+}; /* end Banks */
+
+/*----------------------------------------------------------------------------
+ * Samples
+ *----------------------------------------------------------------------------
+*/
+
+const EAS_SAMPLE eas_samples[] =
+{
+ 0, -1, -7, -18, -16, 3, 6, -18, -29, -21, 6, 6, -12, -10, -10, -2,
+ 4, 15, 21, 14, 4, 9, 29, 27, 15, 9, 13, 20, 0, -21, -11, 3,
+ -11, -37, -36, -7, 11, -4, -26, -23, -4, 15, -4, -15, -11, -10, -1, 9,
+ 23, 15, 8, 6, 24, 35, 19, 14, 6, 17, 10, -15, -26, -7, -1, -23,
+ -40, -24, 4, 10, -16, -25, -13, 11, 10, -14, -15, -12, -6, 1, 19, 22,
+ 6, 7, 15, 40, 31, 22, 7, 7, 15, 5, -22, -21, -5, -9, -33, -32,
+ -15, 5, -4, -19, -20, 1, 17, -4, -15, -11, -12, -4, 9, 27, 13, 7,
+ 11, 32, 44, 30, 14, 2, 7, 9, -8, -25, -18, -9, -19, -30, -21, -4,
+ 2, -10, -21, -13, 15, 8, -11, -13, -16, -12, 0, 20, 24, 9, 9, 19,
+ 44, 47, 25, 6, 2, 11, -1, -21, -27, -19, -17, -20, -23, -17, -5, -5,
+ -14, -20, 0, 15, -4, -7, -17, -17, -8, 7, 25, 18, 7, 11, 31, 54,
+ 41, 14, -1, 10, 3, -14, -27, -29, -21, -16, -16, -18, -17, -11, -10, -16,
+ -16, 10, 6, -3, -9, -19, -16, -2, 18, 27, 16, 8, 19, 45, 54, 29,
+ 5, 6, 4, -9, -18, -29, -30, -18, -10, -9, -14, -20, -12, -13, -17, -5,
+ 12, 7, 2, -16, -22, -13, 8, 25, 23, 12, 11, 32, 53, 48, 17, 9,
+ 9, -7, -16, -24, -37, -28, -12, -4, -2, -18, -21, -18, -23, -17, 7, 11,
+ 10, -4, -16, -22, -7, 21, 28, 18, 9, 20, 43, 55, 37, 11, 11, 1,
+ -14, -17, -31, -34, -21, -10, 4, -2, -20, -23, -26, -29, -3, 6, 13, 8,
+ -5, -23, -27, 7, 28, 27, 15, 12, 32, 48, 50, 24, 9, 8, -12, -19,
+ -26, -37, -35, -19, 3, 6, -12, -21, -31, -38, -17, 6, 4, 14, 6, -8,
+ -35, -15, 19, 26, 21, 7, 23, 43, 52, 42, 14, 9, -2, -16, -21, -25,
+ -33, -33, -4, 13, 3, -13, -30, -47, -37, 1, 6, 7, 12, 6, -15, -31,
+ -1, 22, 30, 14, 9, 37, 48, 50, 26, 8, 6, -11, -22, -22, -25, -45,
+ -22, 13, 15, 1, -22, -43, -56, -21, 7, 6, 10, 15, -7, -25, -21, 7,
+ 28, 21, 2, 25, 47, 53, 39, 11, 10, 1, -20, -22, -18, -37, -42, 2,
+ 15, 15, -8, -32, -57, -48, -9, 9, 5, 14, 8, -14, -20, -13, 17, 33,
+ 8, 9, 37, 52, 55, 25, 7, 8, -11, -23, -15, -23, -47, -20, 12, 21,
+ 10, -23, -50, -57, -31, 2, 10, 9, 11, -8, -10, -18, -3, 30, 22, 1,
+ 22, 45, 57, 45, 11, 11, 0, -23, -22, -16, -36, -36, -6, 15, 18, -3,
+ -43, -56, -49, -21, 5, 11, 12, -2, -8, -9, -15, 12, 31, 9, 10, 34,
+ 48, 60, 29, 16, 10, -17, -24, -21, -27, -31, -17, 5, 14, 12, -22, -50,
+ -48, -41, -9, 7, 13, -1, -8, -2, -11, -5, 26, 19, 6, 21, 37, 55,
+ 49, 25, 22, -5, -20, -24, -26, -29, -23, -9, 8, 13, -1, -41, -48, -47,
+ -30, -3, 12, 7, -12, -2, 1, -8, 10, 20, 11, 16, 26, 40, 54, 42,
+ 36, 7, -13, -18, -26, -29, -26, -14, -2, 7, 8, -18, -45, -48, -45, -19,
+ 6, 11, -10, -8, 5, -7, 2, 12, 17, 15, 16, 23, 45, 55, 48, 27,
+ -7, -12, -18, -30, -31, -17, -5, -3, 4, 5, -24, -40, -50, -38, -8, 8,
+ 1, -13, 1, -1, 3, 8, 15, 18, 15, 15, 28, 52, 59, 47, 9, -9,
+ -10, -25, -40, -29, -3, -5, -13, 1, 3, -26, -47, -53, -23, -2, 5, -11,
+ -6, 4, 6, 10, 10, 19, 11, 13, 15, 42, 61, 61, 32, 3, -5, -12,
+ -35, -43, -14, 1, -16, -13, 9, -1, -33, -57, -46, -19, -3, -1, -16, -2,
+ 5, 15, 6, 17, 9, 5, 9, 26, 54, 65, 52, 21, 3, -6, -21, -48,
+ -33, -4, -7, -22, -2, 18, -4, -47, -58, -33, -11, 2, -13, -15, 3, 16,
+ 15, 9, 15, -1, 5, 14, 48, 66, 65, 37, 23, 2, -11, -42, -48, -20,
+ -4, -23, -18, 18, 19, -17, -61, -52, -27, -2, -5, -24, -7, 9, 20, 14,
+ 18, 2, -3, 7, 34, 63, 68, 50, 37, 19, -9, -25, -49, -38, -14, -14,
+ -28, 6, 25, 9, -40, -62, -45, -17, 3, -21, -16, -3, 12, 18, 18, 10,
+ -6, 0, 16, 56, 69, 63, 43, 38, 2, -20, -42, -44, -33, -14, -25, -10,
+ 22, 18, -11, -55, -57, -37, -3, -10, -26, -10, 2, 17, 21, 20, -3, -2,
+ 5, 39, 67, 73, 56, 47, 22, -17, -35, -43, -36, -28, -18, -20, 9, 22,
+ 8, -33, -56, -53, -26, -2, -23, -24, -11, 4, 23, 30, 4, -3, 0, 27,
+ 60, 71, 69, 58, 43, -6, -35, -43, -35, -36, -27, -18, -2, 19, 17, -9,
+ -41, -51, -45, -10, -12, -31, -21, -9, 8, 33, 17, 0, 2, 8, 50, 69,
+ 75, 70, 64, 16, -31, -47, -35, -32, -34, -25, -14, 10, 16, 5, -26, -42,
+ -50, -32, -8, -22, -31, -22, -8, 22, 32, 7, 6, 4, 28, 63, 69, 71,
+ 78, 43, -20, -51, -41, -23, -34, -30, -22, 2, 13, 9, -10, -31, -45, -44,
+ -21, -19, -27, -36, -26, 2, 29, 24, 12, 11, 13, 45, 69, 68, 81, 70,
+ 6, -47, -52, -28, -23, -27, -26, -14, 11, 10, -2, -18, -33, -43, -34, -25,
+ -22, -36, -44, -14, 14, 29, 27, 22, 18, 29, 60, 65, 71, 86, 41, -32,
+ -57, -41, -23, -22, -22, -24, -4, 10, 0, -11, -21, -31, -37, -37, -29, -29,
+ -53, -33, -3, 16, 33, 31, 27, 23, 46, 64, 65, 85, 71, -4, -48, -47,
+ -31, -20, -15, -15, -19, 1, 0, -5, -14, -21, -26, -40, -39, -28, -51, -50,
+ -18, 4, 26, 38, 36, 30, 32, 61, 64, 70, 80, 31, -35, -43, -40, -25,
+ -19, -6, -17, -11, 0, -6, -11, -17, -15, -31, -47, -39, -45, -62, -34, -8,
+ 12, 36, 38, 41, 37, 51, 65, 60, 71, 56, -7, -36, -38, -35, -27, -11,
+ -5, -17, -8, -8, -9, -17, -13, -20, -37, -47, -50, -62, -49, -16, 1, 25,
+ 34, 43, 53, 49, 62, 58, 62, 65, 22, -23, -31, -34, -35, -19, -3, -8,
+ -18, -7, -4, -14, -17, -9, -27, -40, -56, -68, -64, -27, -4, 9, 28, 37,
+ 54, 60, 61, 59, 54, 63, 43, -1, -25, -23, -37, -28, -8, 4, -10, -15,
+ -3, -7, -26, -15, -11, -29, -51, -71, -71, -49, -8, 5, 14, 30, 48, 66,
+ 68, 63, 51, 56, 51, 18, -18, -20, -25, -39, -23, -5, 3, -13, -6, -5,
+ -21, -28, -10, -18, -43, -67, -78, -64, -25, 1, 8, 15, 38, 60, 75, 74,
+ 58, 54, 57, 28, -7, -18, -8, -31, -34, -15, 4, -2, -4, 0, -15, -32,
+ -21, -8, -28, -57, -77, -75, -44, -12, 6, 8, 20, 49, 68, 77, 71, 53,
+ 61, 41, 3, -18, -4, -14, -36, -28, -7, 1, 7, 6, -8, -34, -35, -16,
+ -15, -41, -66, -75, -59, -31, -4, 10, 11, 31, 56, 72, 85, 63, 58, 55,
+ 14, -17, -10, -1, -24, -30, -22, -6, 11, 16, 9, -27, -43, -27, -15, -26,
+ -53, -63, -66, -46, -21, 3, 13, 15, 40, 63, 83, 78, 60, 67, 38, -9,
+ -20, 4, -7, -23, -25, -22, 6, 24, 20, -8, -48, -43, -24, -16, -39, -55,
+ -57, -57, -36, -10, 9, 11, 22, 50, 73, 88, 71, 65, 59, 14, -24, -9,
+ -3, -15, -18, -31, -14, 25, 28, 10, -36, -56, -41, -21, -29, -48, -47, -51,
+ -50, -26, -2, 11, 6, 30, 57, 82, 83, 69, 65, 35, -12, -16, -3, -10,
+ -12, -21, -26, 11, 33, 26, -7, -56, -54, -32, -26, -41, -39, -37, -52, -41,
+ -16, 5, 7, 11, 37, 72, 91, 78, 66, 55, 11, -19, -14, -15, -9, -7,
+ -19, -8, 22, 30, 13, -32, -62, -44, -31, -34, -40, -29, -40, -46, -29, -8,
+ 2, 1, 18, 49, 84, 86, 72, 67, 39, -6, -21, -25, -16, 2, -6, -12,
+ 6, 23, 18, -7, -51, -56, -39, -33, -41, -30, -24, -41, -36, -17, -8, -5,
+ 7, 32, 66, 84, 78, 75, 58, 20, -15, -29, -33, 2, 4, -1, 1, 11,
+ 17, 4, -27, -56, -46, -36, -37, -39, -22, -24, -38, -23, -16, -16, -7, 18,
+ 46, 72, 82, 81, 72, 44, -1, -25, -41, -18, 14, 8, 10, 0, 5, 2,
+ -11, -40, -50, -42, -38, -37, -33, -11, -27, -26, -20, -22, -13, 0, 31, 52,
+ 73, 82, 80, 61, 26, -16, -37, -40, 0, 18, 17, 12, -5, -8, -10, -21,
+ -41, -43, -44, -36, -35, -20, -12, -23, -14, -32, -21, -12, 15, 38, 58, 76,
+ 82, 72, 46, 4, -26, -41, -23, 14, 25, 18, 2, -23, -18, -14, -29, -35,
+ -45, -42, -35, -29, -15, -19, -3, -23, -32, -19, -1, 27, 40, 67, 81, 76,
+ 61, 26, -11, -33, -36, -6, 28, 28, 13, -15, -38, -19, -20, -26, -36, -46,
+ -35, -32, -20, -21, -4, -4, -28, -25, -16, 13, 31, 49, 74, 79, 69, 48,
+ 8, -16, -33, -21, 7, 33, 24, 5, -32, -38, -22, -21, -22, -41, -43, -32,
+ -25, -23, -16, 6, -9, -29, -28, -5, 20, 39, 58, 75, 71, 60, 22, 0,
+ -18, -27, -10, 16, 30, 15, -14, -49, -34, -19, -16, -27, -43, -31, -25, -27,
+ -24, -1, 7, -15, -32, -22, 3, 28, 50, 61, 67, 59, 38, 13, 5, -17,
+ -16, -3, 21, 23, 3, -34, -49, -28, -17, -19, -34, -34, -18, -26, -32, -16,
+ 14, 6, -23, -34, -11, 14, 44, 54, 56, 60, 44, 25, 18, 2, -18, -9,
+ 3, 21, 9, -14, -45, -39, -24, -22, -28, -32, -18, -17, -36, -30, 5, 21,
+ -5, -31, -27, 1, 30, 56, 46, 50, 47, 32, 29, 20, -4, -9, -9, 4,
+ 10, -2, -28, -38, -31, -33, -26, -26, -19, -9, -24, -38, -11, 24, 10, -17,
+ -36, -14, 14, 48, 49, 37, 43, 39, 33, 32, 11, 1, -5, -11, -6, -3,
+ -16, -29, -28, -40, -38, -27, -20, -6, -11, -31, -29, 15, 22, 0, -25, -25,
+ 2, 31, 51, 38, 35, 41, 41, 34, 27, 8, 6, -7, -20, -14, -7, -20,
+ -19, -34, -52, -35, -23, -11, -5, -18, -33, -5, 26, 5, -11, -27, -11, 17,
+ 43, 43, 30, 29, 46, 45, 38, 17, 6, 2, -22, -27, -16, -16, -18, -18,
+ -53, -48, -23, -19, -3, -10, -21, -19, 18, 12, -6, -20, -21, 6, 29, 42,
+ 31, 23, 37, 54, 48, 31, 13, 13, -13, -31, -25, -16, -16, -11, -38, -62,
+ -33, -28, -7, -3, -12, -17, 6, 16, 2, -11, -17, -5, 15, 35, 34, 25,
+ 23, 44, 55, 41, 22, 14, 2, -29, -32, -23, -11, -13, -18, -60, -50, -32,
+ -16, -6, -11, -4, 3, 11, 3, -1, -14, -8, 5, 20, 39, 31, 24, 26,
+ 54, 57, 37, 16, 8, -19, -32, -27, -17, -17, -17, -36, -58, -40, -32, -11,
+ -11, -2, 6, 10, 2, 0, -4, -12, 1, 8, 33, 42, 26, 17, 35, 63,
+ 53, 26, 6, -9, -27, -30, -25, -17, -23, -24, -47, -51, -38, -17, -11, -6,
+ 12, 17, 8, -8, -1, -10, -7, 4, 18, 40, 33, 18, 22, 49, 70, 46,
+ 12, -11, -16, -23, -27, -19, -28, -26, -36, -50, -48, -31, -13, -13, 10, 20,
+ 21, 0, -6, -1, -7, 3, 15, 32, 40, 24, 17, 31, 58, 67, 28, -2,
+ -21, -14, -31, -18, -24, -33, -33, -49, -49, -41, -18, -20, -2, 20, 23, 19,
+ -12, -7, -2, -3, 10, 22, 33, 37, 21, 23, 39, 72, 51, 10, -17, -18,
+ -20, -29, -17, -33, -36, -49, -48, -47, -24, -16, -15, 11, 24, 30, 6, -12,
+ -2, -2, -1, 18, 29, 35, 27, 25, 25, 56, 63, 23, -5, -19, -14, -23,
+ -21, -21, -38, -52, -54, -40, -32, -15, -23, -7, 18, 31, 22, -2, -7, 3,
+ 0, 12, 21, 34, 31, 28, 22, 35, 66, 40, 2, -19, -19, -15, -21, -16,
+ -30, -60, -62, -42, -33, -18, -18, -22, 4, 29, 29, 14, -2, -4, 5, 6,
+ 16, 28, 36, 24, 29, 26, 52, 49, 14, -13, -24, -19, -14, -15, -17, -53,
+ -75, -54, -32, -18, -11, -26, -18, 18, 28, 25, 13, -6, 4, 13, 9, 17,
+ 37, 24, 28, 33, 37, 49, 26, 1, -19, -26, -16, -11, -7, -33, -77, -71,
+ -45, -15, -8, -18, -30, 1, 25, 21, 26, 3, 5, 17, 14, 11, 33, 27,
+ 17, 34, 38, 42, 33, 6, -11, -28, -22, -12, -3, -16, -60, -81, -62, -25,
+ -2, -14, -27, -17, 20, 18, 24, 18, 6, 16, 21, 12, 24, 28, 12, 26,
+ 44, 42, 35, 13, -5, -21, -31, -18, -2, -2, -42, -77, -76, -47, -10, -6,
+ -22, -24, 4, 16, 18, 25, 18, 18, 24, 20, 20, 32, 14, 11, 40, 49,
+ 41, 21, -6, -13, -27, -28, -5, 5, -18, -59, -79, -63, -30, -6, -11, -28,
+ -11, 8, 10, 23, 27, 17, 24, 25, 25, 28, 19, 8, 25, 47, 46, 27,
+ -5, -15, -18, -33, -20, -5, -3, -35, -70, -66, -51, -22, -7, -21, -19, -3,
+ 6, 17, 29, 26, 23, 28, 25, 32, 23, 14, 11, 39, 50, 38, 3, -19,
+ -10, -21, -29, -12, -3, -12, -55, -65, -56, -42, -14, -10, -23, -11, -3, 10,
+ 23, 32, 26, 28, 28, 35, 29, 22, 13, 18, 41, 43, 18, -16, -18, -14,
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+ -74, -60, -46, -29, -12, -2, 6, -23, 28, -20, 50, 50, -14, 14, -28, -2,
+ 28, 52, -25, -13, -52, 43, -41, -9, -26, 36, -25, -20, -17, 3, -11, 38,
+ 30, -61, 12, -6, 5, 45, 41, -4, 4, -20, -12, 45, 33, -11, -29, -44,
+ 27, -17, -36, -1, 17, -11, -31, -12, 4, -10, 37, 20, -39, -23, 0, -104,
+ -55, -11, 24, 33, 30, 7, -15, -31, -30, -25, -23, -20, -9, 10, 31, 59,
+ 91, 111, 115, 92, 51, 7, -33, -64, -81, -81, -71, -51, -22, 16, 52, 74,
+ 82, 81, 68, 38, 0, -40, -81, -112, -124, -102, -57, -11, 0, 7, -23, -58,
+ -85, -100, -104, -100, -92, -81, -69, -57, -45, -33, -22, -11, -1, 8, 18, 27,
+ 37, 49, 62, 79, 99, 118, 124, 111, 84, 53, 26, 5, -8, -14, -15, -12,
+ -6, 2, 12, 21, 28, 26, 9, -22, -58, -20, 7, 10, 21, -14, 9, 22,
+ 57, 62, 127, 56, 28, 26, -28, 10, -81, -31, -81, -35, -19, -55, -27, -4,
+ -4, 23, 49, 88, 85, 22, 1, 0, -3, 18, -22, -12, -39, -14, -47, -67,
+ -53, -53, -33, -20, 8, -22, 25, 30, 6, -19, -62, -19, 10, 83, 57, -67,
+ -88, -52, 63, 127, 42, -63, -121, -70, 82, 111, 46, -43, -114, -35, 45, 73,
+ 39, -19, -54, -41, 9, 49, 24, -16, -14, -9, -6, 10, 8, -22, 25, 30,
+ 6, -19, -62, -19, 10, 83, 57, -67, -88, -52, 63, 127, 42, -63, -121, -70,
+ 82, 111, 46, -43, -114, -35, 45, 73, 39, -19, -54, -41, 9, 49, 24, -16,
+ -14, -9, -6, 10, 8, -22, 25, 30, 6, -19, -62, -19, 10, 83, 57, -67,
+ -88, -52, 63, 127, 42, -63, -121, -70, 82, 111, 46, -43, -114, -35, 45, 73,
+ 39, -19, -54, -41, 9, 49, 24, -16, -14, -9, -6, 10, -18, -15, 80, 127,
+ 20, 18, 72, 60, -36, 22, 33, -73, -105, -27, 29, 25, 8, 43, 23, -22,
+ -35, -14, -8, -34, -25, -5, -18, -46, -13, -28, -3, -19, -10, -28, -10, 17,
+ -18, 96, 127, 123, 127, 126, 126, 95, 61, 22, -17, -56, -95, -118, -123, -128,
+ -128, -128, -128, -128, -128, -128, -128, -124, -93, -55, -17, 25, 62, 100, 107, 119,
+ 119, 127, 41, -4, -4, -10, -27, -62, -102, -128, -117, -85, -76, -49, -25, -7,
+ 15, 24, 14, 0, -9, -12, -3, 33, 81, 105, 111, 101, 80, 62, 45, 30,
+ 16, 8, 1, -4, 126, 86, -44, -68, -88, -63, 4, 84, 127, 14, -29, -51,
+ -127, -47, 58, 87, 82, 24, -17, -112, -113, 18, 46, 72, 102, 19, -72, -118,
+ -51, 9, 27, 126, 19, -53, -114, -108, 15, 122, 111, 21, -71, -70, 5, 59,
+ 83, 23, -72, -84, -49, 12, 37, 38, -28, -65, -26, 7, -2, 11, 5, 23,
+ 68, 19, 27, -54, 15, -24, -31, 46, 71, 81, -24, -33, -87, -41, -71, -61,
+ -70, -36, -44, -3, 12, -25, -20, 20, 68, 85, 127, 15, 50, -5, 27, 51,
+ 82, 106, 127, 118, 101, 92, 74, 52, 39, 37, 14, -41, -105, -108, -94, -119,
+ -124, -93, -57, -40, -37, -45, -51, -41, -5, 51, 127, 127, -128, -128, -128, -128,
+ -128, -128, -128, -128, -128, -128, 127, 127, 127, 127, 127, 127, 127, 127, 127, 127,
+ -128, -17, -22, -96
+};
+
+const EAS_U32 eas_sampleLengths[] =
+{
+ 11754, 10954, 10295, 9922, 7489, 5779, 5462, 3913,
+ 3779, 3093, 3057, 3024, 2818, 2776, 2171, 2168,
+ 2052, 1902, 1835, 1614, 1603, 1528, 1517, 1480,
+ 1455, 1424, 1387, 1302, 1262, 1254, 1230, 1227,
+ 1185, 1181, 1178, 1168, 1132, 1132, 1120, 1034,
+ 1033, 1018, 994, 964, 926, 907, 886, 881,
+ 866, 830, 817, 816, 813, 749, 748, 739,
+ 720, 652, 610, 610, 583, 564, 561, 556,
+ 549, 542, 535, 530, 530, 516, 508, 492,
+ 478, 461, 448, 437, 431, 423, 418, 402,
+ 403, 400, 394, 387, 387, 367, 357, 347,
+ 347, 341, 336, 334, 329, 325, 312, 294,
+ 284, 277, 265, 255, 233, 230, 213, 207,
+ 205, 194, 193, 184, 181, 181, 167, 164,
+ 158, 152, 152, 145, 139, 128, 103, 100,
+ 88, 87, 84, 84, 72, 71, 55, 46,
+ 45, 43, 40, 40, 40, 37, 35, 32,
+ 32, 30, 29, 27, 23, 22, 21, 21,
+ 21, 21, 20
+};
+
+const EAS_U32 eas_sampleOffsets[] =
+{
+ 0x00000000, 0x00002dea, 0x000058b4, 0x000080eb, 0x0000a7ad, 0x0000c4ee, 0x0000db81, 0x0000f0d7,
+ 0x00010020, 0x00010ee3, 0x00011af8, 0x000126e9, 0x000132b9, 0x00013dbb, 0x00014893, 0x0001510e,
+ 0x00015986, 0x0001618a, 0x000168f8, 0x00017023, 0x00017671, 0x00017cb4, 0x000182ac, 0x00018899,
+ 0x00018e61, 0x00019410, 0x000199a0, 0x00019f0b, 0x0001a421, 0x0001a90f, 0x0001adf5, 0x0001b2c3,
+ 0x0001b78e, 0x0001bc2f, 0x0001c0cc, 0x0001c566, 0x0001c9f6, 0x0001ce62, 0x0001d2ce, 0x0001d72e,
+ 0x0001db38, 0x0001df41, 0x0001e33b, 0x0001e71d, 0x0001eae1, 0x0001ee7f, 0x0001f20a, 0x0001f580,
+ 0x0001f8f1, 0x0001fc53, 0x0001ff91, 0x000202c2, 0x000205f2, 0x0002091f, 0x00020c0c, 0x00020ef8,
+ 0x000211db, 0x000214ab, 0x00021737, 0x00021999, 0x00021bfb, 0x00021e42, 0x00022076, 0x000222a7,
+ 0x000224d3, 0x000226f8, 0x00022916, 0x00022b2d, 0x00022d3f, 0x00022f51, 0x00023155, 0x00023351,
+ 0x0002353d, 0x0002371b, 0x000238e8, 0x00023aa8, 0x00023c5d, 0x00023e0c, 0x00023fb3, 0x00024155,
+ 0x000242e7, 0x0002447a, 0x0002460a, 0x00024794, 0x00024917, 0x00024a9a, 0x00024c09, 0x00024d6e,
+ 0x00024ec9, 0x00025024, 0x00025179, 0x000252c9, 0x00025417, 0x00025560, 0x000256a5, 0x000257dd,
+ 0x00025903, 0x00025a1f, 0x00025b34, 0x00025c3d, 0x00025d3c, 0x00025e25, 0x00025f0b, 0x00025fe0,
+ 0x000260af, 0x0002617c, 0x0002623e, 0x000262ff, 0x000263b7, 0x0002646c, 0x00026521, 0x000265c8,
+ 0x0002666c, 0x0002670a, 0x000267a2, 0x0002683a, 0x000268cb, 0x00026956, 0x000269d6, 0x00026a3d,
+ 0x00026aa1, 0x00026af9, 0x00026b50, 0x00026ba4, 0x00026bf8, 0x00026c40, 0x00026c87, 0x00026cbe,
+ 0x00026cec, 0x00026d19, 0x00026d44, 0x00026d6c, 0x00026d94, 0x00026dbc, 0x00026de1, 0x00026e04,
+ 0x00026e24, 0x00026e44, 0x00026e62, 0x00026e7f, 0x00026e9a, 0x00026eb1, 0x00026ec7, 0x00026edc,
+ 0x00026ef1, 0x00026f06, 0x00026f1b
+};
+
+/*----------------------------------------------------------------------------
+ * S_EAS
+ *----------------------------------------------------------------------------
+*/
+const S_EAS easSoundLib =
+{
+ 0x01534145,
+ 0x00105622,
+ eas_banks,
+ eas_programs,
+ eas_regions,
+ eas_articulations,
+ eas_sampleLengths,
+ eas_sampleOffsets,
+ eas_samples,
+ 0,
+ 1,
+ 1,
+ 364,
+ 184,
+ 147,
+ 0
+}; /* end S_EAS */
+
+/*----------------------------------------------------------------------------
+ * Statistics
+ *
+ * Number of banks: 1
+ * Number of programs: 1
+ * Number of regions: 364
+ * Number of articulations: 184
+ * Number of samples: 147
+ * Size of sample pool: 159412
+ *----------------------------------------------------------------------------
+*/
+/* end C:\Sonic\Trunk\EASLib\WTLibrary\wt_22khz.c */
diff --git a/arm-wt-22k/misc/MMAPI.zip b/arm-wt-22k/misc/MMAPI.zip Binary files differnew file mode 100644 index 0000000..c185dbd --- /dev/null +++ b/arm-wt-22k/misc/MMAPI.zip diff --git a/arm-wt-22k/misc/eas_host.c b/arm-wt-22k/misc/eas_host.c new file mode 100644 index 0000000..89eb87a --- /dev/null +++ b/arm-wt-22k/misc/eas_host.c @@ -0,0 +1,787 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_host.c
+ *
+ * Contents and purpose:
+ * This file contains the host wrapper functions for stdio, stdlib, etc.
+ * This is a sample version that wraps the standard library functions.
+ * Modify this file to suit the needs of your particular system.
+ *
+ * EAS_MAX_FILE_HANDLES sets the maximum number of MIDI streams within
+ * a MIDI type 1 file that can be played. To maintain efficiency, data
+ * is buffered locally when byte access is used (EAS_HWGetByte). The
+ * size of the buffer is set by EAS_FILE_BUFFER_SIZE.
+ *
+ * EAS_HW_FILE is a structure to support local file buffering. It
+ * comprises the OS File handle, some data related to the local file
+ * buffer, the position of the next byte of data to be read, the dup
+ * flag which when set, indicates that the handle has been duplicated,
+ * and the data buffer. Since the data buffer is only used for byte
+ * access, it does not need to be large.
+ *
+ * If the file system supports duplicate file handles and buffering,
+ * this entire subsystem can be replaced with direct calls to the
+ * native file I/O routines.
+ *
+ * If the system has enough memory to support reading the entire file
+ * into memory, it will be much more efficient to do so on the call to
+ * EAS_HWOpenFile and then close the file. Simply substitute a memory
+ * pointer for the FILE* pointer. Calls to EAS_HW_DupHandle will work
+ * as they do in this version. In the call to EAS_HWCloseFile, instead
+ * of calling fclose, free the memory containing the file data.
+ *
+ * Copyright 2005 Sonic Network Inc.
+ + * Licensed under the Apache License, Version 2.0 (the "License"); + * you may not use this file except in compliance with the License. + * You may obtain a copy of the License at + * + * http://www.apache.org/licenses/LICENSE-2.0 + * + * Unless required by applicable law or agreed to in writing, software + * distributed under the License is distributed on an "AS IS" BASIS, + * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. + * See the License for the specific language governing permissions and + * limitations under the License. + *
+ *----------------------------------------------------------------------------
+ * Revision Control:
+ * $Revision: 853 $
+ * $Date: 2007-09-05 09:54:17 -0700 (Wed, 05 Sep 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifdef _lint
+#include "lint_stdlib.h"
+#else
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#endif
+
+#include "eas_host.h"
+
+// #define DEBUG_FILE_IO
+
+/* Only for debugging LED, vibrate, and backlight functions */
+#include "eas_report.h"
+
+#ifndef EAS_MAX_FILE_HANDLES
+#define EAS_MAX_FILE_HANDLES 32
+#endif
+
+#ifndef EAS_FILE_BUFFER_SIZE
+#define EAS_FILE_BUFFER_SIZE 32
+#endif
+
+/*
+ * this structure and the related function are here
+ * to support the ability to create duplicate handles
+ * and buffering into a single file. If the OS supports
+ * duplicate file handles natively, this code can be
+ * stripped to eliminate double-buffering.
+ */
+typedef struct eas_hw_file_tag
+{
+ FILE *pFile;
+ EAS_I32 bytesInBuffer;
+ EAS_I32 readIndex;
+ EAS_I32 filePos;
+ EAS_BOOL dup;
+ EAS_U8 buffer[EAS_FILE_BUFFER_SIZE];
+} EAS_HW_FILE;
+
+typedef struct eas_hw_inst_data_tag
+{
+ EAS_HW_FILE files[EAS_MAX_FILE_HANDLES];
+} EAS_HW_INST_DATA;
+
+/* local memory for files and streams */
+#ifdef _STATIC_MEMORY
+EAS_HW_INST_DATA fileData;
+#endif
+
+/*----------------------------------------------------------------------------
+ * EAS_HWInit
+ *
+ * Initialize host wrapper interface
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWInit (EAS_HW_DATA_HANDLE *pHWInstData)
+{
+
+ /* need to track file opens for duplicate handles */
+#ifndef _STATIC_MEMORY
+ *pHWInstData = malloc(sizeof(EAS_HW_INST_DATA));
+ if (!(*pHWInstData))
+ return EAS_ERROR_MALLOC_FAILED;
+#else
+ *pHWInstData = &fileData;
+#endif
+ EAS_HWMemSet(*pHWInstData, 0, sizeof(EAS_HW_INST_DATA));
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * EAS_HWShutdown
+ *
+ * Shut down host wrapper interface
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWShutdown (EAS_HW_DATA_HANDLE hwInstData)
+{
+
+#ifndef _STATIC_MEMORY
+ free(hwInstData);
+#endif
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMalloc
+ *
+ * Allocates dynamic memory
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+#ifdef _STATIC_MEMORY
+/*lint -esym(715, size) not used in static memory model */
+#endif
+void *EAS_HWMalloc (EAS_HW_DATA_HANDLE hwInstData, EAS_I32 size)
+{
+#ifdef _STATIC_MEMORY
+ return NULL;
+#else
+ return malloc((EAS_U32)size);
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFree
+ *
+ * Frees dynamic memory
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+#ifdef _STATIC_MEMORY
+/*lint -esym(715, p) not used in static memory model */
+#endif
+void EAS_HWFree(EAS_HW_DATA_HANDLE hwInstData, void *p)
+{
+#ifndef _STATIC_MEMORY
+ free(p);
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemCpy
+ *
+ * Copy memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+void *EAS_HWMemCpy (void *dest, const void *src, EAS_I32 amount)
+{
+ return memcpy(dest,src,(size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemSet
+ *
+ * Set memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+void *EAS_HWMemSet (void *dest, int val, EAS_I32 amount)
+{
+ return memset(dest,val,(size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWMemCmp
+ *
+ * Compare memory wrapper
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_I32 EAS_HWMemCmp (const void *s1, const void *s2, EAS_I32 amount)
+{
+ return (EAS_I32) memcmp(s1, s2, (size_t) amount);
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWOpenFile
+ *
+ * Open a file for read or write
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWOpenFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_LOCATOR locator, EAS_FILE_HANDLE *pFile, EAS_FILE_MODE mode)
+{
+ EAS_HW_FILE *file;
+ int i;
+
+ /* set return value to NULL */
+ *pFile = NULL;
+
+ /* only support read mode at this time */
+ if (mode != EAS_FILE_READ)
+ return EAS_ERROR_INVALID_FILE_MODE;
+
+ /* find an empty entry in the file table */
+ file = hwInstData->files;
+ for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+ {
+ /* is this slot being used? */
+ if (file->pFile == NULL)
+ {
+ /* open the file */
+ if (locator->path)
+ file->pFile = fopen((const char*) locator->path, "rb");
+ if (file->pFile == NULL)
+ return EAS_ERROR_FILE_OPEN_FAILED;
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWOpenFile: Open file %d\n", i);
+#endif
+
+ /* initialize some values */
+ file->bytesInBuffer = 0;
+ file->readIndex = 0;
+ file->filePos = 0;
+ file->dup = EAS_FALSE;
+
+ *pFile = file;
+ return EAS_SUCCESS;
+ }
+ file++;
+ }
+
+ /* too many open files */
+ return EAS_ERROR_MAX_FILES_OPEN;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFillBuffer
+ *
+ * Fill buffer from file
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFillBuffer (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file)
+{
+ /* reposition the file pointer */
+ if (fseek(file->pFile, file->filePos, SEEK_SET) != 0)
+ return EAS_ERROR_FILE_SEEK;
+
+ /* read some data from the file */
+ file->bytesInBuffer = (EAS_I32) fread(file->buffer, 1, EAS_FILE_BUFFER_SIZE, file->pFile);
+ file->readIndex = 0;
+ if (file->bytesInBuffer == 0)
+ return EAS_EOF;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWReadFile
+ *
+ * Read data from a file
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWReadFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *pBuffer, EAS_I32 n, EAS_I32 *pBytesRead)
+{
+ EAS_RESULT result;
+ EAS_I32 temp;
+ EAS_U8 *p = pBuffer;
+ EAS_I32 bytesLeft = n;
+
+ *pBytesRead = 0;
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWReadFile: Reading %d bytes from position %d\n", n, file->filePos);
+#endif
+
+ /* try to fulfill request from buffer */
+ for (;bytesLeft > 0;)
+ {
+ /* how many bytes can we get from buffer? */
+ temp = file->bytesInBuffer - file->readIndex;
+ if (temp > bytesLeft)
+ temp = bytesLeft;
+
+ /* copy data from buffer */
+ EAS_HWMemCpy(p, &file->buffer[file->readIndex], temp);
+ *pBytesRead += temp;
+ file->readIndex += temp;
+ file->filePos += temp;
+ bytesLeft -= temp;
+ p += temp;
+
+ /* don't refill buffer if request is bigger than buffer */
+ if ((bytesLeft == 0) || (bytesLeft >= EAS_FILE_BUFFER_SIZE))
+ break;
+
+ /* refill buffer */
+ if ((result = EAS_HWFillBuffer(hwInstData, file)) != EAS_SUCCESS)
+ return result;
+ }
+
+ /* more to read? do unbuffered read directly to target memory */
+ if (bytesLeft)
+ {
+
+ /* position the file pointer */
+ if (fseek(file->pFile, file->filePos, SEEK_SET) != 0)
+ return EAS_ERROR_FILE_SEEK;
+
+ /* read data in the buffer */
+ /*lint -e{826} lint doesn't like this with STATIC_MEMORY defined for some reason */
+ temp = (EAS_I32) fread(p, 1, (size_t) bytesLeft, file->pFile);
+ *pBytesRead += temp;
+ file->filePos += temp;
+
+ /* reset buffer info */
+ file->bytesInBuffer = 0;
+ file->readIndex = 0;
+ }
+
+#ifdef DEBUG_FILE_IO
+ {
+#define BYTES_PER_LINE 16
+ char str[BYTES_PER_LINE * 3 + 1];
+ EAS_INT i;
+ for (i = 0; i < (n > BYTES_PER_LINE ? BYTES_PER_LINE : n) ; i ++)
+ sprintf(&str[i*3], "%02x ", ((EAS_U8*)pBuffer)[i]);
+ if (i)
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "%s\n", str);
+ }
+#endif
+
+ /* were n bytes read? */
+ if (*pBytesRead != n)
+ return EAS_EOF;
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetByte
+ *
+ * Read a byte from a file
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWGetByte (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p)
+{
+ EAS_RESULT result;
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* use local buffer - do we have any data? */
+ if (file->readIndex >= file->bytesInBuffer)
+ {
+ if ((result = EAS_HWFillBuffer(hwInstData, file)) != EAS_SUCCESS)
+ return result;
+
+ /* if nothing to read, return EOF */
+ if (file->bytesInBuffer == 0)
+ return EAS_EOF;
+ }
+
+ /* get a character from the buffer */
+ *((EAS_U8*) p) = file->buffer[file->readIndex++];
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWGetByte: Reading from position %d, byte = 0x%02x\n", file->filePos, *(EAS_U8*)p);
+#endif
+
+ file->filePos++;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetWord
+ *
+ * Read a 16-bit value from the file
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWGetWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst)
+{
+ EAS_RESULT result;
+ EAS_I32 count;
+ EAS_U8 c[2];
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWGetWord: Reading 2 bytes from position %d\n", file->filePos);
+#endif
+
+ /* read 2 bytes from the file */
+ if ((result = EAS_HWReadFile(hwInstData, file, c, 2, &count)) != EAS_SUCCESS)
+ return result;
+
+ /* order them as requested */
+ if (msbFirst)
+ *((EAS_U16*) p) = ((EAS_U16) c[0] << 8) | c[1];
+ else
+ *((EAS_U16*) p) = ((EAS_U16) c[1] << 8) | c[0];
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetDWord
+ *
+ * Read a 16-bit value from the file
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWGetDWord (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, void *p, EAS_BOOL msbFirst)
+{
+ EAS_RESULT result;
+ EAS_I32 count;
+ EAS_U8 c[4];
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWGetDWord: Reading 4 bytes from position %d\n", file->filePos);
+#endif
+
+ /* read 4 bytes from the file */
+ if ((result = EAS_HWReadFile(hwInstData, file, c, 4, &count)) != EAS_SUCCESS)
+ return result;
+
+ /* order them as requested */
+ if (msbFirst)
+ *((EAS_U32*) p) = ((EAS_U32) c[0] << 24) | ((EAS_U32) c[1] << 16) | ((EAS_U32) c[2] << 8) | c[3];
+ else
+ *((EAS_U32*) p) = ((EAS_U32) c[3] << 24) | ((EAS_U32) c[2] << 16) | ((EAS_U32) c[1] << 8) | c[0];
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFilePos
+ *
+ * Returns the current location in the file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFilePos (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pPosition)
+{
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ *pPosition = file->filePos;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileSeek
+ *
+ * Seek to a specific location in the file
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileSeek (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position)
+{
+ EAS_I32 newIndex;
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWFileSeek: Seeking to new position %d\n", file->filePos);
+#endif
+
+ /* is new position in current buffer? */
+ newIndex = position - file->filePos + file->readIndex;
+ if ((newIndex >= 0) && (newIndex < file->bytesInBuffer))
+ {
+ file->readIndex = newIndex;
+ file->filePos = position;
+ return EAS_SUCCESS;
+ }
+
+ /* save new position and reset buffer info so EAS_HWGetByte doesn't fail */
+ file->filePos = position;
+ file->bytesInBuffer = 0;
+ file->readIndex = 0;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileSeekOfs
+ *
+ * Seek forward or back relative to the current position
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileSeekOfs (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 position)
+{
+ EAS_I32 temp;
+
+#ifdef DEBUG_FILE_IO
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "EAS_HWFileSeekOfs: Seeking to new position %d\n", file->filePos + position);
+#endif
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* is new position in current buffer? */
+ temp = position + file->readIndex;
+ if ((temp >= 0) && (temp < file->bytesInBuffer))
+ {
+ file->readIndex = temp;
+ file->filePos += position;
+ return EAS_SUCCESS;
+ }
+
+ /* save new position and reset buffer info so EAS_HWGetByte doesn't fail */
+ file->filePos += position;
+ file->bytesInBuffer = 0;
+ file->readIndex = 0;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWFileLength
+ *
+ * Return the file length
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWFileLength (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_I32 *pLength)
+{
+ long pos;
+
+ /* check handle integrity */
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ if ((pos = ftell(file->pFile)) == -1L)
+ return EAS_ERROR_FILE_LENGTH;
+ if (fseek(file->pFile, 0L, SEEK_END) != 0)
+ return EAS_ERROR_FILE_LENGTH;
+ if ((*pLength = ftell(file->pFile)) == -1L)
+ return EAS_ERROR_FILE_LENGTH;
+ if (fseek(file->pFile, pos, SEEK_SET) != 0)
+ return EAS_ERROR_FILE_LENGTH;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWDupHandle
+ *
+ * Duplicate a file handle
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWDupHandle (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file, EAS_FILE_HANDLE* pDupFile)
+{
+ EAS_HW_FILE *dupfile;
+ int i;
+
+ /* check handle integrity */
+ *pDupFile = NULL;
+ if (file->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* find an empty entry in the file table */
+ dupfile = hwInstData->files;
+ for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+ {
+ /* is this slot being used? */
+ if (dupfile->pFile == NULL)
+ {
+
+ /* copy info from the handle to be duplicated */
+ dupfile->filePos = file->filePos;
+ dupfile->pFile = file->pFile;
+
+ /* set the duplicate handle flag */
+ dupfile->dup = file->dup = EAS_TRUE;
+
+ /* initialize some values */
+ dupfile->bytesInBuffer = 0;
+ dupfile->readIndex = 0;
+
+ *pDupFile = dupfile;
+ return EAS_SUCCESS;
+ }
+ dupfile++;
+ }
+
+ /* too many open files */
+ return EAS_ERROR_MAX_FILES_OPEN;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWClose
+ *
+ * Wrapper for fclose function
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWCloseFile (EAS_HW_DATA_HANDLE hwInstData, EAS_FILE_HANDLE file1)
+{
+ EAS_HW_FILE *file2,*dupFile;
+ int i;
+
+ /* check handle integrity */
+ if (file1->pFile == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* check for duplicate handle */
+ if (file1->dup)
+ {
+ dupFile = NULL;
+ file2 = hwInstData->files;
+ for (i = 0; i < EAS_MAX_FILE_HANDLES; i++)
+ {
+ /* check for duplicate */
+ if ((file1 != file2) && (file2->pFile == file1->pFile))
+ {
+ /* is there more than one duplicate? */
+ if (dupFile != NULL)
+ {
+ /* clear this entry and return */
+ file1->pFile = NULL;
+ return EAS_SUCCESS;
+ }
+
+ /* this is the first duplicate found */
+ dupFile = file2;
+ }
+ file2++;
+ }
+
+ /* there is only one duplicate, clear the dup flag */
+ if (dupFile)
+ dupFile->dup = EAS_FALSE;
+ else
+ /* if we get here, there's a serious problem */
+ return EAS_ERROR_HANDLE_INTEGRITY;
+
+ /* clear this entry and return */
+ file1->pFile = NULL;
+ return EAS_SUCCESS;
+ }
+
+ /* no duplicates - close the file */
+ if (fclose(file1->pFile) != 0)
+ return EAS_ERROR_CLOSE_FAILED;
+
+ /* clear this entry and return */
+ file1->pFile = NULL;
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWVibrate
+ *
+ * Turn on/off vibrate function
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWVibrate (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "Vibrate state: %d\n", state);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWLED
+ *
+ * Turn on/off LED
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWLED (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "LED state: %d\n", state);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWBackLight
+ *
+ * Turn on/off backlight
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_RESULT EAS_HWBackLight (EAS_HW_DATA_HANDLE hwInstData, EAS_BOOL state)
+{
+ EAS_ReportX(_EAS_SEVERITY_NOFILTER, "Backlight state: %d\n", state);
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWYield
+ *
+ * This function is called periodically by the EAS library to give the
+ * host an opportunity to allow other tasks to run. There are two ways to
+ * use this call:
+ *
+ * If you have a multi-tasking OS, you can call the yield function in the
+ * OS to allow other tasks to run. In this case, return EAS_FALSE to tell
+ * the EAS library to continue processing when control returns from this
+ * function.
+ *
+ * If tasks run in a single thread by sequential function calls (sometimes
+ * call a "commutator loop"), return EAS_TRUE to cause the EAS Library to
+ * return to the caller. Be sure to check the number of bytes rendered
+ * before passing the audio buffer to the codec - it may not be filled.
+ * The next call to EAS_Render will continue processing until the buffer
+ * has been filled.
+ *
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, hwInstData) hwInstData available for customer use */
+EAS_BOOL EAS_HWYield (EAS_HW_DATA_HANDLE hwInstData)
+{
+ /* put your code here */
+ return EAS_FALSE;
+}
diff --git a/arm-wt-22k/vectors/Leadsol.mxmf b/arm-wt-22k/vectors/Leadsol.mxmf Binary files differnew file mode 100644 index 0000000..239f500 --- /dev/null +++ b/arm-wt-22k/vectors/Leadsol.mxmf diff --git a/arm-wt-22k/vectors/Leadsol_out.wav b/arm-wt-22k/vectors/Leadsol_out.wav Binary files differnew file mode 100644 index 0000000..43d326c --- /dev/null +++ b/arm-wt-22k/vectors/Leadsol_out.wav diff --git a/arm-wt-22k/vectors/WAVEtest.wav b/arm-wt-22k/vectors/WAVEtest.wav Binary files differnew file mode 100644 index 0000000..0d8178a --- /dev/null +++ b/arm-wt-22k/vectors/WAVEtest.wav diff --git a/arm-wt-22k/vectors/WAVEtest_out.wav b/arm-wt-22k/vectors/WAVEtest_out.wav Binary files differnew file mode 100644 index 0000000..1a66130 --- /dev/null +++ b/arm-wt-22k/vectors/WAVEtest_out.wav diff --git a/arm-wt-22k/vectors/abba.imy b/arm-wt-22k/vectors/abba.imy new file mode 100644 index 0000000..9a056e6 --- /dev/null +++ b/arm-wt-22k/vectors/abba.imy @@ -0,0 +1,10 @@ +BEGIN:IMELODY
+VERSION:1.2
+FORMAT:CLASS1.0
+NAME:DanceQueen
+COMPOSER: downloaded at www.ringtones.perbang.dk
+BEAT:112
+STYLE:S1
+VOLUME:V10
+MELODY:*5#c1*4b2*5d1*5#c4*4b3*4a4*4b3.*5#c3*5#c2*5a4*5a3*5#g4*5#g3*5#f4*5#f1*5#c1*4b2*5d1*5#c4*4b3*4a4*4b3.*5#c3*5#c2*4b2*4a1.r2*5#c2*4b3*4b1r4r5*5#c2*4b3*4b2*5#c2*4a3.*4b3.*4#g3*4a3.*4b3.*4#g3*4a4*4#g4*4#f2.*5#c3*4b3.*4a3*4#g2*4a3*4a1r4*4#g2*4a3*4a2.*4b4*4a3*4#g4*4#g3.*4a3.*4a1
+END:IMELODY
diff --git a/arm-wt-22k/vectors/abba_out.wav b/arm-wt-22k/vectors/abba_out.wav Binary files differnew file mode 100644 index 0000000..acf40d5 --- /dev/null +++ b/arm-wt-22k/vectors/abba_out.wav diff --git a/arm-wt-22k/vectors/ants.mid b/arm-wt-22k/vectors/ants.mid Binary files differnew file mode 100644 index 0000000..d4ead53 --- /dev/null +++ b/arm-wt-22k/vectors/ants.mid diff --git a/arm-wt-22k/vectors/ants_out.wav b/arm-wt-22k/vectors/ants_out.wav Binary files differnew file mode 100644 index 0000000..abb724d --- /dev/null +++ b/arm-wt-22k/vectors/ants_out.wav diff --git a/arm-wt-22k/vectors/greensleeves.rtttl b/arm-wt-22k/vectors/greensleeves.rtttl new file mode 100644 index 0000000..37865d1 --- /dev/null +++ b/arm-wt-22k/vectors/greensleeves.rtttl @@ -0,0 +1 @@ +Greensleaves:d=4, o=5, b=140:g, 2a#, c6, d.6, 8d#6, d6, 2c6, a, f., 8g, a, 2a#, g, g., 8f, g, 2a, f, 2d, g, 2a#, c6, d.6, 8e6, d6, 2c6, a, f., 8g, a, a#., 8a, g, f#., 8e, f#, 2g
\ No newline at end of file diff --git a/arm-wt-22k/vectors/greensleeves_out.wav b/arm-wt-22k/vectors/greensleeves_out.wav Binary files differnew file mode 100644 index 0000000..5d61fb2 --- /dev/null +++ b/arm-wt-22k/vectors/greensleeves_out.wav diff --git a/arm-wt-22k/vectors/test.ota b/arm-wt-22k/vectors/test.ota Binary files differnew file mode 100644 index 0000000..7aa5f99 --- /dev/null +++ b/arm-wt-22k/vectors/test.ota diff --git a/arm-wt-22k/vectors/test_out.wav b/arm-wt-22k/vectors/test_out.wav Binary files differnew file mode 100644 index 0000000..45ef142 --- /dev/null +++ b/arm-wt-22k/vectors/test_out.wav |