diff options
| 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-hybrid-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-hybrid-22k')
92 files changed, 37895 insertions, 0 deletions
diff --git a/arm-hybrid-22k/Makefile b/arm-hybrid-22k/Makefile new file mode 100644 index 0000000..5648139 --- /dev/null +++ b/arm-hybrid-22k/Makefile @@ -0,0 +1,66 @@ +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_fmengine.c \ + lib_src/eas_fmsynth.c \ + lib_src/eas_fmtables.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_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_wavefile.c \ + lib_src/eas_wavefiledata.c \ + lib_src/eas_wtengine.c \ + lib_src/eas_wtsynth.c \ + lib_src/hybrid_22khz_mcu.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 + +LOCAL_CFLAGS+= -O2 -D UNIFIED_DEBUG_MESSAGES -D EAS_HYBRID_SYNTH \ + -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER \ + -D _WAVE_PARSER -D _REVERB_ENABLED \ + -D _CHORUS_ENABLED -D NUM_OUTPUT_CHANNELS=2 \ + -D _SAMPLE_RATE_22050 -D MAX_SYNTH_VOICES=32 \ + -D NUM_PRIMARY_VOICES=8 \ + -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER \ + -D _IMA_DECODER + +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 + +include $(BUILD_SHARED_LIBRARY) diff --git a/arm-hybrid-22k/bin/arm-hybrid-22k b/arm-hybrid-22k/bin/arm-hybrid-22k Binary files differnew file mode 100755 index 0000000..ea80a08 --- /dev/null +++ b/arm-hybrid-22k/bin/arm-hybrid-22k diff --git a/arm-hybrid-22k/host_src/.DS_Store b/arm-hybrid-22k/host_src/.DS_Store Binary files differnew file mode 100644 index 0000000..5008ddf --- /dev/null +++ b/arm-hybrid-22k/host_src/.DS_Store diff --git a/arm-hybrid-22k/host_src/._.DS_Store b/arm-hybrid-22k/host_src/._.DS_Store Binary files differnew file mode 100644 index 0000000..c9474ea --- /dev/null +++ b/arm-hybrid-22k/host_src/._.DS_Store diff --git a/arm-hybrid-22k/host_src/arm-hybrid-22k.mak b/arm-hybrid-22k/host_src/arm-hybrid-22k.mak new file mode 100644 index 0000000..8874f31 --- /dev/null +++ b/arm-hybrid-22k/host_src/arm-hybrid-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_HYBRID_SYNTH -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED $<
+
+%.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-hybrid-22k: $(OBJS)
+ $(LD) -o $@ $(OBJS) libarm-hybrid-22k.a -lm
+
diff --git a/arm-hybrid-22k/host_src/eas.h b/arm-hybrid-22k/host_src/eas.h new file mode 100644 index 0000000..0bb04fe --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_build.h b/arm-hybrid-22k/host_src/eas_build.h new file mode 100644 index 0000000..3663322 --- /dev/null +++ b/arm-hybrid-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_0cbd200ab054487f9ece7d3c8852e426_
+#define _GUID_0cbd200ab054487f9ece7d3c8852e426_
+
+#define _BUILD_VERSION_ "0cbd200a-b054-487f-9ece-7d3c8852e426"
+#define _BUILD_TIME_ 0x4743b9dc
+
+#endif /* _GUID_0cbd200ab054487f9ece7d3c8852e426_ */
diff --git a/arm-hybrid-22k/host_src/eas_chorus.h b/arm-hybrid-22k/host_src/eas_chorus.h new file mode 100644 index 0000000..0e9057f --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_config.c b/arm-hybrid-22k/host_src/eas_config.c new file mode 100644 index 0000000..c45fbb7 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_config.h b/arm-hybrid-22k/host_src/eas_config.h new file mode 100644 index 0000000..d16be4a --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_debugmsgs.h b/arm-hybrid-22k/host_src/eas_debugmsgs.h new file mode 100644 index 0000000..3fff38f --- /dev/null +++ b/arm-hybrid-22k/host_src/eas_debugmsgs.h @@ -0,0 +1,44 @@ +/* Auto-generated from source file: eas_chorusdata.c */
+/* Auto-generated from source file: eas_imelodydata.c */
+/* 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_voicemgt.c */
+/* Auto-generated from source file: eas_ota.c */
+/* Auto-generated from source file: eas_mixbuf.c */
+/* Auto-generated from source file: eas_rtttl.c */
+/* Auto-generated from source file: eas_reverb.c */
+/* Auto-generated from source file: eas_fmsynth.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_fmengine.c */
+/* Auto-generated from source file: eas_smfdata.c */
+/* Auto-generated from source file: eas_fmtables.c */
+/* Auto-generated from source file: eas_imelody.c */
+/* Auto-generated from source file: eas_public.c */
+/* Auto-generated from source file: eas_rtttldata.c */
+/* Auto-generated from source file: eas_reverbdata.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_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" },
+/* 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-hybrid-22k/host_src/eas_host.h b/arm-hybrid-22k/host_src/eas_host.h new file mode 100644 index 0000000..0db0e30 --- /dev/null +++ b/arm-hybrid-22k/host_src/eas_host.h @@ -0,0 +1,83 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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);
+
+/* 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-hybrid-22k/host_src/eas_hostmm.c b/arm-hybrid-22k/host_src/eas_hostmm.c new file mode 100644 index 0000000..7e58838 --- /dev/null +++ b/arm-hybrid-22k/host_src/eas_hostmm.c @@ -0,0 +1,660 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_hostmm.c
+ *
+ * Contents and purpose:
+ * This file contains the host wrapper functions for stdio, stdlib, etc.
+ * This is a sample version that maps the requested files to an
+ * allocated memory block and uses in-memory pointers to replace
+ * file system calls. The file locator (EAS_FILE_LOCATOR) handle passed
+ * HWOpenFile is the same one that is passed to EAS_OpenFile. If your
+ * system stores data in fixed locations (such as flash) instead of
+ * using a file system, you can use the locator handle to point to
+ * your memory. You will need a way of knowing the length of the
+ * data stored at that location in order to respond correctly in the
+ * HW_FileLength function.
+ *
+ * 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 base memory pointer, the file read pointer, and
+ * the dup flag, which when sets, indicates that the file handle has
+ * been duplicated. If your system uses in-memory resources, you
+ * can eliminate the duplicate handle logic, and simply copy the
+ * base memory pointer and file read pointer to the duplicate handle.
+ *
+ * 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>
+#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;
+ EAS_U8 *buffer;
+} EAS_HW_FILE;
+
+typedef struct eas_hw_inst_data_tag
+{
+ EAS_HW_FILE files[EAS_MAX_FILE_HANDLES];
+} EAS_HW_INST_DATA;
+
+/*----------------------------------------------------------------------------
+ * EAS_HWInit
+ *
+ * Initialize host wrapper interface
+ *
+ *----------------------------------------------------------------------------
+*/
+EAS_RESULT EAS_HWInit (EAS_HW_DATA_HANDLE *pHWInstData)
+{
+
+ /* 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));
+ 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)
+{
+ 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)
+{
+ 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;
+ FILE *ioFile;
+ 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->buffer == NULL)
+ {
+ /* open the file */
+ if ((ioFile = fopen(locator,"rb")) == NULL)
+ return EAS_ERROR_FILE_OPEN_FAILED;
+
+ /* determine the file size */
+ if (fseek(ioFile, 0L, SEEK_END) != 0)
+ return EAS_ERROR_FILE_LENGTH;
+ if ((file->fileSize = ftell(ioFile)) == -1L)
+ return EAS_ERROR_FILE_LENGTH;
+ if (fseek(ioFile, 0L, SEEK_SET) != 0)
+ return EAS_ERROR_FILE_LENGTH;
+
+ /* allocate a buffer */
+ file->buffer = EAS_HWMalloc(hwInstData, file->fileSize);
+ if (file->buffer == NULL)
+ {
+ fclose(ioFile);
+ return EAS_ERROR_MALLOC_FAILED;
+ }
+
+ /* read the file into memory */
+ temp = (int) fread(file->buffer, (size_t) file->fileSize, 1, ioFile);
+
+ /* close the file - don't need it any more */
+ fclose(ioFile);
+
+ /* check for error reading file */
+ if (temp != 1)
+ return EAS_ERROR_FILE_READ_FAILED;
+
+ /* 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->buffer == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* calculate the bytes to read */
+ count = file->fileSize - file->filePos;
+ if (n < count)
+ count = n;
+
+ /* copy the data to the requested location, and advance the pointer */
+ if (count)
+ EAS_HWMemCpy(pBuffer, &file->buffer[file->filePos], 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)
+{
+
+ /* make sure we have a valid handle */
+ if (file->buffer == NULL)
+ return EAS_ERROR_INVALID_HANDLE;
+
+ /* check for end of file */
+ if (file->filePos >= file->fileSize)
+ {
+ *((EAS_U8*) p) = 0;
+ return EAS_EOF;
+ }
+
+ /* get a character from the buffer */
+ *((EAS_U8*) p) = file->buffer[file->filePos++];
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ *
+ * EAS_HWGetWord
+ *
+ * Returns the current location in the 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->buffer == NULL)
+ 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->buffer == NULL)
+ 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->buffer == NULL)
+ 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->buffer == NULL)
+ 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->buffer == 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->buffer == NULL)
+ {
+
+ /* copy info from the handle to be duplicated */
+ dupFile->filePos = file->filePos;
+ dupFile->fileSize = file->fileSize;
+ dupFile->buffer = file->buffer;
+
+ /* 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->buffer == 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->buffer == file1->buffer))
+ {
+ /* is there more than one duplicate? */
+ if (dupFile != NULL)
+ {
+ /* clear this entry and return */
+ file1->buffer = NULL;
+ 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->buffer = NULL;
+ return EAS_SUCCESS;
+ }
+
+ /* no duplicates -free the buffer */
+ EAS_HWFree(hwInstData, file1->buffer);
+
+ /* clear this entry and return */
+ file1->buffer = 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_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-hybrid-22k/host_src/eas_main.c b/arm-hybrid-22k/host_src/eas_main.c new file mode 100644 index 0000000..809a132 --- /dev/null +++ b/arm-hybrid-22k/host_src/eas_main.c @@ -0,0 +1,461 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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;
+
+ /* 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 */
+ if ((reportResult = EAS_OpenFile(easData, filename, &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-hybrid-22k/host_src/eas_report.c b/arm-hybrid-22k/host_src/eas_report.c new file mode 100644 index 0000000..d4dd22c --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_report.h b/arm-hybrid-22k/host_src/eas_report.h new file mode 100644 index 0000000..9d7c8e8 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_reverb.h b/arm-hybrid-22k/host_src/eas_reverb.h new file mode 100644 index 0000000..a2535fb --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_types.h b/arm-hybrid-22k/host_src/eas_types.h new file mode 100644 index 0000000..f0293ef --- /dev/null +++ b/arm-hybrid-22k/host_src/eas_types.h @@ -0,0 +1,268 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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 const void *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-hybrid-22k/host_src/eas_wave.c b/arm-hybrid-22k/host_src/eas_wave.c new file mode 100644 index 0000000..02fed6e --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/host_src/eas_wave.h b/arm-hybrid-22k/host_src/eas_wave.h new file mode 100644 index 0000000..ca388f5 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib/libarm-hybrid-22k.a b/arm-hybrid-22k/lib/libarm-hybrid-22k.a Binary files differnew file mode 100644 index 0000000..f3999f0 --- /dev/null +++ b/arm-hybrid-22k/lib/libarm-hybrid-22k.a diff --git a/arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_filter_gnu.s new file mode 100644 index 0000000..871cd7d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/ARM-E_interpolate_loop_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_interpolate_loop_gnu.s new file mode 100644 index 0000000..847a7f0 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/ARM-E_interpolate_noloop_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_interpolate_noloop_gnu.s new file mode 100644 index 0000000..6343762 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_mastergain_gnu.s new file mode 100644 index 0000000..b4e905b --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s b/arm-hybrid-22k/lib_src/ARM-E_voice_gain_gnu.s new file mode 100644 index 0000000..4517a3d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc b/arm-hybrid-22k/lib_src/ARM_synth_constants_gnu.inc new file mode 100644 index 0000000..b4cd29a --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak b/arm-hybrid-22k/lib_src/arm-hybrid-22k_lib.mak new file mode 100644 index 0000000..43edd6d --- /dev/null +++ b/arm-hybrid-22k/lib_src/arm-hybrid-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=32 -D NUM_PRIMARY_VOICES=8 -D EAS_HYBRID_SYNTH -D _8_BIT_SAMPLES -D _FILTER_ENABLED -D _IMELODY_PARSER -D _RTTTL_PARSER -D _OTA_PARSER -D _WAVE_PARSER -D _REVERB_ENABLED -D _CHORUS_ENABLED -D _IMA_DECODER -D NATIVE_EAS_KERNEL $<
+
+%.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_wavefile.o eas_smfdata.o eas_imelody.o eas_math.o eas_fmengine.o ARM-E_filter_gnu.o eas_chorusdata.o eas_wtengine.o eas_ota.o eas_rtttldata.o eas_reverbdata.o eas_public.o ARM-E_interpolate_loop_gnu.o eas_rtttl.o eas_reverb.o ARM-E_voice_gain_gnu.o eas_fmsynth.o eas_midi.o eas_otadata.o eas_wtsynth.o eas_mixbuf.o eas_imaadpcm.o hybrid_22khz_mcu.o eas_smf.o eas_chorus.o eas_pcm.o eas_mixer.o eas_data.o eas_imelodydata.o eas_pcmdata.o eas_ima_tables.o eas_fmtables.o ARM-E_interpolate_noloop_gnu.o ARM-E_mastergain_gnu.o eas_voicemgt.o
+
+arm-hybrid-22k.a: $(OBJS)
+ $(AR) rc lib$@ $(OBJS)
+
diff --git a/arm-hybrid-22k/lib_src/eas_audioconst.h b/arm-hybrid-22k/lib_src/eas_audioconst.h new file mode 100644 index 0000000..1cfa404 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_chorus.c b/arm-hybrid-22k/lib_src/eas_chorus.c new file mode 100644 index 0000000..bc42237 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_chorusdata.c b/arm-hybrid-22k/lib_src/eas_chorusdata.c new file mode 100644 index 0000000..caee1ed --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_chorusdata.h b/arm-hybrid-22k/lib_src/eas_chorusdata.h new file mode 100644 index 0000000..4420ddd --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_ctype.h b/arm-hybrid-22k/lib_src/eas_ctype.h new file mode 100644 index 0000000..8503870 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_data.c b/arm-hybrid-22k/lib_src/eas_data.c new file mode 100644 index 0000000..bb60ef2 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_data.h b/arm-hybrid-22k/lib_src/eas_data.h new file mode 100644 index 0000000..0a47d04 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_data.h @@ -0,0 +1,131 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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
+
+#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-hybrid-22k/lib_src/eas_effects.h b/arm-hybrid-22k/lib_src/eas_effects.h new file mode 100644 index 0000000..01e64c0 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_fmengine.c b/arm-hybrid-22k/lib_src/eas_fmengine.c new file mode 100644 index 0000000..9c3da66 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_fmengine.c @@ -0,0 +1,785 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_fmengine.c
+ *
+ * Contents and purpose:
+ * Implements the low-level FM synthesizer functions.
+ *
+ * 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: 795 $
+ * $Date: 2007-08-01 00:14:45 -0700 (Wed, 01 Aug 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* includes */
+#include "eas_types.h"
+#include "eas_math.h"
+#include "eas_audioconst.h"
+#include "eas_fmengine.h"
+
+#if defined(EAS_FM_SYNTH) || defined(EAS_HYBRID_SYNTH) || defined(EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH)
+#include "eas_data.h"
+#endif
+
+/* externals */
+extern const EAS_I16 sineTable[];
+extern const EAS_U8 fmScaleTable[16];
+
+// saturation constants for 32-bit to 16-bit conversion
+#define _EAS_MAX_OUTPUT 32767
+#define _EAS_MIN_OUTPUT -32767
+
+static S_FM_ENG_VOICE voices[NUM_FM_VOICES];
+
+/* local prototypes */
+void FM_SynthMixVoice (S_FM_ENG_VOICE *p, EAS_U16 gainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer);
+
+/* used in development environment */
+#if defined(_SATURATION_MONITOR)
+static EAS_BOOL bSaturated = EAS_FALSE;
+
+/*----------------------------------------------------------------------------
+ * FM_CheckSaturation()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Allows the sound development tool to check for saturation at the voice
+ * level. Useful for tuning the level controls.
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Returns true if saturation has occurred since the last time the function
+ * was called.
+ *
+ * Side Effects:
+ * Resets the saturation flag
+ *----------------------------------------------------------------------------
+*/
+EAS_BOOL FM_CheckSaturation ()
+{
+ EAS_BOOL bTemp;
+ bTemp = bSaturated;
+ bSaturated = EAS_FALSE;
+ return bTemp;
+}
+#endif
+
+/*----------------------------------------------------------------------------
+ * FM_Saturate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * This inline function saturates a 32-bit number to 16-bits
+ *
+ * Inputs:
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Returns a 16-bit integer
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_I16 FM_Saturate (EAS_I32 nValue)
+{
+ if (nValue > _EAS_MAX_OUTPUT)
+ {
+#if defined(_SATURATION_MONITOR)
+ bSaturated = EAS_TRUE;
+#endif
+ return _EAS_MAX_OUTPUT;
+ }
+ if (nValue < _EAS_MIN_OUTPUT)
+ {
+#if defined(_SATURATION_MONITOR)
+ bSaturated = EAS_TRUE;
+#endif
+ return _EAS_MIN_OUTPUT;
+ }
+ return (EAS_I16) nValue;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_Noise()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * A 31-bit low-cost linear congruential PRNG algorithm used to
+ * generate noise.
+ *
+ * Inputs:
+ * pnSeed - pointer to 32-bit PRNG seed
+ *
+ * Outputs:
+ * Returns a 16-bit integer
+ *----------------------------------------------------------------------------
+*/
+EAS_INLINE EAS_I16 FM_Noise (EAS_U32 *pnSeed)
+{
+ *pnSeed = *pnSeed * 214013L + 2531011L;
+ return (EAS_I16) ((*pnSeed >> 15) & 0xffff);
+}
+
+/*----------------------------------------------------------------------------
+ * FM_PhaseInc()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Transform pitch cents to linear phase increment
+ *
+ * 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:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_I32 FM_PhaseInc (EAS_I32 nCents)
+{
+ EAS_I32 nDents;
+ EAS_I32 nExponentInt, nExponentFrac;
+ EAS_I32 nTemp1, nTemp2;
+ EAS_I32 nResult;
+
+ /* convert cents to dents */
+ nDents = FMUL_15x15(nCents, CENTS_TO_DENTS);
+ nExponentInt = GET_DENTS_INT_PART(nDents) + (32 - SINE_TABLE_SIZE_IN_BITS - NUM_EG1_FRAC_BITS);
+ 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;
+}
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+/*----------------------------------------------------------------------------
+ * FM_CalculatePan()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Assign the left and right gain values corresponding to the given pan value.
+ *
+ * Inputs:
+ * psVoice - ptr to the voice we have assigned for this channel
+ * psArticulation - ptr to this voice's articulation
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * the given voice's m_nGainLeft and m_nGainRight are assigned
+ *----------------------------------------------------------------------------
+*/
+static void FM_CalculatePan (EAS_I16 pan, EAS_U16 *pGainLeft, EAS_U16 *pGainRight)
+{
+ EAS_I32 nTemp;
+ EAS_INT nNetAngle;
+
+ /*
+ 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
+ */
+
+ /*
+ 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
+ */
+ /*lint -e{703} <avoid multiply for performance reasons>*/
+ nNetAngle = ((EAS_I32) pan) << (NUM_EG1_FRAC_BITS -7);
+
+ /* calculate sin */
+ nTemp = EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle);
+ nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle);
+
+ if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN)
+ nTemp = SYNTH_FULL_SCALE_EG1_GAIN;
+ else if (nTemp < 0)
+ nTemp = 0;
+
+ *pGainRight = (EAS_U16) nTemp;
+
+ /* calculate cos */
+ nTemp = -EG1_ONE + FMUL_15x15(COEFF_PAN_G2, nNetAngle);
+ nTemp = COEFF_PAN_G0 + FMUL_15x15(nTemp, nNetAngle);
+
+ if (nTemp > SYNTH_FULL_SCALE_EG1_GAIN)
+ nTemp = SYNTH_FULL_SCALE_EG1_GAIN;
+ else if (nTemp < 0)
+ nTemp = 0;
+
+ *pGainLeft = (EAS_U16) nTemp;
+}
+#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */
+
+/*----------------------------------------------------------------------------
+ * FM_Operator()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesizes a buffer of samples based on passed parameters.
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to synthesize
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void FM_Operator (
+ S_FM_ENG_OPER *p,
+ EAS_I32 numSamplesToAdd,
+ EAS_PCM *pBuffer,
+ EAS_PCM *pModBuffer,
+ EAS_BOOL mix,
+ EAS_U16 gainTarget,
+ EAS_I16 pitch,
+ EAS_U8 feedback,
+ EAS_I16 *pLastOutput)
+{
+ EAS_I32 gain;
+ EAS_I32 gainInc;
+ EAS_U32 phase;
+ EAS_U32 phaseInc;
+ EAS_U32 phaseTemp;
+ EAS_I32 temp;
+ EAS_I32 temp2;
+
+ /* establish local gain variable */
+ gain = (EAS_I32) p->gain << 16;
+
+ /* calculate gain increment */
+ /*lint -e{703} use shift for performance */
+ gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
+
+ /* establish local phase variables */
+ phase = p->phase;
+
+ /* calculate the new phase increment */
+ phaseInc = (EAS_U32) FM_PhaseInc(pitch);
+
+ /* restore final output from previous frame for feedback loop */
+ if (pLastOutput)
+ temp = *pLastOutput;
+ else
+ temp = 0;
+
+ /* generate a buffer of samples */
+ while (numSamplesToAdd--)
+ {
+
+ /* incorporate modulation */
+ if (pModBuffer)
+ {
+ /*lint -e{701} use shift for performance */
+ temp = *pModBuffer++ << FM_MODULATOR_INPUT_SHIFT;
+ }
+
+ /* incorporate feedback */
+ else
+ {
+ /*lint -e{703} use shift for performance */
+ temp = (temp * (EAS_I32) feedback) << FM_FEEDBACK_SHIFT;
+ }
+
+ /*lint -e{737} <use this behavior to avoid extra mask step> */
+ phaseTemp = phase + temp;
+
+ /* fetch sample from wavetable */
+ temp = sineTable[phaseTemp >> (32 - SINE_TABLE_SIZE_IN_BITS)];
+
+ /* increment operator phase */
+ phase += phaseInc;
+
+ /* internal gain for modulation effects */
+ temp = FMUL_15x15(temp, (gain >> 16));
+
+ /* output gain calculation */
+ temp2 = FMUL_15x15(temp, p->outputGain);
+
+ /* saturating add to buffer */
+ if (mix)
+ {
+ temp2 += *pBuffer;
+ *pBuffer++ = FM_Saturate(temp2);
+ }
+
+ /* output to buffer */
+ else
+ *pBuffer++ = (EAS_I16) temp2;
+
+ /* increment gain */
+ gain += gainInc;
+
+ }
+
+ /* save phase and gain */
+ p->phase = phase;
+ p->gain = gainTarget;
+
+ /* save last output for feedback in next frame */
+ if (pLastOutput)
+ *pLastOutput = (EAS_I16) temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_NoiseOperator()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesizes a buffer of samples based on passed parameters.
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to synthesize
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void FM_NoiseOperator (
+ S_FM_ENG_OPER *p,
+ EAS_I32 numSamplesToAdd,
+ EAS_PCM *pBuffer,
+ EAS_BOOL mix,
+ EAS_U16 gainTarget,
+ EAS_U8 feedback,
+ EAS_I16 *pLastOutput)
+{
+ EAS_I32 gain;
+ EAS_I32 gainInc;
+ EAS_U32 phase;
+ EAS_I32 temp;
+ EAS_I32 temp2;
+
+ /* establish local gain variable */
+ gain = (EAS_I32) p->gain << 16;
+
+ /* calculate gain increment */
+ /*lint -e{703} use shift for performance */
+ gainInc = ((EAS_I32) gainTarget - (EAS_I32) p->gain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
+
+ /* establish local phase variables */
+ phase = p->phase;
+
+ /* establish local phase variables */
+ phase = p->phase;
+
+ /* recall last sample for filter Z-1 term */
+ temp = 0;
+ if (pLastOutput)
+ temp = *pLastOutput;
+
+ /* generate a buffer of samples */
+ while (numSamplesToAdd--)
+ {
+
+ /* if using filter */
+ if (pLastOutput)
+ {
+ /* use PRNG for noise */
+ temp2 = FM_Noise(&phase);
+
+ /*lint -e{704} use shift for performance */
+ temp += ((temp2 -temp) * feedback) >> 8;
+ }
+ else
+ {
+ temp = FM_Noise(&phase);
+ }
+
+ /* internal gain for modulation effects */
+ temp2 = FMUL_15x15(temp, (gain >> 16));
+
+ /* output gain calculation */
+ temp2 = FMUL_15x15(temp2, p->outputGain);
+
+ /* saturating add to buffer */
+ if (mix)
+ {
+ temp2 += *pBuffer;
+ *pBuffer++ = FM_Saturate(temp2);
+ }
+
+ /* output to buffer */
+ else
+ *pBuffer++ = (EAS_I16) temp2;
+
+ /* increment gain */
+ gain += gainInc;
+
+ }
+
+ /* save phase and gain */
+ p->phase = phase;
+ p->gain = gainTarget;
+
+ /* save last output for feedback in next frame */
+ if (pLastOutput)
+ *pLastOutput = (EAS_I16) temp;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_ConfigVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Receives parameters to start a new voice.
+ *
+ * Inputs:
+ * voiceNum - voice number to start
+ * vCfg - configuration data
+ * pMixBuffer - pointer to host supplied buffer
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * pFrameBuffer is not used in the test version, but is passed as a
+ * courtesy to split architecture implementations. It can be used as
+ * as pointer to the interprocessor communications buffer when the
+ * synthesis parameters are passed off to a DSP for synthesis.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pFrameBuffer) pFrameBuffer not used in test version - see above */
+void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer)
+{
+ S_FM_ENG_VOICE *pVoice;
+ EAS_INT i;
+
+ /* establish pointer to voice data */
+ pVoice = &voices[voiceNum];
+
+ /* save data */
+ pVoice->feedback = vCfg->feedback;
+ pVoice->flags = vCfg->flags;
+ pVoice->voiceGain = vCfg->voiceGain;
+
+ /* initialize Z-1 terms */
+ pVoice->op1Out = 0;
+ pVoice->op3Out = 0;
+
+ /* initialize operators */
+ for (i = 0; i < 4; i++)
+ {
+ /* save operator data */
+ pVoice->oper[i].gain = vCfg->gain[i];
+ pVoice->oper[i].outputGain = vCfg->outputGain[i];
+ pVoice->oper[i].outputGain = vCfg->outputGain[i];
+
+ /* initalize operator */
+ pVoice->oper[i].phase = 0;
+ }
+
+ /* calculate pan */
+#if NUM_OUTPUT_CHANNELS == 2
+ FM_CalculatePan(vCfg->pan, &pVoice->gainLeft, &pVoice->gainRight);
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * FM_ProcessVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesizes a buffer of samples based on calculated parameters.
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to synthesize
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ * Notes:
+ * pOut is not used in the test version, but is passed as a
+ * courtesy to split architecture implementations. It can be used as
+ * as pointer to the interprocessor communications buffer when the
+ * synthesis parameters are passed off to a DSP for synthesis.
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pOut) pOut not used in test version - see above */
+void FM_ProcessVoice (
+ EAS_I32 voiceNum,
+ S_FM_VOICE_FRAME *pFrame,
+ EAS_I32 numSamplesToAdd,
+ EAS_PCM *pTempBuffer,
+ EAS_PCM *pBuffer,
+ EAS_I32 *pMixBuffer,
+ EAS_FRAME_BUFFER_HANDLE pFrameBuffer)
+{
+ S_FM_ENG_VOICE *p;
+ EAS_PCM *pOutBuf;
+ EAS_PCM *pMod;
+ EAS_BOOL mix;
+ EAS_U8 feedback1;
+ EAS_U8 feedback3;
+ EAS_U8 mode;
+
+ /* establish pointer to voice data */
+ p = &voices[voiceNum];
+ mode = p->flags & 0x07;
+
+ /* lookup feedback values */
+ feedback1 = fmScaleTable[p->feedback >> 4];
+ feedback3 = fmScaleTable[p->feedback & 0x0f];
+
+ /* operator 3 is on output bus in modes 0, 1, and 3 */
+ if ((mode == 0) || (mode == 1) || (mode == 3))
+ pOutBuf = pBuffer;
+ else
+ pOutBuf = pTempBuffer;
+
+ if (p->flags & FLAG_FM_ENG_VOICE_OP3_NOISE)
+ {
+ FM_NoiseOperator(
+ p->oper + 2,
+ numSamplesToAdd,
+ pOutBuf,
+ EAS_FALSE,
+ pFrame->gain[2],
+ feedback3,
+ &p->op3Out);
+ }
+ else
+ {
+ FM_Operator(
+ p->oper + 2,
+ numSamplesToAdd,
+ pOutBuf,
+ 0,
+ EAS_FALSE,
+ pFrame->gain[2],
+ pFrame->pitch[2],
+ feedback3,
+ &p->op3Out);
+ }
+
+ /* operator 4 is on output bus in modes 0, 1, and 2 */
+ if (mode < 3)
+ pOutBuf = pBuffer;
+ else
+ pOutBuf = pTempBuffer;
+
+ /* operator 4 is modulated in modes 2, 4, and 5 */
+ if ((mode == 2) || (mode == 4) || (mode == 5))
+ pMod = pTempBuffer;
+ else
+ pMod = 0;
+
+ /* operator 4 is in mix mode in modes 0 and 1 */
+ mix = (mode < 2);
+
+ if (p->flags & FLAG_FM_ENG_VOICE_OP4_NOISE)
+ {
+ FM_NoiseOperator(
+ p->oper + 3,
+ numSamplesToAdd,
+ pOutBuf,
+ mix,
+ pFrame->gain[3],
+ 0,
+ 0);
+ }
+ else
+ {
+ FM_Operator(
+ p->oper + 3,
+ numSamplesToAdd,
+ pOutBuf,
+ pMod,
+ mix,
+ pFrame->gain[3],
+ pFrame->pitch[3],
+ 0,
+ 0);
+ }
+
+ /* operator 1 is on output bus in mode 0 */
+ if (mode == 0)
+ pOutBuf = pBuffer;
+ else
+ pOutBuf = pTempBuffer;
+
+ /* operator 1 is modulated in modes 3 and 4 */
+ if ((mode == 3) || (mode == 4))
+ pMod = pTempBuffer;
+ else
+ pMod = 0;
+
+ /* operator 1 is in mix mode in modes 0 and 5 */
+ mix = ((mode == 0) || (mode == 5));
+
+ if (p->flags & FLAG_FM_ENG_VOICE_OP1_NOISE)
+ {
+ FM_NoiseOperator(
+ p->oper,
+ numSamplesToAdd,
+ pOutBuf,
+ mix,
+ pFrame->gain[0],
+ feedback1,
+ &p->op1Out);
+ }
+ else
+ {
+ FM_Operator(
+ p->oper,
+ numSamplesToAdd,
+ pOutBuf,
+ pMod,
+ mix,
+ pFrame->gain[0],
+ pFrame->pitch[0],
+ feedback1,
+ &p->op1Out);
+ }
+
+ /* operator 2 is modulated in all modes except 0 */
+ if (mode != 0)
+ pMod = pTempBuffer;
+ else
+ pMod = 0;
+
+ /* operator 1 is in mix mode in modes 0 -3 */
+ mix = (mode < 4);
+
+ if (p->flags & FLAG_FM_ENG_VOICE_OP2_NOISE)
+ {
+ FM_NoiseOperator(
+ p->oper + 1,
+ numSamplesToAdd,
+ pBuffer,
+ mix,
+ pFrame->gain[1],
+ 0,
+ 0);
+ }
+ else
+ {
+ FM_Operator(
+ p->oper + 1,
+ numSamplesToAdd,
+ pBuffer,
+ pMod,
+ mix,
+ pFrame->gain[1],
+ pFrame->pitch[1],
+ 0,
+ 0);
+ }
+
+ /* mix voice output to synthesizer output buffer */
+ FM_SynthMixVoice(p, pFrame->voiceGain, numSamplesToAdd, pBuffer, pMixBuffer);
+}
+
+/*----------------------------------------------------------------------------
+ * FM_SynthMixVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Mixes the voice output buffer into the final mix using an anti-zipper
+ * filter.
+ *
+ * Inputs:
+ * nNumSamplesToAdd - number of samples to synthesize
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+void FM_SynthMixVoice(S_FM_ENG_VOICE *p, EAS_U16 nGainTarget, EAS_I32 numSamplesToAdd, EAS_PCM *pInputBuffer, EAS_I32 *pBuffer)
+{
+ EAS_I32 nGain;
+ EAS_I32 nGainInc;
+ EAS_I32 nTemp;
+
+ /* restore previous gain */
+ /*lint -e{703} <use shift for performance> */
+ nGain = (EAS_I32) p->voiceGain << 16;
+
+ /* calculate gain increment */
+ /*lint -e{703} <use shift for performance> */
+ nGainInc = ((EAS_I32) nGainTarget - (EAS_I32) p->voiceGain) << (16 - SYNTH_UPDATE_PERIOD_IN_BITS);
+
+ /* mix the output buffer */
+ while (numSamplesToAdd--)
+ {
+ /* output gain calculation */
+ nTemp = *pInputBuffer++;
+
+ /* sum to output buffer */
+#if (NUM_OUTPUT_CHANNELS == 2)
+
+ /*lint -e{704} <use shift for performance> */
+ nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_GAIN_SHIFT;
+
+ /*lint -e{704} <use shift for performance> */
+ {
+ EAS_I32 nTemp2;
+ nTemp = nTemp >> FM_STEREO_PRE_GAIN_SHIFT;
+ nTemp2 = (nTemp * p->gainLeft) >> FM_STEREO_POST_GAIN_SHIFT;
+ *pBuffer++ += nTemp2;
+ nTemp2 = (nTemp * p->gainRight) >> FM_STEREO_POST_GAIN_SHIFT;
+ *pBuffer++ += nTemp2;
+ }
+#else
+ /*lint -e{704} <use shift for performance> */
+ nTemp = ((EAS_I32) nTemp * (nGain >> 16)) >> FM_MONO_GAIN_SHIFT;
+ *pBuffer++ += nTemp;
+#endif
+
+ /* increment gain for anti-zipper filter */
+ nGain += nGainInc;
+ }
+
+ /* save gain */
+ p->voiceGain = nGainTarget;
+}
+
diff --git a/arm-hybrid-22k/lib_src/eas_fmengine.h b/arm-hybrid-22k/lib_src/eas_fmengine.h new file mode 100644 index 0000000..4ddc12b --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_fmengine.h @@ -0,0 +1,121 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_fmengine.h
+ *
+ * Contents and purpose:
+ * Declarations, interfaces, and prototypes for FM synthesize low-level.
+ *
+ * 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: 664 $
+ * $Date: 2007-04-25 13:11:22 -0700 (Wed, 25 Apr 2007) $
+ *----------------------------------------------------------------------------
+*/
+
+/* sentinel */
+#ifndef _FMENGINE_H
+#define _FMENGINE_H
+
+/* check for split architecture */
+#if defined (EAS_SPLIT_HYBRID_SYNTH) || defined(EAS_SPLIT_FM_SYNTH)
+#define FM_OFFBOARD
+#endif
+
+/* output level to mix buffer (3 = -24dB) */
+#define FM_GAIN_SHIFT 3
+#define FM_MONO_GAIN_SHIFT 9
+
+/* voice output level for stereo 15 = +6dB */
+#define FM_STEREO_PRE_GAIN_SHIFT 11
+#define FM_STEREO_POST_GAIN_SHIFT 10
+
+/* modulator input level shift (21 = -30dB) */
+#define FM_MODULATOR_INPUT_SHIFT 21
+
+/* feedback control level shift (7 = 0dB) */
+#define FM_FEEDBACK_SHIFT 7
+
+/* synth final output level */
+#define SYNTH_POST_GAIN_SHIFT 14
+
+/* LFO modulation to gain control */
+#define FM_LFO_GAIN_SHIFT 12
+
+/* sine table is always a power of 2 - saves cycles in inner loop */
+#define SINE_TABLE_SIZE_IN_BITS 11
+#define SINE_TABLE_SIZE 2048
+
+/* operator structure for FM engine */
+typedef struct
+{
+ EAS_U32 phase; /* current waveform phase */
+ EAS_U16 gain; /* current internal gain */
+ EAS_U16 outputGain; /* current output gain */
+} S_FM_ENG_OPER;
+
+typedef struct
+{
+ S_FM_ENG_OPER oper[4]; /* operator data */
+ EAS_I16 op1Out; /* op1 output for feedback loop */
+ EAS_I16 op3Out; /* op3 output for feedback loop */
+ EAS_U16 voiceGain; /* LFO + channel parameters */
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_U16 gainLeft; /* left gain multiplier */
+ EAS_U16 gainRight; /* right gain multiplier */
+#endif
+ EAS_U8 flags; /* mode bits and noise waveform flags */
+ EAS_U8 feedback; /* feedback for Op1 and Op3 */
+} S_FM_ENG_VOICE;
+
+typedef struct
+{
+ EAS_U16 gain[4]; /* initial operator gain value */
+ EAS_U16 outputGain[4]; /* initial operator output gain value */
+ EAS_U16 voiceGain; /* initial voice gain */
+ EAS_U8 flags; /* mode bits and noise waveform flags */
+ EAS_U8 feedback; /* feedback for Op1 and Op3 */
+#if (NUM_OUTPUT_CHANNELS == 2)
+ EAS_I8 pan; /* pan value +/-64 */
+#endif
+} S_FM_VOICE_CONFIG;
+
+typedef struct
+{
+ EAS_U16 gain[4]; /* new operator gain value */
+ EAS_I16 pitch[4]; /* new pitch value */
+ EAS_U16 voiceGain; /* new voice gain */
+} S_FM_VOICE_FRAME;
+
+/* bit definitions for S_FM_ENG_VOICE.flags */
+#define FLAG_FM_ENG_VOICE_OP1_NOISE 0x10 /* operator 1 source is PRNG */
+#define FLAG_FM_ENG_VOICE_OP2_NOISE 0x20 /* operator 2 source is PRNG */
+#define FLAG_FM_ENG_VOICE_OP3_NOISE 0x40 /* operator 3 source is PRNG */
+#define FLAG_FM_ENG_VOICE_OP4_NOISE 0x80 /* operator 4 source is PRNG */
+
+#ifdef FM_OFFBOARD
+extern EAS_BOOL FM_StartFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+extern EAS_BOOL FM_EndFrame (EAS_FRAME_BUFFER_HANDLE pFrameBuffe, EAS_I32 *pMixBuffer, EAS_I16 masterGain);
+#endif
+
+/* FM engine prototypes */
+extern void FM_ConfigVoice (EAS_I32 voiceNum, S_FM_VOICE_CONFIG *vCfg, EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+extern void FM_ProcessVoice (EAS_I32 voiceNum, S_FM_VOICE_FRAME *pFrame, EAS_I32 numSamplesToAdd, EAS_PCM *pTempBuffer, EAS_PCM *pBuffer, EAS_I32 *pMixBuffer, EAS_FRAME_BUFFER_HANDLE pFrameBuffer);
+
+#endif
+/* #ifndef _FMENGINE_H */
+
diff --git a/arm-hybrid-22k/lib_src/eas_fmsynth.c b/arm-hybrid-22k/lib_src/eas_fmsynth.c new file mode 100644 index 0000000..83f0087 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_fmsynth.c @@ -0,0 +1,910 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * fmsynth.c
+ *
+ * Contents and purpose:
+ * Implements the high-level FM 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_host.h"
+#include "eas_report.h"
+
+#include "eas_data.h"
+#include "eas_synth_protos.h"
+#include "eas_audioconst.h"
+#include "eas_fmengine.h"
+#include "eas_math.h"
+
+/* option sanity check */
+#ifdef _REVERB
+#error "No reverb for FM synthesizer"
+#endif
+#ifdef _CHORUS
+#error "No chorus for FM synthesizer"
+#endif
+
+/*
+ * WARNING: These macros can cause unwanted side effects. Use them
+ * with care. For example, min(x++,y++) will cause either x or y to be
+ * incremented twice.
+ */
+#define min(a,b) ((a) < (b) ? (a) : (b))
+#define max(a,b) ((a) > (b) ? (a) : (b))
+
+/* pivot point for keyboard scalars */
+#define EG_SCALE_PIVOT_POINT 64
+#define KEY_SCALE_PIVOT_POINT 36
+
+/* This number is the negative of the frequency of the note (in cents) of
+ * the sine wave played at unity. The number can be calculated as follows:
+ *
+ * MAGIC_NUMBER = 1200 * log(base2) (SINE_TABLE_SIZE * 8.175798916 / SAMPLE_RATE)
+ *
+ * 8.17578 is a reference to the frequency of MIDI note 0
+ */
+#if defined (_SAMPLE_RATE_8000)
+#define MAGIC_NUMBER 1279
+#elif defined (_SAMPLE_RATE_16000)
+#define MAGIC_NUMBER 79
+#elif defined (_SAMPLE_RATE_20000)
+#define MAGIC_NUMBER -308
+#elif defined (_SAMPLE_RATE_22050)
+#define MAGIC_NUMBER -477
+#elif defined (_SAMPLE_RATE_24000)
+#define MAGIC_NUMBER -623
+#elif defined (_SAMPLE_RATE_32000)
+#define MAGIC_NUMBER -1121
+#elif defined (_SAMPLE_RATE_44100)
+#define MAGIC_NUMBER -1677
+#elif defined (_SAMPLE_RATE_48000)
+#define MAGIC_NUMBER -1823
+#endif
+
+/* externs */
+extern const EAS_I16 fmControlTable[128];
+extern const EAS_U16 fmRateTable[256];
+extern const EAS_U16 fmAttackTable[16];
+extern const EAS_U8 fmDecayTable[16];
+extern const EAS_U8 fmReleaseTable[16];
+extern const EAS_U8 fmScaleTable[16];
+
+/* local prototypes */
+/*lint -esym(715, pVoiceMgr) standard synthesizer interface - pVoiceMgr not used */
+static EAS_RESULT FM_Initialize (S_VOICE_MGR *pVoiceMgr) { return EAS_SUCCESS; }
+static EAS_RESULT FM_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex);
+static EAS_BOOL FM_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples);
+static void FM_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+static void FM_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum);
+static void FM_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum);
+static void FM_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel);
+
+
+/*----------------------------------------------------------------------------
+ * Synthesizer interface
+ *----------------------------------------------------------------------------
+*/
+const S_SYNTH_INTERFACE fmSynth =
+{
+ FM_Initialize,
+ FM_StartVoice,
+ FM_UpdateVoice,
+ FM_ReleaseVoice,
+ FM_MuteVoice,
+ FM_SustainPedal,
+ FM_UpdateChannel
+};
+
+#ifdef FM_OFFBOARD
+const S_FRAME_INTERFACE fmFrameInterface =
+{
+ FM_StartFrame,
+ FM_EndFrame
+};
+#endif
+
+/*----------------------------------------------------------------------------
+ * inline functions
+ *----------------------------------------------------------------------------
+ */
+EAS_INLINE S_FM_VOICE *GetFMVoicePtr (S_VOICE_MGR *pVoiceMgr, EAS_INT voiceNum)
+{
+ return &pVoiceMgr->fmVoices[voiceNum];
+}
+EAS_INLINE S_SYNTH_CHANNEL *GetChannelPtr (S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice)
+{
+ return &pSynth->channels[pVoice->channel & 15];
+}
+EAS_INLINE const S_FM_REGION *GetFMRegionPtr (S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice)
+{
+#ifdef _SECONDARY_SYNTH
+ return &pSynth->pEAS->pFMRegions[pVoice->regionIndex & REGION_INDEX_MASK];
+#else
+ return &pSynth->pEAS->pFMRegions[pVoice->regionIndex];
+#endif
+}
+
+/*----------------------------------------------------------------------------
+ * FM_SynthIsOutputOperator
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Returns true if the operator is a direct output and not muted
+ *
+ * Inputs:
+ *
+ * Outputs:
+ * Returns boolean
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL FM_SynthIsOutputOperator (const S_FM_REGION *pRegion, EAS_INT operIndex)
+{
+
+ /* see if voice is muted */
+ if ((pRegion->oper[operIndex].gain & 0xfc) == 0)
+ return 0;
+
+ /* check based on mode */
+ switch (pRegion->region.keyGroupAndFlags & 7)
+ {
+
+ /* mode 0 - all operators are external */
+ case 0:
+ return EAS_TRUE;
+
+ /* mode 1 - operators 1-3 are external */
+ case 1:
+ if (operIndex != 0)
+ return EAS_TRUE;
+ break;
+
+ /* mode 2 - operators 1 & 3 are external */
+ case 2:
+ if ((operIndex == 1) || (operIndex == 3))
+ return EAS_TRUE;
+ break;
+
+ /* mode 2 - operators 1 & 2 are external */
+ case 3:
+ if ((operIndex == 1) || (operIndex == 2))
+ return EAS_TRUE;
+ break;
+
+ /* modes 4 & 5 - operator 1 is external */
+ case 4:
+ case 5:
+ if (operIndex == 1)
+ return EAS_TRUE;
+ break;
+
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL,"Invalid voice mode: %d",
+ pRegion->region.keyGroupAndFlags & 7); */ }
+ }
+
+ return EAS_FALSE;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_CalcEGRate()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ *
+ * Inputs:
+ * nKeyNumber - MIDI note
+ * nLogRate - logarithmic scale rate from patch data
+ * nKeyScale - key scaling factor for this EG
+ *
+ * Outputs:
+ * 16-bit linear multiplier
+ *----------------------------------------------------------------------------
+*/
+
+static EAS_U16 FM_CalcEGRate (EAS_U8 nKeyNumber, EAS_U8 nLogRate, EAS_U8 nEGScale)
+{
+ EAS_I32 temp;
+
+ /* incorporate key scaling on release rate */
+ temp = (EAS_I32) nLogRate << 7;
+ temp += ((EAS_I32) nKeyNumber - EG_SCALE_PIVOT_POINT) * (EAS_I32) nEGScale;
+
+ /* saturate */
+ temp = max(temp, 0);
+ temp = min(temp, 32767);
+
+ /* look up in rate table */
+ /*lint -e{704} use shift for performance */
+ return fmRateTable[temp >> 8];
+}
+
+/*----------------------------------------------------------------------------
+ * FM_ReleaseVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice is being released.
+ *
+ * Inputs:
+ * psEASData - pointer to S_EAS_DATA
+ * pVoice - pointer to voice to release
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+static void FM_ReleaseVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
+{
+ EAS_INT operIndex;
+ const S_FM_REGION *pRegion;
+ S_FM_VOICE *pFMVoice;
+
+ /* check to see if voice responds to NOTE-OFF's */
+ pRegion = GetFMRegionPtr(pSynth, pVoice);
+ if (pRegion->region.keyGroupAndFlags & REGION_FLAG_ONE_SHOT)
+ return;
+
+ /* set all envelopes to release state */
+ pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum);
+ for (operIndex = 0; operIndex < 4; operIndex++)
+ {
+ pFMVoice->oper[operIndex].envState = eFMEnvelopeStateRelease;
+
+ /* incorporate key scaling on release rate */
+ pFMVoice->oper[operIndex].envRate = FM_CalcEGRate(
+ pVoice->note,
+ fmReleaseTable[pRegion->oper[operIndex].velocityRelease & 0x0f],
+ fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]);
+ } /* end for (operIndex = 0; operIndex < 4; operIndex++) */
+}
+
+/*----------------------------------------------------------------------------
+ * FM_MuteVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice is being muted.
+ *
+ * Inputs:
+ * pVoice - pointer to voice to release
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pSynth) standard interface, pVoiceMgr not used */
+static void FM_MuteVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum)
+{
+ S_FM_VOICE *pFMVoice;
+
+ /* clear deferred action flags */
+ pVoice->voiceFlags &=
+ ~(VOICE_FLAG_DEFER_MIDI_NOTE_OFF |
+ VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF |
+ VOICE_FLAG_DEFER_MUTE);
+
+ /* set all envelopes to muted state */
+ pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum);
+ pFMVoice->oper[0].envState = eFMEnvelopeStateMuted;
+ pFMVoice->oper[1].envState = eFMEnvelopeStateMuted;
+ pFMVoice->oper[2].envState = eFMEnvelopeStateMuted;
+ pFMVoice->oper[3].envState = eFMEnvelopeStateMuted;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_SustainPedal()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * The selected voice is held due to sustain pedal
+ *
+ * Inputs:
+ * pVoice - pointer to voice to sustain
+ *
+ * Outputs:
+ * None
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pChannel) standard interface, pVoiceMgr not used */
+static void FM_SustainPedal (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, S_SYNTH_CHANNEL *pChannel, EAS_I32 voiceNum)
+{
+ const S_FM_REGION *pRegion;
+ S_FM_VOICE *pFMVoice;
+ EAS_INT operIndex;
+
+ pRegion = GetFMRegionPtr(pSynth, pVoice);
+ pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum);
+
+ /* check to see if any envelopes are above the sustain level */
+ for (operIndex = 0; operIndex < 4; operIndex++)
+ {
+
+ /* if level control or envelope gain is zero, skip this envelope */
+ if (((pRegion->oper[operIndex].gain & 0xfc) == 0) ||
+ (pFMVoice->oper[operIndex].envGain == 0))
+ {
+ continue;
+ }
+
+ /* if the envelope gain is above the sustain level, we need to catch this voice */
+ if (pFMVoice->oper[operIndex].envGain >= ((EAS_U16) (pRegion->oper[operIndex].sustain & 0xfc) << 7))
+ {
+
+ /* reset envelope to decay state */
+ pFMVoice->oper[operIndex].envState = eFMEnvelopeStateDecay;
+
+ pFMVoice->oper[operIndex].envRate = FM_CalcEGRate(
+ pVoice->note,
+ fmDecayTable[pRegion->oper[operIndex].attackDecay & 0x0f],
+ fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]);
+
+ /* set voice to decay state */
+ pVoice->voiceState = eVoiceStatePlay;
+
+ /* set sustain flag */
+ pVoice->voiceFlags |= VOICE_FLAG_SUSTAIN_PEDAL_DEFER_NOTE_OFF;
+ }
+ } /* end for (operIndex = 0; operIndex < 4; operIndex++) */
+}
+
+/*----------------------------------------------------------------------------
+ * FM_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
+ * psEASData - 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 FM_StartVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_U16 regionIndex)
+{
+ S_FM_VOICE *pFMVoice;
+ S_SYNTH_CHANNEL *pChannel;
+ const S_FM_REGION *pRegion;
+ EAS_I32 temp;
+ EAS_INT operIndex;
+
+ /* establish pointers to data */
+ pVoice->regionIndex = regionIndex;
+ pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum);
+ pChannel = GetChannelPtr(pSynth, pVoice);
+ pRegion = GetFMRegionPtr(pSynth, pVoice);
+
+ /* update static channel parameters */
+ if (pChannel->channelFlags & CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS)
+ FM_UpdateChannel(pVoiceMgr, pSynth, pVoice->channel & 15);
+
+ /* init the LFO */
+ pFMVoice->lfoValue = 0;
+ pFMVoice->lfoPhase = 0;
+ pFMVoice->lfoDelay = (EAS_U16) (fmScaleTable[pRegion->lfoFreqDelay & 0x0f] >> 1);
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ /* calculate pan gain values only if stereo output */
+ /* set up panning only at note start */
+ temp = (EAS_I32) pChannel->pan - 64;
+ temp += (EAS_I32) pRegion->pan;
+ if (temp < -64)
+ temp = -64;
+ if (temp > 64)
+ temp = 64;
+ pFMVoice->pan = (EAS_I8) temp;
+#endif /* #if (NUM_OUTPUT_CHANNELS == 2) */
+
+ /* no samples have been synthesized for this note yet */
+ pVoice->voiceFlags = VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
+
+ /* initialize gain value for anti-zipper filter */
+ pFMVoice->voiceGain = (EAS_I16) EAS_LogToLinear16(pChannel->staticGain);
+ pFMVoice->voiceGain = (EAS_I16) FMUL_15x15(pFMVoice->voiceGain, pSynth->masterVolume);
+
+ /* initialize the operators */
+ for (operIndex = 0; operIndex < 4; operIndex++)
+ {
+
+ /* establish operator output gain level */
+ /*lint -e{701} <use shift for performance> */
+ pFMVoice->oper[operIndex].outputGain = EAS_LogToLinear16(((EAS_I16) (pRegion->oper[operIndex].gain & 0xfc) - 0xfc) << 7);
+
+ /* check for linear velocity flag */
+ /*lint -e{703} <use shift for performance> */
+ if (pRegion->oper[operIndex].flags & FM_OPER_FLAG_LINEAR_VELOCITY)
+ temp = (EAS_I32) (pVoice->velocity - 127) << 5;
+ else
+ temp = (EAS_I32) fmControlTable[pVoice->velocity];
+
+ /* scale velocity */
+ /*lint -e{704} use shift for performance */
+ temp = (temp * (EAS_I32)(pRegion->oper[operIndex].velocityRelease & 0xf0)) >> 7;
+
+ /* include key scalar */
+ temp -= ((EAS_I32) pVoice->note - KEY_SCALE_PIVOT_POINT) * (EAS_I32) fmScaleTable[pRegion->oper[operIndex].egKeyScale & 0x0f];
+
+ /* saturate */
+ temp = min(temp, 0);
+ temp = max(temp, -32768);
+
+ /* save static gain parameters */
+ pFMVoice->oper[operIndex].baseGain = (EAS_I16) EAS_LogToLinear16(temp);
+
+ /* incorporate key scaling on decay rate */
+ pFMVoice->oper[operIndex].envRate = FM_CalcEGRate(
+ pVoice->note,
+ fmDecayTable[pRegion->oper[operIndex].attackDecay & 0x0f],
+ fmScaleTable[pRegion->oper[operIndex].egKeyScale >> 4]);
+
+ /* if zero attack time, max out envelope and jump to decay state */
+ if ((pRegion->oper[operIndex].attackDecay & 0xf0) == 0xf0)
+ {
+
+ /* start out envelope at max */
+ pFMVoice->oper[operIndex].envGain = 0x7fff;
+
+ /* set envelope to decay state */
+ pFMVoice->oper[operIndex].envState = eFMEnvelopeStateDecay;
+ }
+
+ /* start envelope at zero and start in attack state */
+ else
+ {
+ pFMVoice->oper[operIndex].envGain = 0;
+ pFMVoice->oper[operIndex].envState = eFMEnvelopeStateAttack;
+ }
+ }
+
+ return EAS_SUCCESS;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_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.
+ * Called when CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS flag is set.
+ *
+ * Inputs:
+ * nChannel - channel to update
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - the given channel's static gain and static pitch are updated
+ *----------------------------------------------------------------------------
+*/
+/*lint -esym(715, pVoiceMgr) standard interface, pVoiceMgr not used */
+static void FM_UpdateChannel (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, EAS_U8 channel)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ EAS_I32 temp;
+
+ pChannel = &pSynth->channels[channel];
+
+ /* convert CC7 volume controller to log scale */
+ temp = fmControlTable[pChannel->volume];
+
+ /* incorporate CC11 expression controller */
+ temp += fmControlTable[pChannel->expression];
+
+ /* saturate */
+ pChannel->staticGain = (EAS_I16) max(temp,-32768);
+
+ /* calculate pitch bend */
+ /*lint -e{703} <avoid multiply for performance>*/
+ temp = (((EAS_I32)(pChannel->pitchBend) << 2) - 32768);
+
+ temp = FMUL_15x15(temp, pChannel->pitchBendSensitivity);
+
+ /* include "magic number" compensation for sample rate and lookup table size */
+ temp += MAGIC_NUMBER;
+
+ /* if this is not a drum channel, then add in the per-channel tuning */
+ if (!(pChannel->channelFlags & CHANNEL_FLAG_RHYTHM_CHANNEL))
+ temp += (pChannel->finePitch + (pChannel->coarsePitch * 100));
+
+ /* save static pitch */
+ pChannel->staticPitch = temp;
+
+ /* Calculate LFO modulation depth */
+ /* mod wheel to LFO depth */
+ temp = FMUL_15x15(DEFAULT_LFO_MOD_WHEEL_TO_PITCH_CENTS,
+ pChannel->modWheel << (NUM_EG1_FRAC_BITS -7));
+
+ /* channel pressure to LFO depth */
+ pChannel->lfoAmt = (EAS_I16) (temp +
+ FMUL_15x15(DEFAULT_LFO_CHANNEL_PRESSURE_TO_PITCH_CENTS,
+ pChannel->channelPressure << (NUM_EG1_FRAC_BITS -7)));
+
+ /* clear update flag */
+ pChannel->channelFlags &= ~CHANNEL_FLAG_UPDATE_CHANNEL_PARAMETERS;
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_UpdateLFO()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate the LFO for the given voice
+ *
+ * Inputs:
+ * pVoice - ptr to the voice whose LFO we want to update
+ * psEASData - pointer to overall EAS data structure - used for debug only
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ * - updates LFO values for the given voice
+ *----------------------------------------------------------------------------
+*/
+static void FM_UpdateLFO (S_FM_VOICE *pFMVoice, const S_FM_REGION *pRegion)
+{
+
+ /* increment the LFO phase if the delay time has elapsed */
+ if (!pFMVoice->lfoDelay)
+ {
+ /*lint -e{701} <use shift for performance> */
+ pFMVoice->lfoPhase = pFMVoice->lfoPhase + (EAS_U16) (-fmControlTable[((15 - (pRegion->lfoFreqDelay >> 4)) << 3) + 4]);
+
+ /* square wave LFO? */
+ if (pRegion->region.keyGroupAndFlags & REGION_FLAG_SQUARE_WAVE)
+ pFMVoice->lfoValue = (EAS_I16)(pFMVoice->lfoPhase & 0x8000 ? -32767 : 32767);
+
+ /* trick to get a triangle wave out of a sawtooth */
+ else
+ {
+ pFMVoice->lfoValue = (EAS_I16) (pFMVoice->lfoPhase << 1);
+ /*lint -e{502} <shortcut to turn sawtooth into sine wave> */
+ if ((pFMVoice->lfoPhase > 0x3fff) && (pFMVoice->lfoPhase < 0xC000))
+ pFMVoice->lfoValue = ~pFMVoice->lfoValue;
+ }
+ }
+
+ /* still in delay */
+ else
+ pFMVoice->lfoDelay--;
+
+ return;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_UpdateEG()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate the synthesis parameters for an operator to be used during
+ * the next buffer
+ *
+ * Inputs:
+ * pVoice - pointer to the voice being updated
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL FM_UpdateEG (S_SYNTH_VOICE *pVoice, S_OPERATOR *pOper, const S_FM_OPER *pOperData, EAS_BOOL oneShot)
+{
+ EAS_U32 temp;
+ EAS_BOOL done;
+
+ /* set flag assuming the envelope is not done */
+ done = EAS_FALSE;
+
+ /* take appropriate action based on state */
+ switch (pOper->envState)
+ {
+
+ case eFMEnvelopeStateAttack:
+
+ /* the envelope is linear during the attack, so add the value */
+ temp = pOper->envGain + fmAttackTable[pOperData->attackDecay >> 4];
+
+ /* check for end of attack */
+ if (temp >= 0x7fff)
+ {
+ pOper->envGain = 0x7fff;
+ pOper->envState = eFMEnvelopeStateDecay;
+ }
+ else
+ pOper->envGain = (EAS_U16) temp;
+ break;
+
+ case eFMEnvelopeStateDecay:
+
+ /* decay is exponential, multiply by decay rate */
+ pOper->envGain = (EAS_U16) FMUL_15x15(pOper->envGain, pOper->envRate);
+
+ /* check for sustain level reached */
+ temp = (EAS_U32) (pOperData->sustain & 0xfc) << 7;
+ if (pOper->envGain <= (EAS_U16) temp)
+ {
+ /* if this is a one-shot patch, go directly to release phase */
+ if (oneShot)
+ {
+ pOper->envRate = FM_CalcEGRate(
+ pVoice->note,
+ fmReleaseTable[pOperData->velocityRelease & 0x0f],
+ fmScaleTable[pOperData->egKeyScale >> 4]);
+ pOper->envState = eFMEnvelopeStateRelease;
+ }
+
+ /* normal sustaining type */
+ else
+ {
+ pOper->envGain = (EAS_U16) temp;
+ pOper->envState = eFMEnvelopeStateSustain;
+ }
+ }
+ break;
+
+ case eFMEnvelopeStateSustain:
+ pOper->envGain = (EAS_U16)((EAS_U16)(pOperData->sustain & 0xfc) << 7);
+ break;
+
+ case eFMEnvelopeStateRelease:
+
+ /* release is exponential, multiply by release rate */
+ pOper->envGain = (EAS_U16) FMUL_15x15(pOper->envGain, pOper->envRate);
+
+ /* fully released */
+ if (pOper->envGain == 0)
+ {
+ pOper->envGain = 0;
+ pOper->envState = eFMEnvelopeStateMuted;
+ done = EAS_TRUE;
+ }
+ break;
+
+ case eFMEnvelopeStateMuted:
+ pOper->envGain = 0;
+ done = EAS_TRUE;
+ break;
+ default:
+ { /* dpp: EAS_ReportEx(_EAS_SEVERITY_FATAL,"Invalid operator state: %d", pOper->envState); */ }
+ } /* end switch (pOper->m_eState) */
+
+ return done;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_UpdateDynamic()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Calculate the synthesis parameters for this voice that will be used to
+ * synthesize the next buffer
+ *
+ * Inputs:
+ * pVoice - pointer to the voice being updated
+ * psEASData - pointer to overall EAS data structure
+ *
+ * Outputs:
+ * Returns EAS_TRUE if voice will be fully ramped to zero at the end of
+ * the next synthesized buffer.
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL FM_UpdateDynamic (S_SYNTH_VOICE *pVoice, S_FM_VOICE *pFMVoice, const S_FM_REGION *pRegion, S_SYNTH_CHANNEL *pChannel)
+{
+ EAS_I32 temp;
+ EAS_I32 pitch;
+ EAS_I32 lfoPitch;
+ EAS_INT operIndex;
+ EAS_BOOL done;
+
+ /* increment LFO phase */
+ FM_UpdateLFO(pFMVoice, pRegion);
+
+ /* base pitch in cents */
+ pitch = pVoice->note * 100;
+
+ /* LFO amount includes LFO depth from programming + channel dynamics */
+ temp = (fmScaleTable[pRegion->vibTrem >> 4] >> 1) + pChannel->lfoAmt;
+
+ /* multiply by LFO output to get final pitch modulation */
+ lfoPitch = FMUL_15x15(pFMVoice->lfoValue, temp);
+
+ /* flag to indicate this voice is done */
+ done = EAS_TRUE;
+
+ /* iterate through operators to establish parameters */
+ for (operIndex = 0; operIndex < 4; operIndex++)
+ {
+ EAS_BOOL bTemp;
+
+ /* set base phase increment for each operator */
+ temp = pRegion->oper[operIndex].tuning +
+ pChannel->staticPitch;
+
+ /* add vibrato effect unless it is disabled for this operator */
+ if ((pRegion->oper[operIndex].flags & FM_OPER_FLAG_NO_VIBRATO) == 0)
+ temp += lfoPitch;
+
+ /* if note is monotonic, bias to MIDI note 60 */
+ if (pRegion->oper[operIndex].flags & FM_OPER_FLAG_MONOTONE)
+ temp += 6000;
+ else
+ temp += pitch;
+ pFMVoice->oper[operIndex].pitch = (EAS_I16) temp;
+
+ /* calculate envelope, returns true if done */
+ bTemp = FM_UpdateEG(pVoice, &pFMVoice->oper[operIndex], &pRegion->oper[operIndex], pRegion->region.keyGroupAndFlags & REGION_FLAG_ONE_SHOT);
+
+ /* check if all output envelopes have completed */
+ if (FM_SynthIsOutputOperator(pRegion, operIndex))
+ done = done && bTemp;
+ }
+
+ return done;
+}
+
+/*----------------------------------------------------------------------------
+ * FM_UpdateVoice()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Synthesize a block of samples for the given voice.
+ *
+ * Inputs:
+ * 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
+ *
+ *----------------------------------------------------------------------------
+*/
+static EAS_BOOL FM_UpdateVoice (S_VOICE_MGR *pVoiceMgr, S_SYNTH *pSynth, S_SYNTH_VOICE *pVoice, EAS_I32 voiceNum, EAS_I32 *pMixBuffer, EAS_I32 numSamples)
+{
+ S_SYNTH_CHANNEL *pChannel;
+ const S_FM_REGION *pRegion;
+ S_FM_VOICE *pFMVoice;
+ S_FM_VOICE_CONFIG vCfg;
+ S_FM_VOICE_FRAME vFrame;
+ EAS_I32 temp;
+ EAS_INT oper;
+ EAS_U16 voiceGainTarget;
+ EAS_BOOL done;
+
+ /* setup some pointers */
+ pChannel = GetChannelPtr(pSynth, pVoice);
+ pRegion = GetFMRegionPtr(pSynth, pVoice);
+ pFMVoice = GetFMVoicePtr(pVoiceMgr, voiceNum);
+
+ /* if the voice is just starting, get the voice configuration data */
+ if (pVoice->voiceFlags & VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET)
+ {
+
+ /* split architecture may limit the number of voice starts */
+#ifdef MAX_VOICE_STARTS
+ if (pVoiceMgr->numVoiceStarts == MAX_VOICE_STARTS)
+ return EAS_FALSE;
+ pVoiceMgr->numVoiceStarts++;
+#endif
+
+ /* get initial parameters */
+ vCfg.feedback = pRegion->feedback;
+ vCfg.voiceGain = (EAS_U16) pFMVoice->voiceGain;
+
+#if (NUM_OUTPUT_CHANNELS == 2)
+ vCfg.pan = pFMVoice->pan;
+#endif
+
+ /* get voice mode */
+ vCfg.flags = pRegion->region.keyGroupAndFlags & 7;
+
+ /* get operator parameters */
+ for (oper = 0; oper < 4; oper++)
+ {
+ /* calculate initial gain */
+ vCfg.gain[oper] = (EAS_U16) FMUL_15x15(pFMVoice->oper[oper].baseGain, pFMVoice->oper[oper].envGain);
+ vCfg.outputGain[oper] = pFMVoice->oper[oper].outputGain;
+
+ /* copy noise waveform flag */
+ if (pRegion->oper[oper].flags & FM_OPER_FLAG_NOISE)
+ vCfg.flags |= (EAS_U8) (FLAG_FM_ENG_VOICE_OP1_NOISE << oper);
+ }
+
+#ifdef FM_OFFBOARD
+ FM_ConfigVoice(voiceNum, &vCfg, pVoiceMgr->pFrameBuffer);
+#else
+ FM_ConfigVoice(voiceNum, &vCfg, NULL);
+#endif
+
+ /* clear startup flag */
+ pVoice->voiceFlags &= ~VOICE_FLAG_NO_SAMPLES_SYNTHESIZED_YET;
+ }
+
+ /* calculate new synthesis parameters */
+ done = FM_UpdateDynamic(pVoice, pFMVoice, pRegion, pChannel);
+
+ /* calculate LFO gain modulation */
+ /*lint -e{702} <use shift for performance> */
+ temp = ((fmScaleTable[pRegion->vibTrem & 0x0f] >> 1) * pFMVoice->lfoValue) >> FM_LFO_GAIN_SHIFT;
+
+ /* include channel gain */
+ temp += pChannel->staticGain;
+
+ /* -32768 or lower is infinite attenuation */
+ if (temp < -32767)
+ voiceGainTarget = 0;
+
+ /* translate to linear gain multiplier */
+ else
+ voiceGainTarget = EAS_LogToLinear16(temp);
+
+ /* include synth master volume */
+ voiceGainTarget = (EAS_U16) FMUL_15x15(voiceGainTarget, pSynth->masterVolume);
+
+ /* save target values for this frame */
+ vFrame.voiceGain = voiceGainTarget;
+
+ /* assume voice output is zero */
+ pVoice->gain = 0;
+
+ /* save operator targets for this frame */
+ for (oper = 0; oper < 4; oper++)
+ {
+ vFrame.gain[oper] = (EAS_U16) FMUL_15x15(pFMVoice->oper[oper].baseGain, pFMVoice->oper[oper].envGain);
+ vFrame.pitch[oper] = pFMVoice->oper[oper].pitch;
+
+ /* use the highest output envelope level as the gain for the voice stealing algorithm */
+ if (FM_SynthIsOutputOperator(pRegion, oper))
+ pVoice->gain = max(pVoice->gain, (EAS_I16) vFrame.gain[oper]);
+ }
+
+ /* consider voice gain multiplier in calculating gain for stealing algorithm */
+ pVoice->gain = (EAS_I16) FMUL_15x15(voiceGainTarget, pVoice->gain);
+
+ /* synthesize samples */
+#ifdef FM_OFFBOARD
+ FM_ProcessVoice(voiceNum, &vFrame, numSamples, pVoiceMgr->operMixBuffer, pVoiceMgr->voiceBuffer, pMixBuffer, pVoiceMgr->pFrameBuffer);
+#else
+ FM_ProcessVoice(voiceNum, &vFrame, numSamples, pVoiceMgr->operMixBuffer, pVoiceMgr->voiceBuffer, pMixBuffer, NULL);
+#endif
+
+ return done;
+}
+
diff --git a/arm-hybrid-22k/lib_src/eas_fmsynth.h b/arm-hybrid-22k/lib_src/eas_fmsynth.h new file mode 100644 index 0000000..76f8adc --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_fmsynth.h @@ -0,0 +1,81 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_fmsynth.h
+ *
+ * Contents and purpose:
+ * Implements the FM 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: 90 $
+ * $Date: 2006-07-11 20:18:13 -0700 (Tue, 11 Jul 2006) $
+ *----------------------------------------------------------------------------
+*/
+
+#ifndef fmsynthH
+#define fmsynthH
+
+#include "eas_data.h"
+
+#if defined (_FM_SYNTH)
+
+/* FM envelope state */
+typedef enum {
+ eFMEnvelopeStateAttack = 0,
+ eFMEnvelopeStateDecay,
+ eFMEnvelopeStateSustain,
+ eFMEnvelopeStateRelease,
+ eFMEnvelopeStateMuted,
+ eFMEnvelopeStateInvalid /* should never be in this state! */
+} E_FM_ENVELOPE_STATE;
+
+/*------------------------------------
+ * S_OPERATOR data structure
+ *------------------------------------
+*/
+typedef struct s_operator_tag
+{
+ EAS_I16 pitch; /* operator pitch in cents */
+ EAS_U16 envGain; /* envelope target */
+ EAS_I16 baseGain; /* patch gain (inc. vel & key scale) */
+ EAS_U16 outputGain; /* current output gain */
+ EAS_U16 envRate; /* calculated envelope rate */
+ EAS_U8 envState; /* envelope state */
+ EAS_U8 pad; /* pad to 16-bits */
+} S_OPERATOR;
+#endif
+
+typedef struct s_fm_voice_tag
+{
+ S_OPERATOR oper[4]; /* operator data */
+ EAS_I16 voiceGain; /* LFO + channel parameters */
+ EAS_U16 lfoPhase; /* LFO current phase */
+ EAS_I16 lfoValue; /* LFO current value */
+ EAS_U16 lfoDelay; /* keeps track of elapsed delay time */
+ EAS_I8 pan; /* stereo pan value (-64 to +64) */
+ EAS_I8 pad; /* reserved to maintain alignment */
+} S_FM_VOICE;
+
+#ifdef _FM_EDITOR
+extern S_FM_REGION newPatch;
+extern S_FM_REGION OriginalPatch;
+#endif
+
+extern EAS_U32 freqTable[];
+
+#endif
diff --git a/arm-hybrid-22k/lib_src/eas_fmtables.c b/arm-hybrid-22k/lib_src/eas_fmtables.c new file mode 100644 index 0000000..25c6961 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_fmtables.c @@ -0,0 +1,368 @@ +/*----------------------------------------------------------------------------
+ *
+ * File:
+ * eas_fmtables.c
+ *
+ * Contents and purpose:
+ * Contains lookup tables for the FM synthesizer
+ *
+ *
+ * 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"
+
+/* this table is needed by the DSP and the main processor */
+const EAS_U8 fmScaleTable[16] =
+{
+ 0,8,16,24,32,40,48,56,64,72,80,96,128,160,192,255
+};
+
+/* these tables are needed on the main processor */
+#ifndef _DSP_CODE
+const EAS_I16 fmControlTable[128] =
+{
+ -32768,-14313,-12265,-11067,-10217,-9558,-9019,-8563,
+ -8169,-7821,-7510,-7228,-6971,-6734,-6515,-6312,
+ -6121,-5942,-5773,-5613,-5462,-5317,-5180,-5049,
+ -4923,-4802,-4686,-4575,-4467,-4364,-4264,-4167,
+ -4073,-3982,-3894,-3808,-3725,-3644,-3565,-3488,
+ -3414,-3341,-3269,-3200,-3132,-3066,-3001,-2937,
+ -2875,-2814,-2754,-2696,-2638,-2582,-2527,-2473,
+ -2419,-2367,-2316,-2265,-2216,-2167,-2119,-2071,
+ -2025,-1979,-1934,-1889,-1846,-1803,-1760,-1718,
+ -1677,-1636,-1596,-1556,-1517,-1478,-1440,-1403,
+ -1366,-1329,-1293,-1257,-1221,-1186,-1152,-1118,
+ -1084,-1051,-1018,-985,-953,-921,-889,-858,
+ -827,-796,-766,-736,-706,-677,-648,-619,
+ -590,-562,-534,-506,-479,-452,-425,-398,
+ -371,-345,-319,-293,-268,-242,-217,-192,
+ -168,-143,-119,-95,-71,-47,-23,0
+};
+
+const EAS_U16 fmRateTable[128] =
+{
+ 32767,32764,32758,32747,32731,32712,32688,32659,
+ 32627,32590,32548,32503,32453,32398,32340,32277,
+ 32211,32140,32065,31985,31902,31815,31724,31628,
+ 31529,31426,31319,31208,31094,30976,30854,30728,
+ 30599,30466,30330,30191,30048,29902,29752,29599,
+ 29443,29285,29123,28958,28790,28619,28445,28269,
+ 28090,27909,27725,27538,27349,27158,26964,26769,
+ 26571,26371,26169,25965,25760,25552,25343,25132,
+ 24920,24706,24490,24274,24056,23836,23616,23394,
+ 23172,22948,22724,22499,22273,22046,21819,21591,
+ 21363,21135,20906,20676,20447,20217,19987,19758,
+ 19528,19298,19069,18840,18610,18382,18153,17926,
+ 17698,17471,17245,17020,16795,16571,16347,16125,
+ 15903,15683,15463,15245,15027,14811,14596,14382,
+ 14169,13957,13747,13538,13331,13125,12920,12717,
+ 12516,12316,12117,11921,11725,11532,11340,0
+};
+
+const EAS_U16 fmAttackTable[15] =
+{
+ 27,54,109,327,655,1310,2730,4095,
+ 4681,5461,6553,8191,10922,16383,32767
+};
+
+const EAS_U8 fmDecayTable[16] =
+{
+ 4,7,10,15,20,25,30,35,40,50,60,70,80,90,100,127
+};
+
+const EAS_U8 fmReleaseTable[16] =
+{
+ 10,15,20,25,30,35,40,45,50,60,70,80,90,100,113,127
+};
+#endif
+
+/* this table is needed only on the DSP */
+#if defined(_DSP_CODE) || !defined(_SPLIT_ARCHITECTURE)
+//---------------------------------------------------------------------
+// sineTable
+//
+// Contains sine lookup table
+//---------------------------------------------------------------------
+
+const EAS_I16 sineTable[2048] =
+{
+ 0,101,201,302,402,503,603,704,
+ 804,905,1005,1106,1206,1307,1407,1507,
+ 1608,1708,1809,1909,2009,2110,2210,2310,
+ 2410,2511,2611,2711,2811,2911,3012,3112,
+ 3212,3312,3412,3512,3612,3712,3811,3911,
+ 4011,4111,4210,4310,4410,4509,4609,4708,
+ 4808,4907,5007,5106,5205,5305,5404,5503,
+ 5602,5701,5800,5899,5998,6096,6195,6294,
+ 6393,6491,6590,6688,6786,6885,6983,7081,
+ 7179,7277,7375,7473,7571,7669,7767,7864,
+ 7962,8059,8157,8254,8351,8448,8545,8642,
+ 8739,8836,8933,9030,9126,9223,9319,9416,
+ 9512,9608,9704,9800,9896,9992,10087,10183,
+ 10278,10374,10469,10564,10659,10754,10849,10944,
+ 11039,11133,11228,11322,11417,11511,11605,11699,
+ 11793,11886,11980,12074,12167,12260,12353,12446,
+ 12539,12632,12725,12817,12910,13002,13094,13187,
+ 13279,13370,13462,13554,13645,13736,13828,13919,
+ 14010,14101,14191,14282,14372,14462,14553,14643,
+ 14732,14822,14912,15001,15090,15180,15269,15358,
+ 15446,15535,15623,15712,15800,15888,15976,16063,
+ 16151,16238,16325,16413,16499,16586,16673,16759,
+ 16846,16932,17018,17104,17189,17275,17360,17445,
+ 17530,17615,17700,17784,17869,17953,18037,18121,
+ 18204,18288,18371,18454,18537,18620,18703,18785,
+ 18868,18950,19032,19113,19195,19276,19357,19438,
+ 19519,19600,19680,19761,19841,19921,20000,20080,
+ 20159,20238,20317,20396,20475,20553,20631,20709,
+ 20787,20865,20942,21019,21096,21173,21250,21326,
+ 21403,21479,21554,21630,21705,21781,21856,21930,
+ 22005,22079,22154,22227,22301,22375,22448,22521,
+ 22594,22667,22739,22812,22884,22956,23027,23099,
+ 23170,23241,23311,23382,23452,23522,23592,23662,
+ 23731,23801,23870,23938,24007,24075,24143,24211,
+ 24279,24346,24413,24480,24547,24613,24680,24746,
+ 24811,24877,24942,25007,25072,25137,25201,25265,
+ 25329,25393,25456,25519,25582,25645,25708,25770,
+ 25832,25893,25955,26016,26077,26138,26198,26259,
+ 26319,26378,26438,26497,26556,26615,26674,26732,
+ 26790,26848,26905,26962,27019,27076,27133,27189,
+ 27245,27300,27356,27411,27466,27521,27575,27629,
+ 27683,27737,27790,27843,27896,27949,28001,28053,
+ 28105,28157,28208,28259,28310,28360,28411,28460,
+ 28510,28560,28609,28658,28706,28755,28803,28850,
+ 28898,28945,28992,29039,29085,29131,29177,29223,
+ 29268,29313,29358,29403,29447,29491,29534,29578,
+ 29621,29664,29706,29749,29791,29832,29874,29915,
+ 29956,29997,30037,30077,30117,30156,30195,30234,
+ 30273,30311,30349,30387,30424,30462,30498,30535,
+ 30571,30607,30643,30679,30714,30749,30783,30818,
+ 30852,30885,30919,30952,30985,31017,31050,31082,
+ 31113,31145,31176,31206,31237,31267,31297,31327,
+ 31356,31385,31414,31442,31470,31498,31526,31553,
+ 31580,31607,31633,31659,31685,31710,31736,31760,
+ 31785,31809,31833,31857,31880,31903,31926,31949,
+ 31971,31993,32014,32036,32057,32077,32098,32118,
+ 32137,32157,32176,32195,32213,32232,32250,32267,
+ 32285,32302,32318,32335,32351,32367,32382,32397,
+ 32412,32427,32441,32455,32469,32482,32495,32508,
+ 32521,32533,32545,32556,32567,32578,32589,32599,
+ 32609,32619,32628,32637,32646,32655,32663,32671,
+ 32678,32685,32692,32699,32705,32711,32717,32722,
+ 32728,32732,32737,32741,32745,32748,32752,32755,
+ 32757,32759,32761,32763,32765,32766,32766,32767,
+ 32767,32767,32766,32766,32765,32763,32761,32759,
+ 32757,32755,32752,32748,32745,32741,32737,32732,
+ 32728,32722,32717,32711,32705,32699,32692,32685,
+ 32678,32671,32663,32655,32646,32637,32628,32619,
+ 32609,32599,32589,32578,32567,32556,32545,32533,
+ 32521,32508,32495,32482,32469,32455,32441,32427,
+ 32412,32397,32382,32367,32351,32335,32318,32302,
+ 32285,32267,32250,32232,32213,32195,32176,32157,
+ 32137,32118,32098,32077,32057,32036,32014,31993,
+ 31971,31949,31926,31903,31880,31857,31833,31809,
+ 31785,31760,31736,31710,31685,31659,31633,31607,
+ 31580,31553,31526,31498,31470,31442,31414,31385,
+ 31356,31327,31297,31267,31237,31206,31176,31145,
+ 31113,31082,31050,31017,30985,30952,30919,30885,
+ 30852,30818,30783,30749,30714,30679,30643,30607,
+ 30571,30535,30498,30462,30424,30387,30349,30311,
+ 30273,30234,30195,30156,30117,30077,30037,29997,
+ 29956,29915,29874,29832,29791,29749,29706,29664,
+ 29621,29578,29534,29491,29447,29403,29358,29313,
+ 29268,29223,29177,29131,29085,29039,28992,28945,
+ 28898,28850,28803,28755,28706,28658,28609,28560,
+ 28510,28460,28411,28360,28310,28259,28208,28157,
+ 28105,28053,28001,27949,27896,27843,27790,27737,
+ 27683,27629,27575,27521,27466,27411,27356,27300,
+ 27245,27189,27133,27076,27019,26962,26905,26848,
+ 26790,26732,26674,26615,26556,26497,26438,26378,
+ 26319,26259,26198,26138,26077,26016,25955,25893,
+ 25832,25770,25708,25645,25582,25519,25456,25393,
+ 25329,25265,25201,25137,25072,25007,24942,24877,
+ 24811,24746,24680,24613,24547,24480,24413,24346,
+ 24279,24211,24143,24075,24007,23938,23870,23801,
+ 23731,23662,23592,23522,23452,23382,23311,23241,
+ 23170,23099,23027,22956,22884,22812,22739,22667,
+ 22594,22521,22448,22375,22301,22227,22154,22079,
+ 22005,21930,21856,21781,21705,21630,21554,21479,
+ 21403,21326,21250,21173,21096,21019,20942,20865,
+ 20787,20709,20631,20553,20475,20396,20317,20238,
+ 20159,20080,20000,19921,19841,19761,19680,19600,
+ 19519,19438,19357,19276,19195,19113,19032,18950,
+ 18868,18785,18703,18620,18537,18454,18371,18288,
+ 18204,18121,18037,17953,17869,17784,17700,17615,
+ 17530,17445,17360,17275,17189,17104,17018,16932,
+ 16846,16759,16673,16586,16499,16413,16325,16238,
+ 16151,16063,15976,15888,15800,15712,15623,15535,
+ 15446,15358,15269,15180,15090,15001,14912,14822,
+ 14732,14643,14553,14462,14372,14282,14191,14101,
+ 14010,13919,13828,13736,13645,13554,13462,13370,
+ 13279,13187,13094,13002,12910,12817,12725,12632,
+ 12539,12446,12353,12260,12167,12074,11980,11886,
+ 11793,11699,11605,11511,11417,11322,11228,11133,
+ 11039,10944,10849,10754,10659,10564,10469,10374,
+ 10278,10183,10087,9992,9896,9800,9704,9608,
+ 9512,9416,9319,9223,9126,9030,8933,8836,
+ 8739,8642,8545,8448,8351,8254,8157,8059,
+ 7962,7864,7767,7669,7571,7473,7375,7277,
+ 7179,7081,6983,6885,6786,6688,6590,6491,
+ 6393,6294,6195,6096,5998,5899,5800,5701,
+ 5602,5503,5404,5305,5205,5106,5007,4907,
+ 4808,4708,4609,4509,4410,4310,4210,4111,
+ 4011,3911,3811,3712,3612,3512,3412,3312,
+ 3212,3112,3012,2911,2811,2711,2611,2511,
+ 2410,2310,2210,2110,2009,1909,1809,1708,
+ 1608,1507,1407,1307,1206,1106,1005,905,
+ 804,704,603,503,402,302,201,101,
+ 0,-101,-201,-302,-402,-503,-603,-704,
+ -804,-905,-1005,-1106,-1206,-1307,-1407,-1507,
+ -1608,-1708,-1809,-1909,-2009,-2110,-2210,-2310,
+ -2410,-2511,-2611,-2711,-2811,-2911,-3012,-3112,
+ -3212,-3312,-3412,-3512,-3612,-3712,-3811,-3911,
+ -4011,-4111,-4210,-4310,-4410,-4509,-4609,-4708,
+ -4808,-4907,-5007,-5106,-5205,-5305,-5404,-5503,
+ -5602,-5701,-5800,-5899,-5998,-6096,-6195,-6294,
+ -6393,-6491,-6590,-6688,-6786,-6885,-6983,-7081,
+ -7179,-7277,-7375,-7473,-7571,-7669,-7767,-7864,
+ -7962,-8059,-8157,-8254,-8351,-8448,-8545,-8642,
+ -8739,-8836,-8933,-9030,-9126,-9223,-9319,-9416,
+ -9512,-9608,-9704,-9800,-9896,-9992,-10087,-10183,
+ -10278,-10374,-10469,-10564,-10659,-10754,-10849,-10944,
+ -11039,-11133,-11228,-11322,-11417,-11511,-11605,-11699,
+ -11793,-11886,-11980,-12074,-12167,-12260,-12353,-12446,
+ -12539,-12632,-12725,-12817,-12910,-13002,-13094,-13187,
+ -13279,-13370,-13462,-13554,-13645,-13736,-13828,-13919,
+ -14010,-14101,-14191,-14282,-14372,-14462,-14553,-14643,
+ -14732,-14822,-14912,-15001,-15090,-15180,-15269,-15358,
+ -15446,-15535,-15623,-15712,-15800,-15888,-15976,-16063,
+ -16151,-16238,-16325,-16413,-16499,-16586,-16673,-16759,
+ -16846,-16932,-17018,-17104,-17189,-17275,-17360,-17445,
+ -17530,-17615,-17700,-17784,-17869,-17953,-18037,-18121,
+ -18204,-18288,-18371,-18454,-18537,-18620,-18703,-18785,
+ -18868,-18950,-19032,-19113,-19195,-19276,-19357,-19438,
+ -19519,-19600,-19680,-19761,-19841,-19921,-20000,-20080,
+ -20159,-20238,-20317,-20396,-20475,-20553,-20631,-20709,
+ -20787,-20865,-20942,-21019,-21096,-21173,-21250,-21326,
+ -21403,-21479,-21554,-21630,-21705,-21781,-21856,-21930,
+ -22005,-22079,-22154,-22227,-22301,-22375,-22448,-22521,
+ -22594,-22667,-22739,-22812,-22884,-22956,-23027,-23099,
+ -23170,-23241,-23311,-23382,-23452,-23522,-23592,-23662,
+ -23731,-23801,-23870,-23938,-24007,-24075,-24143,-24211,
+ -24279,-24346,-24413,-24480,-24547,-24613,-24680,-24746,
+ -24811,-24877,-24942,-25007,-25072,-25137,-25201,-25265,
+ -25329,-25393,-25456,-25519,-25582,-25645,-25708,-25770,
+ -25832,-25893,-25955,-26016,-26077,-26138,-26198,-26259,
+ -26319,-26378,-26438,-26497,-26556,-26615,-26674,-26732,
+ -26790,-26848,-26905,-26962,-27019,-27076,-27133,-27189,
+ -27245,-27300,-27356,-27411,-27466,-27521,-27575,-27629,
+ -27683,-27737,-27790,-27843,-27896,-27949,-28001,-28053,
+ -28105,-28157,-28208,-28259,-28310,-28360,-28411,-28460,
+ -28510,-28560,-28609,-28658,-28706,-28755,-28803,-28850,
+ -28898,-28945,-28992,-29039,-29085,-29131,-29177,-29223,
+ -29268,-29313,-29358,-29403,-29447,-29491,-29534,-29578,
+ -29621,-29664,-29706,-29749,-29791,-29832,-29874,-29915,
+ -29956,-29997,-30037,-30077,-30117,-30156,-30195,-30234,
+ -30273,-30311,-30349,-30387,-30424,-30462,-30498,-30535,
+ -30571,-30607,-30643,-30679,-30714,-30749,-30783,-30818,
+ -30852,-30885,-30919,-30952,-30985,-31017,-31050,-31082,
+ -31113,-31145,-31176,-31206,-31237,-31267,-31297,-31327,
+ -31356,-31385,-31414,-31442,-31470,-31498,-31526,-31553,
+ -31580,-31607,-31633,-31659,-31685,-31710,-31736,-31760,
+ -31785,-31809,-31833,-31857,-31880,-31903,-31926,-31949,
+ -31971,-31993,-32014,-32036,-32057,-32077,-32098,-32118,
+ -32137,-32157,-32176,-32195,-32213,-32232,-32250,-32267,
+ -32285,-32302,-32318,-32335,-32351,-32367,-32382,-32397,
+ -32412,-32427,-32441,-32455,-32469,-32482,-32495,-32508,
+ -32521,-32533,-32545,-32556,-32567,-32578,-32589,-32599,
+ -32609,-32619,-32628,-32637,-32646,-32655,-32663,-32671,
+ -32678,-32685,-32692,-32699,-32705,-32711,-32717,-32722,
+ -32728,-32732,-32737,-32741,-32745,-32748,-32752,-32755,
+ -32757,-32759,-32761,-32763,-32765,-32766,-32766,-32767,
+ -32767,-32767,-32766,-32766,-32765,-32763,-32761,-32759,
+ -32757,-32755,-32752,-32748,-32745,-32741,-32737,-32732,
+ -32728,-32722,-32717,-32711,-32705,-32699,-32692,-32685,
+ -32678,-32671,-32663,-32655,-32646,-32637,-32628,-32619,
+ -32609,-32599,-32589,-32578,-32567,-32556,-32545,-32533,
+ -32521,-32508,-32495,-32482,-32469,-32455,-32441,-32427,
+ -32412,-32397,-32382,-32367,-32351,-32335,-32318,-32302,
+ -32285,-32267,-32250,-32232,-32213,-32195,-32176,-32157,
+ -32137,-32118,-32098,-32077,-32057,-32036,-32014,-31993,
+ -31971,-31949,-31926,-31903,-31880,-31857,-31833,-31809,
+ -31785,-31760,-31736,-31710,-31685,-31659,-31633,-31607,
+ -31580,-31553,-31526,-31498,-31470,-31442,-31414,-31385,
+ -31356,-31327,-31297,-31267,-31237,-31206,-31176,-31145,
+ -31113,-31082,-31050,-31017,-30985,-30952,-30919,-30885,
+ -30852,-30818,-30783,-30749,-30714,-30679,-30643,-30607,
+ -30571,-30535,-30498,-30462,-30424,-30387,-30349,-30311,
+ -30273,-30234,-30195,-30156,-30117,-30077,-30037,-29997,
+ -29956,-29915,-29874,-29832,-29791,-29749,-29706,-29664,
+ -29621,-29578,-29534,-29491,-29447,-29403,-29358,-29313,
+ -29268,-29223,-29177,-29131,-29085,-29039,-28992,-28945,
+ -28898,-28850,-28803,-28755,-28706,-28658,-28609,-28560,
+ -28510,-28460,-28411,-28360,-28310,-28259,-28208,-28157,
+ -28105,-28053,-28001,-27949,-27896,-27843,-27790,-27737,
+ -27683,-27629,-27575,-27521,-27466,-27411,-27356,-27300,
+ -27245,-27189,-27133,-27076,-27019,-26962,-26905,-26848,
+ -26790,-26732,-26674,-26615,-26556,-26497,-26438,-26378,
+ -26319,-26259,-26198,-26138,-26077,-26016,-25955,-25893,
+ -25832,-25770,-25708,-25645,-25582,-25519,-25456,-25393,
+ -25329,-25265,-25201,-25137,-25072,-25007,-24942,-24877,
+ -24811,-24746,-24680,-24613,-24547,-24480,-24413,-24346,
+ -24279,-24211,-24143,-24075,-24007,-23938,-23870,-23801,
+ -23731,-23662,-23592,-23522,-23452,-23382,-23311,-23241,
+ -23170,-23099,-23027,-22956,-22884,-22812,-22739,-22667,
+ -22594,-22521,-22448,-22375,-22301,-22227,-22154,-22079,
+ -22005,-21930,-21856,-21781,-21705,-21630,-21554,-21479,
+ -21403,-21326,-21250,-21173,-21096,-21019,-20942,-20865,
+ -20787,-20709,-20631,-20553,-20475,-20396,-20317,-20238,
+ -20159,-20080,-20000,-19921,-19841,-19761,-19680,-19600,
+ -19519,-19438,-19357,-19276,-19195,-19113,-19032,-18950,
+ -18868,-18785,-18703,-18620,-18537,-18454,-18371,-18288,
+ -18204,-18121,-18037,-17953,-17869,-17784,-17700,-17615,
+ -17530,-17445,-17360,-17275,-17189,-17104,-17018,-16932,
+ -16846,-16759,-16673,-16586,-16499,-16413,-16325,-16238,
+ -16151,-16063,-15976,-15888,-15800,-15712,-15623,-15535,
+ -15446,-15358,-15269,-15180,-15090,-15001,-14912,-14822,
+ -14732,-14643,-14553,-14462,-14372,-14282,-14191,-14101,
+ -14010,-13919,-13828,-13736,-13645,-13554,-13462,-13370,
+ -13279,-13187,-13094,-13002,-12910,-12817,-12725,-12632,
+ -12539,-12446,-12353,-12260,-12167,-12074,-11980,-11886,
+ -11793,-11699,-11605,-11511,-11417,-11322,-11228,-11133,
+ -11039,-10944,-10849,-10754,-10659,-10564,-10469,-10374,
+ -10278,-10183,-10087,-9992,-9896,-9800,-9704,-9608,
+ -9512,-9416,-9319,-9223,-9126,-9030,-8933,-8836,
+ -8739,-8642,-8545,-8448,-8351,-8254,-8157,-8059,
+ -7962,-7864,-7767,-7669,-7571,-7473,-7375,-7277,
+ -7179,-7081,-6983,-6885,-6786,-6688,-6590,-6491,
+ -6393,-6294,-6195,-6096,-5998,-5899,-5800,-5701,
+ -5602,-5503,-5404,-5305,-5205,-5106,-5007,-4907,
+ -4808,-4708,-4609,-4509,-4410,-4310,-4210,-4111,
+ -4011,-3911,-3811,-3712,-3612,-3512,-3412,-3312,
+ -3212,-3112,-3012,-2911,-2811,-2711,-2611,-2511,
+ -2410,-2310,-2210,-2110,-2009,-1909,-1809,-1708,
+ -1608,-1507,-1407,-1307,-1206,-1106,-1005,-905,
+ -804,-704,-603,-503,-402,-302,-201,-101
+};
+#endif
+
diff --git a/arm-hybrid-22k/lib_src/eas_ima_tables.c b/arm-hybrid-22k/lib_src/eas_ima_tables.c new file mode 100644 index 0000000..56bd1eb --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_imaadpcm.c b/arm-hybrid-22k/lib_src/eas_imaadpcm.c new file mode 100644 index 0000000..68bf257 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_imelody.c b/arm-hybrid-22k/lib_src/eas_imelody.c new file mode 100644 index 0000000..9f4d541 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_imelody.c @@ -0,0 +1,1738 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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;
+
+ /* 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-hybrid-22k/lib_src/eas_imelodydata.c b/arm-hybrid-22k/lib_src/eas_imelodydata.c new file mode 100644 index 0000000..e72dc0b --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_imelodydata.h b/arm-hybrid-22k/lib_src/eas_imelodydata.h new file mode 100644 index 0000000..303b8f6 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_math.c b/arm-hybrid-22k/lib_src/eas_math.c new file mode 100644 index 0000000..12d788e --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_math.h b/arm-hybrid-22k/lib_src/eas_math.h new file mode 100644 index 0000000..719270b --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_midi.c b/arm-hybrid-22k/lib_src/eas_midi.c new file mode 100644 index 0000000..08aed72 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_midi.h b/arm-hybrid-22k/lib_src/eas_midi.h new file mode 100644 index 0000000..37a03ee --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_midictrl.h b/arm-hybrid-22k/lib_src/eas_midictrl.h new file mode 100644 index 0000000..0c4217d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_mididata.c b/arm-hybrid-22k/lib_src/eas_mididata.c new file mode 100644 index 0000000..2ee907e --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_miditypes.h b/arm-hybrid-22k/lib_src/eas_miditypes.h new file mode 100644 index 0000000..0b7f96e --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_mixbuf.c b/arm-hybrid-22k/lib_src/eas_mixbuf.c new file mode 100644 index 0000000..73e969a --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_mixer.c b/arm-hybrid-22k/lib_src/eas_mixer.c new file mode 100644 index 0000000..c4a2f9f --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_mixer.h b/arm-hybrid-22k/lib_src/eas_mixer.h new file mode 100644 index 0000000..2ba2d3d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_ota.c b/arm-hybrid-22k/lib_src/eas_ota.c new file mode 100644 index 0000000..fb81d62 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_otadata.c b/arm-hybrid-22k/lib_src/eas_otadata.c new file mode 100644 index 0000000..237f832 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_otadata.h b/arm-hybrid-22k/lib_src/eas_otadata.h new file mode 100644 index 0000000..63e963f --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pan.c b/arm-hybrid-22k/lib_src/eas_pan.c new file mode 100644 index 0000000..373d90e --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pan.h b/arm-hybrid-22k/lib_src/eas_pan.h new file mode 100644 index 0000000..cefa650 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_parser.h b/arm-hybrid-22k/lib_src/eas_parser.h new file mode 100644 index 0000000..5512c82 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pcm.c b/arm-hybrid-22k/lib_src/eas_pcm.c new file mode 100644 index 0000000..64b8f71 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pcm.h b/arm-hybrid-22k/lib_src/eas_pcm.h new file mode 100644 index 0000000..c161757 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pcmdata.c b/arm-hybrid-22k/lib_src/eas_pcmdata.c new file mode 100644 index 0000000..5649f07 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_pcmdata.h b/arm-hybrid-22k/lib_src/eas_pcmdata.h new file mode 100644 index 0000000..be2f8e5 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_public.c b/arm-hybrid-22k/lib_src/eas_public.c new file mode 100644 index 0000000..ac43261 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_public.c @@ -0,0 +1,2597 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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;
+}
+
+/*----------------------------------------------------------------------------
+ * 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;
+
+ /* 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-hybrid-22k/lib_src/eas_reverb.c b/arm-hybrid-22k/lib_src/eas_reverb.c new file mode 100644 index 0000000..6d99862 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_reverbdata.c b/arm-hybrid-22k/lib_src/eas_reverbdata.c new file mode 100644 index 0000000..5d48c1b --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_reverbdata.h b/arm-hybrid-22k/lib_src/eas_reverbdata.h new file mode 100644 index 0000000..ef424da --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_rtttl.c b/arm-hybrid-22k/lib_src/eas_rtttl.c new file mode 100644 index 0000000..486ad60 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_rtttldata.c b/arm-hybrid-22k/lib_src/eas_rtttldata.c new file mode 100644 index 0000000..7a500bd --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_rtttldata.h b/arm-hybrid-22k/lib_src/eas_rtttldata.h new file mode 100644 index 0000000..bf4c38b --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_smf.c b/arm-hybrid-22k/lib_src/eas_smf.c new file mode 100644 index 0000000..7b56e97 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_smf.c @@ -0,0 +1,1203 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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;
+ }
+
+ /* 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 & 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-hybrid-22k/lib_src/eas_smf.h b/arm-hybrid-22k/lib_src/eas_smf.h new file mode 100644 index 0000000..9f66ab9 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_smfdata.c b/arm-hybrid-22k/lib_src/eas_smfdata.c new file mode 100644 index 0000000..5c27551 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_smfdata.h b/arm-hybrid-22k/lib_src/eas_smfdata.h new file mode 100644 index 0000000..cf59cdc --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_sndlib.h b/arm-hybrid-22k/lib_src/eas_sndlib.h new file mode 100644 index 0000000..e05bee0 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_synth.h b/arm-hybrid-22k/lib_src/eas_synth.h new file mode 100644 index 0000000..b242b03 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_synth_protos.h b/arm-hybrid-22k/lib_src/eas_synth_protos.h new file mode 100644 index 0000000..a2ef10d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_synthcfg.h b/arm-hybrid-22k/lib_src/eas_synthcfg.h new file mode 100644 index 0000000..2491e6d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_vm_protos.h b/arm-hybrid-22k/lib_src/eas_vm_protos.h new file mode 100644 index 0000000..eb49ba8 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_voicemgt.c b/arm-hybrid-22k/lib_src/eas_voicemgt.c new file mode 100644 index 0000000..873f29d --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wavefile.c b/arm-hybrid-22k/lib_src/eas_wavefile.c new file mode 100644 index 0000000..d3f3ba0 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wavefile.h b/arm-hybrid-22k/lib_src/eas_wavefile.h new file mode 100644 index 0000000..b8b76df --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wavefiledata.c b/arm-hybrid-22k/lib_src/eas_wavefiledata.c new file mode 100644 index 0000000..3742aa6 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wt_IPC_frame.h b/arm-hybrid-22k/lib_src/eas_wt_IPC_frame.h new file mode 100644 index 0000000..376fd3a --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wtengine.c b/arm-hybrid-22k/lib_src/eas_wtengine.c new file mode 100644 index 0000000..4215924 --- /dev/null +++ b/arm-hybrid-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)
+ 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-hybrid-22k/lib_src/eas_wtengine.h b/arm-hybrid-22k/lib_src/eas_wtengine.h new file mode 100644 index 0000000..6401322 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/eas_wtsynth.c b/arm-hybrid-22k/lib_src/eas_wtsynth.c new file mode 100644 index 0000000..8098e09 --- /dev/null +++ b/arm-hybrid-22k/lib_src/eas_wtsynth.c @@ -0,0 +1,1251 @@ +/*----------------------------------------------------------------------------
+ *
+ * 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);
+ pWTIntFrame->numSamples = 1 + (numSamples / pWTIntFrame->frame.phaseIncrement);
+
+ /* 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;
+
+#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-hybrid-22k/lib_src/eas_wtsynth.h b/arm-hybrid-22k/lib_src/eas_wtsynth.h new file mode 100644 index 0000000..106ab40 --- /dev/null +++ b/arm-hybrid-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-hybrid-22k/lib_src/hybrid_22khz_mcu.c b/arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c new file mode 100644 index 0000000..880819c --- /dev/null +++ b/arm-hybrid-22k/lib_src/hybrid_22khz_mcu.c @@ -0,0 +1,5149 @@ +/*----------------------------------------------------------------------------
+ *
+ * Filename: C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c
+ * Source: C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.dls
+ * CmdLine: -w C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c -l C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz.log -d 0 -c -f C:\Sonic\Release3-5\EASLib\FMSynth\GMdblib-3.fml C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.dls -w -l -d -c -f C:\Sonic\Trunk\Wavetables\Sonic_20Khz_Drums.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:
+ * $Revision:$
+ * $Date:$
+ *----------------------------------------------------------------------------
+*/
+
+#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
+ }
+}; /*end Articulations */
+
+/*----------------------------------------------------------------------------
+ * Regions
+ *----------------------------------------------------------------------------
+*/
+const S_WT_REGION eas_regions[] =
+{
+ { { 0, 27, 27 }, -2868, 16422, 0, 0, 19, 0 }, /* region 0 */
+ { { 0, 28, 28 }, -3568, 32767, 0, 0, 13, 0 }, /* region 1 */
+ { { 0, 29, 29 }, -4553, 32767, 0, 0, 9, 1 }, /* region 2 */
+ { { 0, 30, 30 }, -4853, 32767, 0, 0, 9, 2 }, /* region 3 */
+ { { 0, 31, 31 }, -3868, 23197, 0, 0, 15, 3 }, /* region 4 */
+ { { 1536, 32, 32 }, -3368, 20675, 0, 0, 29, 4 }, /* region 5 */
+ { { 1537, 33, 33 }, -3868, 20675, 792, 800, 17, 5 }, /* region 6 */
+ { { 1537, 34, 34 }, -3968, 16422, 792, 800, 17, 6 }, /* region 7 */
+ { { 0, 35, 35 }, -4968, 32767, 0, 0, 20, 7 }, /* region 8 */
+ { { 0, 36, 36 }, -4968, 32767, 0, 0, 20, 7 }, /* region 9 */
+ { { 0, 37, 37 }, -4051, 18426, 0, 0, 16, 8 }, /* region 10 */
+ { { 0, 38, 38 }, -4151, 23197, 0, 0, 5, 9 }, /* region 11 */
+ { { 0, 39, 39 }, -4668, 23197, 0, 0, 12, 10 }, /* region 12 */
+ { { 0, 40, 40 }, -4151, 23197, 0, 0, 5, 4 }, /* region 13 */
+ { { 1, 41, 41 }, -5855, 26028, 798, 993, 14, 11 }, /* region 14 */
+ { { 257, 42, 42 }, -4368, 26028, 4288, 6792, 2, 12 }, /* region 15 */
+ { { 1, 43, 43 }, -5755, 26028, 798, 993, 14, 13 }, /* region 16 */
+ { { 257, 44, 44 }, -4568, 26028, 4288, 6792, 2, 14 }, /* region 17 */
+ { { 1, 45, 45 }, -5755, 26028, 798, 993, 14, 15 }, /* region 18 */
+ { { 257, 46, 46 }, -4768, 26028, 4288, 6792, 2, 16 }, /* region 19 */
+ { { 1, 47, 47 }, -5455, 26028, 798, 993, 14, 17 }, /* region 20 */
+ { { 1, 48, 48 }, -5355, 26028, 798, 993, 14, 18 }, /* region 21 */
+ { { 1, 49, 49 }, -5368, 16422, 1294, 5241, 3, 19 }, /* region 22 */
+ { { 1, 50, 50 }, -5255, 26028, 798, 993, 14, 20 }, /* region 23 */
+ { { 1, 51, 51 }, -5268, 16422, 6592, 9921, 0, 21 }, /* region 24 */
+ { { 1, 52, 52 }, -5768, 32767, 1294, 5241, 3, 22 }, /* region 25 */
+ { { 1, 53, 53 }, -5418, 14636, 6592, 9921, 0, 23 }, /* region 26 */
+ { { 0, 54, 54 }, -5751, 26028, 0, 0, 11, 24 }, /* region 27 */
+ { { 1, 55, 55 }, -5468, 32767, 1294, 5241, 3, 25 }, /* region 28 */
+ { { 0, 56, 56 }, -7255, 32767, 0, 0, 24, 26 }, /* region 29 */
+ { { 1, 57, 57 }, -5868, 32767, 1294, 5241, 3, 27 }, /* region 30 */
+ { { 1, 58, 58 }, -7053, 23197, 0, 166, 26, 28 }, /* region 31 */
+ { { 1, 59, 59 }, -5968, 16422, 6592, 9921, 0, 29 }, /* region 32 */
+ { { 1, 60, 60 }, -6453, 23197, 432, 582, 18, 30 }, /* region 33 */
+ { { 1, 61, 61 }, -6853, 16422, 432, 582, 18, 30 }, /* region 34 */
+ { { 1, 62, 62 }, -7253, 20675, 432, 582, 18, 31 }, /* region 35 */
+ { { 1, 63, 63 }, -7353, 23197, 432, 582, 18, 32 }, /* region 36 */
+ { { 1, 64, 64 }, -7953, 23197, 432, 582, 18, 33 }, /* region 37 */
+ { { 0, 65, 65 }, -7555, 32767, 0, 0, 4, 34 }, /* region 38 */
+ { { 0, 66, 66 }, -7955, 20675, 0, 0, 4, 34 }, /* region 39 */
+ { { 512, 67, 67 }, -7155, 18426, 0, 0, 24, 35 }, /* region 40 */
+ { { 512, 68, 68 }, -7755, 18426, 0, 0, 24, 35 }, /* region 41 */
+ { { 0, 69, 69 }, -7755, 32767, 0, 0, 21, 36 }, /* region 42 */
+ { { 0, 70, 70 }, -6855, 21900, 0, 0, 21, 37 }, /* region 43 */
+ { { 769, 71, 71 }, -6355, 23197, 0, 1226, 10, 38 }, /* region 44 */
+ { { 769, 72, 72 }, -6955, 26028, 0, 1226, 10, 38 }, /* region 45 */
+ { { 1024, 73, 73 }, -7955, 32767, 0, 0, 7, 39 }, /* region 46 */
+ { { 1024, 74, 74 }, -8455, 32767, 0, 0, 7, 40 }, /* region 47 */
+ { { 1, 75, 75 }, -8068, 23197, 0, 29, 30, 41 }, /* region 48 */
+ { { 0, 76, 76 }, -10455, 23197, 0, 0, 28, 42 }, /* region 49 */
+ { { 0, 77, 77 }, -10055, 23197, 0, 0, 28, 43 }, /* region 50 */
+ { { 0, 78, 78 }, -8853, 16422, 0, 0, 23, 44 }, /* region 51 */
+ { { 0, 79, 79 }, -10253, 16422, 0, 0, 23, 45 }, /* region 52 */
+ { { 1280, 80, 80 }, -6468, 13045, 0, 0, 25, 46 }, /* region 53 */
+ { { 1280, 81, 81 }, -6568, 16422, 0, 0, 25, 47 }, /* region 54 */
+ { { 0, 82, 82 }, -8455, 20675, 0, 0, 22, 48 }, /* region 55 */
+ { { 0, 83, 83 }, -9068, 32767, 0, 0, 6, 49 }, /* region 56 */
+ { { 1, 84, 84 }, -8568, 23197, 0, 9337, 1, 50 }, /* region 57 */
+ { { 0, 85, 85 }, -9655, 32767, 0, 0, 27, 0 }, /* region 58 */
+ { { 0, 86, 86 }, -9068, 16422, 0, 0, 8, 51 }, /* region 59 */
+ { { 32769, 87, 87 }, -9168, 32767, 1335, 1603, 8, 52 } /* region 60 */
+}; /* end Regions */
+
+/*----------------------------------------------------------------------------
+ * FM Regions
+ *----------------------------------------------------------------------------
+*/
+const S_FM_REGION eas_fmRegions[] =
+{
+
+ { /* FM region 0 */
+ { 37, 0, 127 }, 0, 255, 8, 0,
+ {
+ { 514, 239, 47, 97, 0, 184, 3 },
+ { 1, 244, 89, 114, 0, 248, 2 },
+ { 3370, 244, 49, 76, 40, 192, 2 },
+ { -1, 227, 97, 51, 160, 212, 2 }
+ }
+ },
+ { /* FM region 1 */
+ { 37, 0, 127 }, 160, 255, 8, 0,
+ {
+ { 2514, 223, 95, 72, 0, 176, 3 },
+ { 1, 244, 73, 145, 0, 244, 2 },
+ { 3600, 245, 81, 198, 40, 192, 2 },
+ { 3, 246, 81, 163, 108, 212, 2 }
+ }
+ },
+ { /* FM region 2 */
+ { 37, 0, 127 }, 160, 255, 119, 0,
+ {
+ { 0, 216, 79, 72, 0, 216, 2 },
+ { 2, 244, 73, 145, 0, 244, 2 },
+ { 3370, 247, 33, 182, 60, 204, 2 },
+ { 1200, 246, 65, 163, 108, 204, 2 }
+ }
+ },
+ { /* FM region 3 */
+ { 37, 0, 127 }, 160, 255, 1, 0,
+ {
+ { 3369, 248, 65, 71, 40, 208, 2 },
+ { -3, 245, 88, 113, 0, 244, 2 },
+ { 2784, 225, 65, 133, 80, 192, 2 },
+ { 3, 241, 81, 113, 80, 216, 2 }
+ }
+ },
+ { /* FM region 4 */
+ { 34, 0, 127 }, 0, 255, 128, 0,
+ {
+ { 0, 229, 155, 183, 0, 228, 2 },
+ { -3, 243, 90, 81, 0, 244, 2 },
+ { 4800, 248, 109, 180, 36, 192, 2 },
+ { 3, 245, 90, 85, 16, 244, 2 }
+ }
+ },
+ { /* FM region 5 */
+ { 34, 0, 127 }, 9, 96, 192, 0,
+ {
+ { 1200, 229, 157, 180, 0, 216, 2 },
+ { -3, 244, 90, 81, 0, 244, 2 },
+ { 1902, 255, 111, 182, 80, 208, 2 },
+ { 3, 246, 92, 83, 0, 244, 2 }
+ }
+ },
+ { /* FM region 6 */
+ { 34, 0, 127 }, 0, 255, 154, 0,
+ {
+ { 3102, 244, 63, 102, 228, 228, 2 },
+ { 1200, 247, 93, 97, 0, 236, 2 },
+ { 1902, 255, 63, 98, 156, 220, 2 },
+ { 1200, 244, 92, 98, 0, 236, 2 }
+ }
+ },
+ { /* FM region 7 */
+ { 37, 0, 127 }, 0, 255, 202, 0,
+ {
+ { 0, 251, 131, 19, 216, 220, 2 },
+ { 1201, 247, 62, 113, 0, 240, 2 },
+ { 0, 243, 154, 36, 240, 224, 2 },
+ { 2784, 250, 61, 36, 240, 208, 2 }
+ }
+ },
+ { /* FM region 8 */
+ { 33, 0, 127 }, 0, 255, 80, 0,
+ {
+ { -1, 213, 191, 183, 0, 204, 2 },
+ { 1, 245, 154, 129, 0, 244, 2 },
+ { 3831, 252, 159, 100, 0, 200, 2 },
+ { 1197, 246, 91, 182, 0, 244, 2 }
+ }
+ },
+ { /* FM region 9 */
+ { 34, 0, 127 }, 48, 80, 21, 0,
+ {
+ { 2982, 255, 43, 96, 0, 196, 3 },
+ { 3, 247, 71, 130, 0, 244, 2 },
+ { 3358, 253, 40, 98, 144, 208, 2 },
+ { -2, 246, 70, 130, 0, 236, 2 }
+ }
+ },
+ { /* FM region 10 */
+ { 34, 0, 127 }, 48, 80, 26, 0,
+ {
+ { 3096, 249, 72, 100, 0, 208, 2 },
+ { 2185, 249, 102, 130, 0, 240, 2 },
+ { 3386, 247, 66, 100, 144, 212, 2 },
+ { -2, 247, 102, 130, 0, 240, 2 }
+ }
+ },
+ { /* FM region 11 */
+ { 34, 0, 127 }, 92, 67, 21, 0,
+ {
+ { 2982, 255, 27, 146, 0, 200, 3 },
+ { 3, 246, 68, 146, 0, 240, 2 },
+ { 3358, 250, 149, 116, 144, 208, 2 },
+ { -3, 245, 68, 146, 0, 240, 0 }
+ }
+ },
+ { /* FM region 12 */
+ { 34, 0, 127 }, 0, 67, 0, 0,
+ {
+ { 1500, 239, 60, 151, 0, 220, 2 },
+ { 0, 247, 76, 146, 0, 240, 2 },
+ { 2398, 234, 156, 151, 0, 212, 2 },
+ { 0, 246, 105, 146, 0, 244, 2 }
+ }
+ },
+ { /* FM region 13 */
+ { 34, 0, 127 }, 0, 67, 0, 0,
+ {
+ { 2500, 255, 60, 151, 0, 220, 2 },
+ { 0, 249, 92, 146, 0, 244, 2 },
+ { 3369, 250, 156, 151, 0, 196, 2 },
+ { 0, 248, 89, 146, 0, 244, 2 }
+ }
+ },
+ { /* FM region 14 */
+ { 37, 0, 127 }, 160, 255, 0, 0,
+ {
+ { 2300, 229, 112, 49, 0, 208, 2 },
+ { -3, 247, 67, 50, 0, 248, 2 },
+ { 1074, 255, 41, 49, 0, 196, 2 },
+ { 686, 240, 97, 18, 0, 196, 2 }
+ }
+ },
+ { /* FM region 15 */
+ { 37, 0, 127 }, 160, 255, 219, 0,
+ {
+ { 3369, 255, 65, 70, 40, 216, 2 },
+ { 1, 246, 72, 113, 0, 240, 2 },
+ { 1902, 225, 33, 129, 80, 204, 2 },
+ { 2400, 225, 97, 113, 80, 200, 2 }
+ }
+ },
+ { /* FM region 16 */
+ { 35, 0, 127 }, 32, 48, 151, 0,
+ {
+ { 1201, 215, 35, 66, 252, 208, 0 },
+ { -9581, 254, 63, 177, 240, 240, 3 },
+ { 1902, 248, 47, 64, 112, 244, 2 },
+ { 0, 247, 35, 66, 208, 212, 2 }
+ }
+ },
+ { /* FM region 17 */
+ { 33, 0, 127 }, 0, 255, 153, 0,
+ {
+ { 1, 252, 31, 3, 244, 196, 2 },
+ { -1, 208, 31, 4, 248, 244, 2 },
+ { 1205, 209, 31, 4, 248, 236, 2 },
+ { 1899, 250, 31, 32, 0, 240, 2 }
+ }
+ },
+ { /* FM region 18 */
+ { 34, 0, 127 }, 32, 49, 201, 0,
+ {
+ { 1, 220, 47, 3, 244, 220, 0 },
+ { -10000, 208, 63, 1, 248, 240, 3 },
+ { 1586, 255, 47, 3, 188, 216, 2 },
+ { -1, 202, 63, 32, 80, 232, 2 }
+ }
+ },
+ { /* FM region 19 */
+ { 33, 0, 127 }, 0, 143, 29, 0,
+ {
+ { -1200, 223, 64, 0, 252, 216, 2 },
+ { 1200, 96, 41, 35, 248, 240, 2 },
+ { 1200, 143, 41, 64, 252, 224, 2 },
+ { 3102, 161, 41, 96, 248, 216, 2 }
+ }
+ },
+ { /* FM region 20 */
+ { 34, 0, 127 }, 0, 143, 34, 0,
+ {
+ { -1200, 133, 79, 1, 252, 212, 2 },
+ { 1201, 112, 46, 34, 248, 232, 2 },
+ { 0, 116, 79, 65, 252, 200, 2 },
+ { 1900, 161, 46, 98, 248, 232, 2 }
+ }
+ },
+ { /* FM region 21 */
+ { 34, 0, 127 }, 0, 143, 187, 0,
+ {
+ { 1202, 80, 74, 1, 252, 216, 2 },
+ { 2402, 112, 46, 34, 248, 232, 2 },
+ { 0, 99, 78, 97, 184, 216, 2 },
+ { 1899, 81, 46, 98, 236, 232, 2 }
+ }
+ },
+ { /* FM region 22 */
+ { 37, 0, 127 }, 22, 141, 34, 0,
+ {
+ { 2787, 176, 79, 4, 252, 208, 2 },
+ { 2785, 144, 45, 34, 248, 236, 2 },
+ { 3369, 83, 77, 100, 184, 172, 2 },
+ { 1902, 102, 45, 100, 172, 212, 0 }
+ }
+ },
+ { /* FM region 23 */
+ { 34, 0, 127 }, 0, 143, 135, 0,
+ {
+ { 1900, 112, 79, 3, 252, 220, 2 },
+ { 2400, 128, 45, 34, 248, 232, 2 },
+ { 1200, 115, 77, 98, 184, 220, 2 },
+ { 1904, 97, 45, 98, 236, 232, 2 }
+ }
+ },
+ { /* FM region 24 */
+ { 37, 0, 127 }, 0, 255, 157, 0,
+ {
+ { 1200, 244, 54, 4, 20, 200, 2 },
+ { 0, 245, 92, 130, 0, 244, 2 },
+ { 3802, 247, 68, 21, 0, 196, 2 },
+ { 1, 245, 43, 114, 0, 204, 2 }
+ }
+ },
+ { /* FM region 25 */
+ { 37, 0, 127 }, 0, 128, 83, 0,
+ {
+ { 0, 244, 51, 4, 200, 204, 0 },
+ { 0, 247, 108, 129, 0, 248, 0 },
+ { 2786, 243, 31, 70, 200, 220, 0 },
+ { 1902, 246, 44, 113, 12, 188, 0 }
+ }
+ },
+ { /* FM region 26 */
+ { 37, 0, 127 }, 0, 128, 61, 0,
+ {
+ { 0, 246, 51, 97, 76, 204, 0 },
+ { 0, 244, 60, 97, 0, 240, 0 },
+ { 1786, 255, 31, 64, 0, 180, 0 },
+ { 1200, 247, 60, 97, 12, 204, 0 }
+ }
+ },
+ { /* FM region 27 */
+ { 37, 0, 127 }, 0, 128, 153, 0,
+ {
+ { -2, 243, 53, 99, 96, 200, 0 },
+ { 0, 243, 60, 97, 0, 240, 0 },
+ { 3983, 247, 63, 100, 24, 204, 0 },
+ { 2, 242, 53, 99, 52, 212, 0 }
+ }
+ },
+ { /* FM region 28 */
+ { 37, 0, 127 }, 0, 128, 205, 0,
+ {
+ { -2, 244, 47, 97, 20, 208, 0 },
+ { 0, 252, 75, 193, 0, 248, 0 },
+ { 0, 254, 63, 98, 132, 224, 0 },
+ { 2786, 251, 63, 98, 52, 192, 0 }
+ }
+ },
+ { /* FM region 29 */
+ { 37, 0, 127 }, 0, 128, 221, 0,
+ {
+ { -1, 208, 191, 99, 220, 224, 0 },
+ { 1200, 243, 92, 97, 0, 244, 0 },
+ { 3984, 212, 11, 96, 168, 196, 0 },
+ { 1, 242, 127, 98, 108, 204, 0 }
+ }
+ },
+ { /* FM region 30 */
+ { 37, 0, 127 }, 0, 128, 174, 0,
+ {
+ { -3, 212, 207, 99, 0, 228, 0 },
+ { 1902, 241, 108, 97, 0, 248, 0 },
+ { 3805, 212, 59, 98, 0, 220, 0 },
+ { 1902, 146, 107, 98, 144, 196, 0 }
+ }
+ },
+ { /* FM region 31 */
+ { 41, 0, 127 }, 0, 255, 128, 0,
+ {
+ { 1206, 239, 43, 69, 0, 216, 2 },
+ { 4, 254, 42, 66, 0, 244, 2 },
+ { 702, 88, 55, 66, 0, 204, 2 },
+ { -4, 71, 55, 66, 0, 240, 2 }
+ }
+ },
+ { /* FM region 32 */
+ { 37, 0, 127 }, 0, 255, 85, 0,
+ {
+ { 500, 239, 95, 82, 0, 184, 3 },
+ { 0, 248, 73, 132, 0, 252, 2 },
+ { 2786, 203, 59, 130, 0, 176, 2 },
+ { 0, 216, 42, 100, 0, 208, 2 }
+ }
+ },
+ { /* FM region 33 */
+ { 37, 0, 127 }, 0, 128, 73, 0,
+ {
+ { 1, 229, 54, 131, 160, 208, 0 },
+ { -1, 244, 62, 97, 0, 248, 0 },
+ { 3986, 227, 127, 69, 140, 184, 0 },
+ { 1201, 249, 92, 114, 0, 204, 0 }
+ }
+ },
+ { /* FM region 34 */
+ { 37, 0, 127 }, 0, 128, 73, 0,
+ {
+ { 1, 225, 54, 100, 200, 212, 0 },
+ { -1, 244, 94, 97, 0, 248, 0 },
+ { 3986, 249, 127, 88, 112, 188, 0 },
+ { 1201, 249, 92, 85, 52, 208, 0 }
+ }
+ },
+ { /* FM region 35 */
+ { 37, 0, 127 }, 0, 128, 188, 0,
+ {
+ { -3, 198, 92, 179, 28, 212, 0 },
+ { 0, 243, 90, 145, 0, 248, 0 },
+ { 1901, 215, 95, 69, 28, 196, 0 },
+ { 3, 84, 108, 196, 32, 208, 0 }
+ }
+ },
+ { /* FM region 36 */
+ { 37, 0, 127 }, 0, 136, 6, 0,
+ {
+ { 0, 226, 99, 36, 224, 216, 0 },
+ { 1902, 248, 78, 33, 0, 252, 0 },
+ { 3369, 239, 250, 33, 0, 204, 0 },
+ { 0, 230, 253, 33, 0, 208, 0 }
+ }
+ },
+ { /* FM region 37 */
+ { 37, 0, 127 }, 0, 136, 195, 0,
+ {
+ { 0, 245, 99, 36, 152, 208, 0 },
+ { 1200, 248, 78, 33, 0, 252, 0 },
+ { 3369, 246, 250, 33, 0, 216, 0 },
+ { 0, 246, 61, 33, 0, 180, 0 }
+ }
+ },
+ { /* FM region 38 */
+ { 34, 0, 127 }, 0, 133, 221, 0,
+ {
+ { 1, 244, 67, 35, 80, 220, 0 },
+ { 3, 246, 94, 33, 0, 244, 0 },
+ { -1, 245, 70, 35, 80, 236, 2 },
+ { -3, 246, 63, 33, 0, 236, 2 }
+ }
+ },
+ { /* FM region 39 */
+ { 34, 0, 127 }, 0, 133, 220, 0,
+ {
+ { 0, 114, 51, 34, 132, 208, 0 },
+ { 3, 214, 62, 33, 0, 248, 0 },
+ { 0, 85, 54, 34, 44, 224, 2 },
+ { -3, 214, 63, 33, 0, 236, 2 }
+ }
+ },
+ { /* FM region 40 */
+ { 37, 0, 127 }, 48, 142, 187, 0,
+ {
+ { -1, 33, 22, 33, 200, 208, 0 },
+ { 0, 81, 105, 33, 220, 240, 0 },
+ { 2786, 245, 19, 50, 208, 192, 0 },
+ { 1, 245, 21, 82, 200, 220, 0 }
+ }
+ },
+ { /* FM region 41 */
+ { 37, 0, 127 }, 48, 126, 103, 0,
+ {
+ { -1, 193, 22, 33, 228, 212, 0 },
+ { 0, 81, 105, 33, 220, 244, 0 },
+ { 0, 245, 19, 50, 216, 228, 0 },
+ { 1200, 245, 19, 82, 200, 188, 0 }
+ }
+ },
+ { /* FM region 42 */
+ { 37, 0, 127 }, 16, 126, 202, 0,
+ {
+ { -1, 49, 24, 41, 200, 212, 0 },
+ { 0, 81, 71, 49, 220, 244, 0 },
+ { 3371, 243, 19, 36, 232, 192, 0 },
+ { 1, 242, 24, 36, 220, 212, 0 }
+ }
+ },
+ { /* FM region 43 */
+ { 37, 0, 127 }, 16, 124, 205, 0,
+ {
+ { 0, 129, 24, 49, 208, 200, 0 },
+ { 0, 67, 102, 81, 224, 244, 0 },
+ { 3804, 246, 23, 36, 160, 196, 0 },
+ { 1200, 244, 24, 35, 208, 200, 0 }
+ }
+ },
+ { /* FM region 44 */
+ { 37, 0, 127 }, 48, 144, 208, 0,
+ {
+ { -3, 209, 22, 33, 200, 204, 2 },
+ { 0, 81, 89, 33, 220, 240, 2 },
+ { -5000, 208, 6, 33, 244, 188, 3 },
+ { 3, 97, 89, 33, 224, 200, 0 }
+ }
+ },
+ { /* FM region 45 */
+ { 37, 0, 127 }, 0, 255, 186, 0,
+ {
+ { 500, 223, 95, 0, 0, 192, 3 },
+ { 0, 247, 89, 100, 0, 248, 2 },
+ { 3369, 255, 59, 168, 0, 212, 2 },
+ { 0, 216, 42, 97, 0, 212, 2 }
+ }
+ },
+ { /* FM region 46 */
+ { 34, 0, 127 }, 0, 255, 221, 0,
+ {
+ { 1206, 235, 70, 69, 0, 216, 2 },
+ { 4, 248, 84, 66, 0, 244, 2 },
+ { 1902, 247, 52, 137, 80, 216, 2 },
+ { -4, 245, 84, 131, 0, 240, 2 }
+ }
+ },
+ { /* FM region 47 */
+ { 37, 0, 127 }, 0, 255, 105, 0,
+ {
+ { 387, 231, 115, 34, 4, 216, 2 },
+ { 0, 248, 37, 65, 0, 252, 2 },
+ { 3308, 248, 117, 34, 8, 200, 2 },
+ { 1900, 213, 82, 50, 0, 192, 2 }
+ }
+ },
+ { /* FM region 48 */
+ { 34, 0, 127 }, 32, 160, 221, 0,
+ {
+ { -7, 209, 22, 33, 200, 204, 2 },
+ { -7, 81, 73, 33, 220, 244, 0 },
+ { 7, 209, 22, 33, 200, 208, 0 },
+ { 7, 97, 73, 33, 224, 244, 2 }
+ }
+ },
+ { /* FM region 49 */
+ { 34, 0, 127 }, 64, 128, 189, 0,
+ {
+ { -2, 209, 54, 32, 224, 216, 2 },
+ { -7726, 97, 105, 33, 220, 240, 3 },
+ { 1902, 209, 54, 34, 216, 208, 0 },
+ { 2, 81, 105, 33, 224, 236, 0 }
+ }
+ },
+ { /* FM region 50 */
+ { 34, 0, 127 }, 80, 144, 206, 0,
+ {
+ { -3, 179, 38, 33, 160, 220, 2 },
+ { -7726, 81, 69, 34, 220, 244, 3 },
+ { 3, 193, 38, 33, 240, 212, 0 },
+ { -8000, 65, 69, 34, 224, 236, 3 }
+ }
+ },
+ { /* FM region 51 */
+ { 37, 0, 127 }, 96, 128, 204, 0,
+ {
+ { -3, 97, 38, 33, 180, 216, 0 },
+ { 0, 81, 69, 34, 220, 240, 2 },
+ { 3369, 145, 38, 33, 240, 196, 2 },
+ { -13190, 65, 69, 34, 240, 200, 3 }
+ }
+ },
+ { /* FM region 52 */
+ { 34, 0, 127 }, 64, 128, 108, 0,
+ {
+ { -3, 193, 37, 35, 236, 208, 0 },
+ { 2394, 97, 90, 36, 224, 232, 2 },
+ { 3, 65, 40, 35, 236, 204, 2 },
+ { 1203, 97, 89, 33, 224, 240, 0 }
+ }
+ },
+ { /* FM region 53 */
+ { 37, 0, 127 }, 128, 128, 122, 0,
+ {
+ { 0, 193, 21, 34, 236, 188, 0 },
+ { 3, 97, 74, 36, 224, 248, 2 },
+ { 1906, 251, 24, 32, 96, 192, 3 },
+ { 1200, 97, 73, 32, 224, 184, 0 }
+ }
+ },
+ { /* FM region 54 */
+ { 34, 0, 127 }, 64, 133, 135, 0,
+ {
+ { 0, 194, 25, 35, 120, 200, 2 },
+ { 0, 97, 75, 36, 224, 240, 0 },
+ { 2906, 254, 28, 48, 0, 184, 3 },
+ { 0, 216, 75, 80, 204, 240, 2 }
+ }
+ },
+ { /* FM region 55 */
+ { 41, 0, 127 }, 208, 64, 255, 0,
+ {
+ { 475, 249, 16, 32, 252, 240, 2 },
+ { 702, 248, 71, 32, 0, 244, 2 },
+ { 1136, 232, 27, 32, 216, 248, 0 },
+ { 0, 249, 23, 48, 0, 248, 2 }
+ }
+ },
+ { /* FM region 56 */
+ { 37, 0, 127 }, 0, 132, 233, 0,
+ {
+ { 0, 195, 95, 64, 240, 208, 0 },
+ { 0, 225, 94, 64, 248, 240, 0 },
+ { 0, 254, 127, 0, 4, 196, 4 },
+ { 1902, 228, 95, 1, 248, 200, 0 }
+ }
+ },
+ { /* FM region 57 */
+ { 37, 0, 127 }, 16, 140, 238, 0,
+ {
+ { 0, 163, 90, 67, 228, 208, 0 },
+ { 0, 209, 77, 65, 248, 240, 0 },
+ { 1969, 173, 58, 65, 0, 176, 0 },
+ { 0, 210, 61, 52, 204, 220, 0 }
+ }
+ },
+ { /* FM region 58 */
+ { 37, 0, 127 }, 16, 140, 222, 0,
+ {
+ { 0, 119, 74, 67, 160, 212, 0 },
+ { 0, 146, 61, 65, 248, 244, 0 },
+ { 1900, 137, 58, 65, 100, 196, 0 },
+ { 0, 119, 61, 52, 120, 200, 0 }
+ }
+ },
+ { /* FM region 59 */
+ { 37, 0, 127 }, 16, 135, 219, 0,
+ {
+ { 0, 176, 79, 69, 240, 216, 0 },
+ { 0, 193, 79, 64, 248, 236, 0 },
+ { 0, 178, 123, 54, 92, 228, 0 },
+ { 3369, 212, 95, 38, 144, 212, 0 }
+ }
+ },
+ { /* FM region 60 */
+ { 34, 0, 127 }, 0, 119, 203, 0,
+ {
+ { 2, 65, 77, 66, 228, 204, 0 },
+ { 2, 161, 74, 64, 240, 240, 0 },
+ { -2, 85, 60, 66, 180, 216, 2 },
+ { -2, 162, 74, 64, 220, 240, 2 }
+ }
+ },
+ { /* FM region 61 */
+ { 34, 0, 127 }, 16, 154, 237, 0,
+ {
+ { 0, 179, 42, 64, 216, 208, 0 },
+ { 0, 209, 61, 64, 248, 244, 0 },
+ { -1200, 226, 55, 65, 244, 220, 2 },
+ { 1902, 162, 62, 52, 204, 236, 2 }
+ }
+ },
+ { /* FM region 62 */
+ { 34, 0, 127 }, 48, 119, 221, 0,
+ {
+ { 2, 119, 79, 64, 208, 212, 0 },
+ { 2, 209, 110, 64, 248, 236, 0 },
+ { -2, 84, 79, 64, 136, 212, 2 },
+ { -2, 209, 110, 64, 240, 240, 2 }
+ }
+ },
+ { /* FM region 63 */
+ { 34, 0, 127 }, 32, 135, 221, 0,
+ {
+ { 2, 165, 79, 64, 152, 216, 0 },
+ { 2, 225, 110, 64, 248, 236, 0 },
+ { -2, 132, 79, 64, 72, 224, 2 },
+ { -2, 241, 110, 64, 252, 236, 2 }
+ }
+ },
+ { /* FM region 64 */
+ { 37, 0, 127 }, 17, 127, 190, 0,
+ {
+ { 0, 209, 60, 67, 244, 208, 0 },
+ { 1200, 145, 94, 65, 248, 244, 2 },
+ { 3369, 197, 47, 4, 128, 192, 0 },
+ { 1902, 167, 94, 6, 200, 200, 0 }
+ }
+ },
+ { /* FM region 65 */
+ { 37, 0, 127 }, 17, 143, 190, 0,
+ {
+ { 0, 209, 60, 67, 244, 216, 0 },
+ { 1902, 145, 62, 65, 248, 240, 2 },
+ { 3369, 197, 47, 4, 128, 196, 0 },
+ { 2400, 167, 94, 6, 200, 212, 2 }
+ }
+ },
+ { /* FM region 66 */
+ { 37, 0, 127 }, 17, 143, 190, 0,
+ {
+ { 0, 209, 60, 67, 244, 208, 0 },
+ { 1902, 145, 62, 65, 248, 240, 2 },
+ { 3369, 197, 47, 4, 128, 192, 0 },
+ { 1902, 167, 94, 6, 200, 216, 2 }
+ }
+ },
+ { /* FM region 67 */
+ { 37, 0, 127 }, 17, 125, 190, 0,
+ {
+ { 0, 114, 109, 67, 244, 224, 0 },
+ { 1902, 166, 93, 97, 200, 240, 0 },
+ { 2786, 165, 95, 52, 160, 200, 0 },
+ { 2400, 173, 78, 54, 240, 212, 2 }
+ }
+ },
+ { /* FM region 68 */
+ { 34, 0, 127 }, 16, 140, 205, 0,
+ {
+ { 0, 211, 55, 66, 244, 208, 0 },
+ { 1902, 193, 93, 65, 248, 240, 0 },
+ { 0, 204, 47, 4, 244, 216, 0 },
+ { 3600, 183, 95, 6, 160, 232, 0 }
+ }
+ },
+ { /* FM region 69 */
+ { 34, 0, 127 }, 16, 126, 222, 0,
+ {
+ { 0, 243, 36, 66, 172, 200, 0 },
+ { 1200, 193, 110, 67, 248, 244, 0 },
+ { 0, 215, 33, 2, 232, 212, 0 },
+ { 3369, 178, 63, 6, 184, 240, 0 }
+ }
+ },
+ { /* FM region 70 */
+ { 34, 0, 127 }, 16, 140, 221, 0,
+ {
+ { 1200, 213, 61, 66, 136, 200, 0 },
+ { 1902, 193, 93, 68, 248, 240, 0 },
+ { 0, 197, 47, 2, 228, 216, 0 },
+ { 3369, 183, 95, 2, 160, 236, 0 }
+ }
+ },
+ { /* FM region 71 */
+ { 34, 0, 127 }, 16, 124, 201, 0,
+ {
+ { 1200, 195, 55, 68, 240, 208, 0 },
+ { 0, 209, 76, 65, 248, 236, 0 },
+ { 1902, 147, 47, 19, 208, 212, 0 },
+ { 0, 183, 79, 22, 156, 228, 0 }
+ }
+ },
+ { /* FM region 72 */
+ { 37, 0, 127 }, 32, 110, 234, 0,
+ {
+ { 500, 237, 60, 68, 0, 192, 1 },
+ { 1, 161, 93, 65, 248, 240, 2 },
+ { 3365, 154, 47, 16, 48, 180, 6 },
+ { 1200, 165, 92, 52, 160, 212, 2 }
+ }
+ },
+ { /* FM region 73 */
+ { 37, 0, 127 }, 32, 142, 200, 0,
+ {
+ { 0, 193, 60, 68, 248, 200, 0 },
+ { 1, 129, 61, 65, 248, 240, 2 },
+ { 3365, 154, 47, 16, 68, 184, 6 },
+ { 1200, 169, 92, 52, 160, 204, 2 }
+ }
+ },
+ { /* FM region 74 */
+ { 35, 0, 127 }, 32, 135, 36, 0,
+ {
+ { 1199, 165, 79, 66, 152, 192, 2 },
+ { -3, 145, 110, 64, 248, 240, 2 },
+ { 0, 199, 79, 66, 44, 236, 2 },
+ { 2986, 136, 110, 67, 100, 196, 2 }
+ }
+ },
+ { /* FM region 75 */
+ { 37, 0, 127 }, 32, 190, 71, 0,
+ {
+ { 868, 202, 140, 16, 24, 188, 2 },
+ { 0, 176, 77, 65, 248, 240, 2 },
+ { 3750, 169, 127, 16, 36, 228, 6 },
+ { 2400, 195, 60, 17, 232, 172, 2 }
+ }
+ },
+ { /* FM region 76 */
+ { 37, 0, 127 }, 224, 16, 123, 0,
+ {
+ { 275, 202, 14, 2, 44, 196, 2 },
+ { 0, 165, 89, 65, 56, 244, 2 },
+ { 0, 255, 12, 2, 64, 216, 6 },
+ { 963, 169, 14, 4, 40, 196, 2 }
+ }
+ },
+ { /* FM region 77 */
+ { 50, 0, 127 }, 192, 128, 100, 0,
+ {
+ { 1500, 202, 79, 68, 76, 204, 2 },
+ { -2, 97, 26, 64, 248, 232, 2 },
+ { 1588, 202, 223, 69, 4, 220, 0 },
+ { 3, 188, 121, 67, 48, 252, 2 }
+ }
+ },
+ { /* FM region 78 */
+ { 34, 0, 127 }, 112, 140, 205, 0,
+ {
+ { 0, 68, 47, 66, 60, 176, 2 },
+ { -2, 113, 94, 64, 248, 236, 0 },
+ { 5000, 121, 47, 64, 32, 168, 7 },
+ { 3, 136, 94, 64, 0, 236, 0 }
+ }
+ },
+ { /* FM region 79 */
+ { 35, 0, 127 }, 32, 135, 33, 0,
+ {
+ { 1199, 197, 79, 66, 152, 184, 2 },
+ { 0, 161, 110, 64, 248, 240, 2 },
+ { 0, 199, 79, 66, 44, 236, 2 },
+ { 2400, 255, 110, 65, 36, 208, 6 }
+ }
+ },
+ { /* FM region 80 */
+ { 34, 0, 127 }, 0, 192, 170, 0,
+ {
+ { 1199, 192, 77, 33, 200, 212, 0 },
+ { 0, 209, 107, 33, 232, 240, 0 },
+ { 1201, 80, 77, 33, 200, 212, 0 },
+ { 0, 241, 107, 33, 232, 240, 0 }
+ }
+ },
+ { /* FM region 81 */
+ { 34, 0, 127 }, 0, 192, 221, 0,
+ {
+ { -1, 192, 45, 33, 200, 212, 0 },
+ { -1, 209, 107, 33, 232, 244, 0 },
+ { 1, 80, 45, 33, 200, 212, 0 },
+ { 1, 241, 107, 33, 232, 244, 0 }
+ }
+ },
+ { /* FM region 82 */
+ { 37, 0, 127 }, 0, 112, 255, 0,
+ {
+ { 4750, 221, 45, 34, 48, 172, 4 },
+ { -10000, 161, 107, 33, 200, 244, 3 },
+ { 2204, 137, 45, 37, 64, 184, 0 },
+ { -2, 211, 107, 33, 160, 208, 0 }
+ }
+ },
+ { /* FM region 83 */
+ { 37, 0, 127 }, 16, 127, 238, 0,
+ {
+ { 2, 248, 45, 32, 204, 208, 0 },
+ { -9500, 241, 107, 33, 200, 240, 3 },
+ { 3369, 186, 45, 38, 24, 208, 0 },
+ { -2, 211, 107, 32, 220, 212, 0 }
+ }
+ },
+ { /* FM region 84 */
+ { 37, 0, 127 }, 0, 128, 221, 0,
+ {
+ { -1, 192, 191, 99, 220, 216, 0 },
+ { 1200, 243, 92, 97, 0, 244, 0 },
+ { 3984, 200, 11, 96, 168, 192, 0 },
+ { 1, 194, 127, 98, 108, 200, 0 }
+ }
+ },
+ { /* FM region 85 */
+ { 34, 0, 127 }, 128, 128, 111, 0,
+ {
+ { 1, 194, 25, 35, 120, 204, 2 },
+ { -9750, 193, 107, 36, 224, 244, 3 },
+ { 3906, 255, 28, 50, 12, 188, 3 },
+ { -1, 216, 107, 80, 204, 240, 2 }
+ }
+ },
+ { /* FM region 86 */
+ { 34, 0, 127 }, 32, 134, 222, 0,
+ {
+ { 0, 195, 52, 33, 200, 208, 0 },
+ { 0, 177, 90, 33, 232, 240, 2 },
+ { 702, 195, 52, 33, 200, 208, 2 },
+ { 702, 177, 90, 34, 232, 240, 2 }
+ }
+ },
+ { /* FM region 87 */
+ { 34, 0, 127 }, 32, 134, 205, 0,
+ {
+ { 0, 198, 75, 36, 120, 220, 2 },
+ { 0, 225, 78, 52, 40, 244, 2 },
+ { 0, 246, 47, 32, 220, 208, 2 },
+ { 1902, 241, 124, 32, 240, 236, 2 }
+ }
+ },
+ { /* FM region 88 */
+ { 35, 0, 127 }, 32, 120, 14, 0,
+ {
+ { 3600, 244, 67, 34, 88, 208, 0 },
+ { 3, 194, 84, 33, 84, 240, 2 },
+ { -3, 194, 84, 33, 172, 236, 2 },
+ { 902, 254, 114, 34, 0, 224, 3 }
+ }
+ },
+ { /* FM region 89 */
+ { 34, 0, 127 }, 64, 169, 170, 0,
+ {
+ { -3, 83, 69, 34, 184, 212, 0 },
+ { -7500, 50, 69, 33, 176, 244, 3 },
+ { 3, 81, 69, 34, 212, 212, 2 },
+ { -8500, 66, 69, 33, 176, 244, 3 }
+ }
+ },
+ { /* FM region 90 */
+ { 34, 0, 127 }, 64, 120, 221, 0,
+ {
+ { -2, 82, 69, 34, 244, 216, 0 },
+ { 0, 145, 102, 33, 228, 240, 0 },
+ { 2, 81, 69, 34, 244, 208, 2 },
+ { 0, 145, 102, 33, 224, 240, 2 }
+ }
+ },
+ { /* FM region 91 */
+ { 35, 0, 127 }, 32, 138, 14, 0,
+ {
+ { 2400, 148, 67, 34, 176, 200, 0 },
+ { 3, 194, 85, 33, 220, 236, 2 },
+ { -3, 194, 69, 33, 220, 236, 2 },
+ { 1905, 254, 114, 34, 48, 224, 2 }
+ }
+ },
+ { /* FM region 92 */
+ { 34, 0, 127 }, 82, 67, 71, 0,
+ {
+ { 2982, 228, 22, 146, 88, 192, 3 },
+ { 3, 102, 84, 146, 196, 240, 2 },
+ { 3358, 50, 149, 116, 144, 208, 2 },
+ { -3, 85, 84, 146, 120, 240, 0 }
+ }
+ },
+ { /* FM region 93 */
+ { 37, 0, 127 }, 48, 126, 219, 0,
+ {
+ { -3, 49, 19, 33, 120, 200, 0 },
+ { 0, 81, 70, 33, 220, 240, 0 },
+ { 3804, 242, 18, 50, 200, 200, 0 },
+ { 1203, 82, 19, 82, 200, 176, 0 }
+ }
+ },
+ { /* FM region 94 */
+ { 35, 0, 127 }, 32, 138, 13, 0,
+ {
+ { 2786, 116, 67, 34, 204, 184, 0 },
+ { 1902, 114, 69, 33, 192, 232, 2 },
+ { -3, 178, 69, 33, 188, 232, 2 },
+ { 3804, 254, 82, 34, 164, 228, 2 }
+ }
+ },
+ { /* FM region 95 */
+ { 34, 0, 127 }, 48, 135, 238, 0,
+ {
+ { -2, 34, 85, 34, 184, 224, 0 },
+ { 1, 113, 70, 33, 228, 236, 0 },
+ { 2, 19, 85, 34, 156, 224, 2 },
+ { -1, 129, 70, 33, 224, 236, 2 }
+ }
+ },
+ { /* FM region 96 */
+ { 50, 0, 127 }, 240, 112, 221, 0,
+ {
+ { 3369, 213, 69, 32, 0, 204, 0 },
+ { 0, 193, 70, 33, 112, 232, 2 },
+ { 0, 145, 69, 34, 244, 208, 2 },
+ { -9000, 145, 70, 33, 224, 236, 3 }
+ }
+ },
+ { /* FM region 97 */
+ { 34, 0, 127 }, 96, 122, 168, 0,
+ {
+ { -1, 99, 51, 33, 200, 208, 0 },
+ { -8500, 81, 83, 33, 232, 240, 3 },
+ { 702, 99, 52, 33, 200, 208, 2 },
+ { -9500, 65, 83, 34, 224, 240, 3 }
+ }
+ },
+ { /* FM region 98 */
+ { 34, 0, 127 }, 0, 67, 0, 0,
+ {
+ { 1500, 217, 55, 151, 20, 224, 2 },
+ { 3, 231, 70, 146, 88, 220, 2 },
+ { 2369, 115, 148, 151, 32, 196, 2 },
+ { -3, 118, 36, 146, 64, 244, 2 }
+ }
+ },
+ { /* FM region 99 */
+ { 34, 0, 127 }, 64, 169, 204, 0,
+ {
+ { -3, 228, 69, 34, 148, 220, 0 },
+ { -7448, 243, 69, 33, 200, 240, 3 },
+ { 3, 81, 68, 34, 212, 212, 2 },
+ { -8526, 65, 68, 33, 196, 240, 3 }
+ }
+ },
+ { /* FM region 100 */
+ { 34, 0, 127 }, 64, 119, 187, 0,
+ {
+ { 2786, 228, 22, 146, 176, 192, 0 },
+ { 3, 102, 68, 146, 196, 236, 2 },
+ { 3369, 178, 149, 116, 176, 208, 2 },
+ { -3, 231, 68, 146, 120, 240, 0 }
+ }
+ },
+ { /* FM region 101 */
+ { 34, 0, 127 }, 240, 144, 239, 0,
+ {
+ { -2, 49, 69, 34, 236, 208, 2 },
+ { -9000, 113, 102, 33, 228, 236, 3 },
+ { 2400, 149, 69, 34, 12, 216, 1 },
+ { 0, 145, 102, 33, 224, 236, 2 }
+ }
+ },
+ { /* FM region 102 */
+ { 50, 0, 127 }, 241, 176, 6, 0,
+ {
+ { 1200, 247, 49, 64, 252, 204, 0 },
+ { 3804, 246, 101, 32, 0, 232, 2 },
+ { 1902, 247, 32, 32, 112, 188, 2 },
+ { 0, 228, 84, 32, 0, 240, 2 }
+ }
+ },
+ { /* FM region 103 */
+ { 37, 0, 127 }, 64, 101, 221, 0,
+ {
+ { 1, 194, 68, 97, 196, 200, 2 },
+ { -10001, 247, 100, 114, 176, 240, 3 },
+ { 3370, 213, 33, 70, 52, 200, 2 },
+ { -1, 178, 68, 49, 208, 212, 0 }
+ }
+ },
+ { /* FM region 104 */
+ { 34, 0, 127 }, 0, 255, 203, 0,
+ {
+ { -3, 245, 82, 99, 200, 232, 2 },
+ { 2787, 244, 84, 96, 0, 236, 2 },
+ { 1198, 133, 81, 100, 196, 220, 2 },
+ { 1902, 147, 67, 80, 0, 232, 2 }
+ }
+ },
+ { /* FM region 105 */
+ { 37, 0, 127 }, 0, 255, 140, 0,
+ {
+ { 500, 255, 137, 179, 0, 200, 3 },
+ { 1902, 248, 90, 160, 0, 244, 2 },
+ { 3804, 245, 57, 35, 164, 204, 2 },
+ { 0, 245, 38, 51, 196, 208, 2 }
+ }
+ },
+ { /* FM region 106 */
+ { 37, 0, 127 }, 0, 255, 72, 0,
+ {
+ { 1000, 238, 57, 65, 0, 188, 3 },
+ { 1902, 247, 103, 112, 0, 244, 2 },
+ { 2786, 250, 36, 81, 68, 212, 2 },
+ { 0, 249, 50, 49, 172, 204, 2 }
+ }
+ },
+ { /* FM region 107 */
+ { 37, 0, 127 }, 16, 119, 72, 0,
+ {
+ { 1500, 255, 89, 65, 0, 196, 3 },
+ { 2790, 246, 39, 112, 0, 240, 0 },
+ { 1905, 246, 36, 81, 168, 208, 0 },
+ { 0, 249, 114, 49, 172, 212, 0 }
+ }
+ },
+ { /* FM region 108 */
+ { 37, 0, 127 }, 0, 255, 237, 0,
+ {
+ { 1902, 254, 89, 65, 0, 212, 2 },
+ { 0, 248, 87, 112, 0, 240, 2 },
+ { 3369, 231, 62, 81, 0, 208, 2 },
+ { 3, 245, 118, 49, 96, 196, 2 }
+ }
+ },
+ { /* FM region 109 */
+ { 34, 0, 127 }, 16, 188, 205, 0,
+ {
+ { -2, 179, 47, 50, 244, 224, 2 },
+ { 1900, 145, 94, 49, 248, 232, 2 },
+ { 3, 210, 46, 2, 244, 208, 2 },
+ { 2789, 133, 93, 4, 180, 244, 2 }
+ }
+ },
+ { /* FM region 110 */
+ { 37, 0, 127 }, 48, 135, 220, 0,
+ {
+ { 1901, 162, 25, 35, 144, 208, 0 },
+ { 0, 113, 105, 65, 220, 240, 0 },
+ { 3369, 233, 88, 51, 120, 212, 0 },
+ { 0, 229, 24, 84, 200, 208, 0 }
+ }
+ },
+ { /* FM region 111 */
+ { 34, 0, 127 }, 112, 32, 190, 0,
+ {
+ { 0, 53, 79, 66, 152, 212, 2 },
+ { 1200, 53, 75, 64, 136, 244, 2 },
+ { 500, 149, 60, 66, 16, 208, 2 },
+ { 1902, 200, 78, 64, 0, 248, 0 }
+ }
+ },
+ { /* FM region 112 */
+ { 37, 0, 127 }, 0, 144, 130, 0,
+ {
+ { 2514, 255, 68, 53, 0, 204, 2 },
+ { 2400, 247, 133, 48, 0, 240, 2 },
+ { 4151, 243, 67, 50, 0, 212, 2 },
+ { 3369, 243, 66, 56, 0, 204, 2 }
+ }
+ },
+ { /* FM region 113 */
+ { 37, 0, 127 }, 0, 255, 0, 0,
+ {
+ { 514, 253, 79, 51, 0, 196, 3 },
+ { 1905, 252, 89, 51, 0, 244, 2 },
+ { 4349, 245, 35, 51, 0, 208, 2 },
+ { 1205, 247, 34, 51, 0, 208, 2 }
+ }
+ },
+ { /* FM region 114 */
+ { 37, 0, 127 }, 0, 255, 0, 0,
+ {
+ { 514, 221, 69, 35, 0, 204, 3 },
+ { 0, 250, 86, 115, 0, 252, 2 },
+ { 1884, 244, 116, 51, 0, 200, 2 },
+ { 1208, 210, 35, 51, 0, 208, 2 }
+ }
+ },
+ { /* FM region 115 */
+ { 37, 0, 127 }, 0, 255, 16, 0,
+ {
+ { 514, 222, 85, 163, 0, 192, 3 },
+ { 0, 254, 108, 163, 0, 252, 2 },
+ { 3800, 255, 143, 160, 0, 176, 2 },
+ { 1200, 250, 105, 163, 0, 212, 2 }
+ }
+ },
+ { /* FM region 116 */
+ { 37, 0, 127 }, 0, 255, 16, 0,
+ {
+ { 1514, 249, 101, 163, 0, 204, 3 },
+ { -1200, 249, 87, 160, 0, 252, 2 },
+ { 0, 235, 143, 160, 0, 204, 2 },
+ { 1200, 234, 73, 163, 0, 204, 2 }
+ }
+ },
+ { /* FM region 117 */
+ { 37, 0, 127 }, 0, 255, 16, 0,
+ {
+ { 500, 239, 101, 160, 0, 204, 3 },
+ { -1195, 248, 104, 160, 0, 252, 2 },
+ { 1898, 252, 72, 163, 0, 216, 2 },
+ { 1239, 248, 87, 163, 0, 196, 2 }
+ }
+ },
+ { /* FM region 118 */
+ { 37, 0, 127 }, 0, 255, 255, 0,
+ {
+ { 500, 255, 98, 160, 0, 196, 3 },
+ { -1, 249, 105, 160, 0, 252, 2 },
+ { 1907, 250, 71, 160, 0, 252, 2 },
+ { 1182, 249, 87, 161, 0, 192, 2 }
+ }
+ },
+ { /* FM region 119 */
+ { 37, 0, 127 }, 0, 0, 100, 0,
+ {
+ { 600, 32, 15, 0, 252, 224, 6 },
+ { 0, 47, 111, 65, 0, 244, 2 },
+ { 1826, 16, 47, 0, 252, 216, 2 },
+ { 3551, 240, 47, 0, 252, 212, 2 }
+ }
+ },
+ { /* FM region 120 */
+ { 52, 0, 127 }, 240, 128, 235, 0,
+ {
+ { 1228, 161, 47, 17, 196, 200, 3 },
+ { 3000, 123, 75, 17, 0, 240, 2 },
+ { 7022, 72, 43, 17, 0, 216, 0 },
+ { 4000, 150, 79, 17, 48, 196, 3 }
+ }
+ },
+ { /* FM region 121 */
+ { 37, 0, 127 }, 224, 16, 86, 0,
+ {
+ { 275, 251, 6, 0, 36, 200, 2 },
+ { 0, 101, 104, 65, 56, 240, 2 },
+ { 0, 240, 6, 0, 252, 208, 6 },
+ { 1000, 195, 8, 0, 248, 200, 2 }
+ }
+ },
+ { /* FM region 122 */
+ { 34, 0, 127 }, 0, 0, 185, 0,
+ {
+ { 600, 35, 66, 17, 72, 224, 4 },
+ { -13000, 81, 67, 17, 228, 244, 2 },
+ { 702, 97, 38, 17, 212, 196, 6 },
+ { -14000, 81, 65, 17, 224, 244, 3 }
+ }
+ },
+ { /* FM region 123 */
+ { 50, 0, 127 }, 240, 112, 237, 0,
+ {
+ { -6528, 153, 127, 16, 0, 252, 3 },
+ { 1200, 105, 109, 16, 0, 216, 2 },
+ { -6022, 179, 139, 17, 0, 248, 3 },
+ { 2000, 104, 79, 17, 0, 240, 0 }
+ }
+ },
+ { /* FM region 124 */
+ { 50, 0, 127 }, 240, 240, 16, 0,
+ {
+ { 1914, 240, 64, 160, 240, 208, 2 },
+ { 1200, 240, 73, 163, 240, 244, 0 },
+ { 1900, 240, 64, 160, 240, 148, 2 },
+ { 4151, 240, 73, 163, 240, 244, 0 }
+ }
+ },
+ { /* FM region 125 */
+ { 34, 0, 127 }, 240, 56, 235, 0,
+ {
+ { -5522, 97, 32, 17, 196, 240, 3 },
+ { 0, 84, 75, 17, 180, 248, 3 },
+ { 702, 65, 38, 17, 224, 212, 6 },
+ { -4000, 161, 73, 17, 224, 252, 1 }
+ }
+ },
+ { /* FM region 126 */
+ { 53, 0, 127 }, 240, 248, 37, 0,
+ {
+ { 1050, 243, 0, 0, 252, 224, 7 },
+ { 2000, 49, 68, 0, 224, 236, 3 },
+ { 350, 240, 0, 0, 252, 216, 1 },
+ { 700, 240, 0, 0, 252, 212, 3 }
+ }
+ },
+ { /* FM region 127 */
+ { 53, 0, 127 }, 240, 248, 37, 0,
+ {
+ { 1050, 245, 85, 0, 0, 244, 7 },
+ { -5000, 247, 71, 0, 0, 252, 3 },
+ { 350, 240, 0, 0, 0, 164, 0 },
+ { 700, 32, 0, 0, 0, 252, 2 }
+ }
+ }}; /* end FM Regions */
+
+/*----------------------------------------------------------------------------
+ * Programs
+ *----------------------------------------------------------------------------
+*/
+const S_PROGRAM eas_programs[] =
+{
+ { 7864320, 0 } /* program 0 */
+}; /* end Programs */
+
+/*----------------------------------------------------------------------------
+ * Banks
+ *----------------------------------------------------------------------------
+*/
+const S_BANK eas_banks[] =
+{
+ { /* bank 0 */
+ 30976,
+ {
+ 32768, 32769, 32770, 32771, 32772, 32773, 32774, 32775,
+ 32776, 32777, 32778, 32779, 32780, 32781, 32782, 32783,
+ 32784, 32785, 32786, 32787, 32788, 32789, 32790, 32791,
+ 32792, 32793, 32794, 32795, 32796, 32797, 32798, 32799,
+ 32800, 32801, 32802, 32803, 32804, 32805, 32806, 32807,
+ 32808, 32809, 32810, 32811, 32812, 32813, 32814, 32815,
+ 32816, 32817, 32818, 32819, 32820, 32821, 32822, 32823,
+ 32824, 32825, 32826, 32827, 32828, 32829, 32830, 32831,
+ 32832, 32833, 32834, 32835, 32836, 32837, 32838, 32839,
+ 32840, 32841, 32842, 32843, 32844, 32845, 32846, 32847,
+ 32848, 32849, 32850, 32851, 32852, 32853, 32854, 32855,
+ 32856, 32857, 32858, 32859, 32860, 32861, 32862, 32863,
+ 32864, 32865, 32866, 32867, 32868, 32869, 32870, 32871,
+ 32872, 32873, 32874, 32875, 32876, 32877, 32878, 32879,
+ 32880, 32881, 32882, 32883, 32884, 32885, 32886, 32887,
+ 32888, 32889, 32890, 32891, 32892, 32893, 32894, 32895
+ }
+ }
+}; /* end Banks */
+
+/*----------------------------------------------------------------------------
+ * Samples
+ *----------------------------------------------------------------------------
+*/
+
+const EAS_SAMPLE eas_samples[] =
+{
+ 13, -24, 28, -32, 33, -37, 39, -61, 119, -76, 120, -70, 99, -122, 89, -113,
+ 91, -123, 122, -123, 77, 86, -116, 6, -118, 123, -23, 64, -93, 17, 24, -125,
+ 124, -125, 124, -24, -12, 56, 87, -54, 38, -91, 64, -2, -41, 126, -127, 20,
+ 8, -48, -62, 127, -128, 88, -43, -18, 86, -100, 44, -32, -26, 71, -13, 6,
+ 51, -33, -50, 106, -59, 33, 5, -20, 69, -56, 54, -48, -6, 38, -5, -38,
+ 35, -1, 12, -1, 4, 23, -56, 19, -7, 5, -4, 31, -38, 3, -12, -9,
+ -25, -7, 24, -32, 17, -21, -22, 24, -20, 6, -18, 0, -15, 5, -16, 15,
+ 3, 2, -10, 3, 27, -31, 37, -12, 32, -5, 2, -21, 27, -14, 20, -8,
+ 5, 15, -7, -11, 20, -37, 7, -23, 17, -22, 11, -14, 8, -44, 39, -52,
+ 26, 3, 14, -10, 51, -45, 29, -19, 19, -3, 41, -21, 46, -23, 10, 16,
+ -25, 3, 7, -30, 19, -9, -5, -1, 1, -19, 1, -24, -23, 51, -48, -10,
+ 17, -34, -38, 60, -71, 26, -43, 65, -59, 70, -26, 61, -51, 63, -51, 42,
+ -25, 58, -36, 31, 10, -14, -9, 8, -18, 8, 4, 8, 7, 31, -52, 39,
+ -53, 53, -56, 24, 11, 1, -20, -8, -32, -18, -10, -25, 4, 1, 37, -37,
+ 42, -42, 53, -57, 52, -18, 30, 11, 20, -52, 55, -61, 56, -50, 24, -26,
+ 38, -42, 36, -1, -43, 58, -46, 6, 18, -35, 31, -36, 28, -55, 55, -98,
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+ 13, -19, 1, 20, -28, 13, -6, 0, -17, 33, -48, 38, -16, -20, 53, -75,
+ 75, -61, 30, -12, -9, 2, 5, -9, -7, 15, -17, 9, 4, -46, 70, -85,
+ 86, -67, 29, 18, -60, 76, -70, 29, -8, -1, 1, 14, -58, 93, -118, 127,
+ -125, 85, -36, -2, 17, -39, 46, -61, 70, -99, 122, -120, 56, 15, -85, 116,
+ -101, 30, 43, -108, 127, -122, 72, -38, -3, 20, -31, 29, -37, 42, -50, 63,
+ -72, 40, -7, -46, 66, -82, 53, -25, 1, 1, -1, -3, 3, -24, 24, -34,
+ 27, -24, 36, -49, 37, -38, 35, -42, 13, -4, -14, 14, -29, 42, -58, 58,
+ -60, 41, -39, 27, -35, 27, -32, 25, -37, 43, -50, 32, -20, -4, 4, -14,
+ 16, -43, 66, -93, 79, -58, 27, -11, -21, 34, -40, 24, -7, -25, 31, -29,
+ 19, -18, -2, 1, -6, 4, -31, 47, -63, 62, -64, 32, 2, -29, 23, -21,
+ 19, -41, 59, -76, 62, -52, 26, -11, -1, -10, 6, -10, -7, 17, -42, 34,
+ -12, -34, 73, -106, 92, -61, 26, -24, 8, -7, -2, 3, -24, 31, -44, 27,
+ -9, -22, 38, -43, 15, -4, -3, -2, -12, 5, 1, -7, 7, -25, 17, -17,
+ 4, -15, 27, -40, 34, -18, 6, -13, 12, -3, -9, -6, 8, -18, 18, -36,
+ 54, -55, 30, -25, 26, -41, 38, -49, 48, -38, 8, 16, -40, 47, -52, 34,
+ -12, -4, 2, -13, 0, 1, 6, -11, 19, -25, 24, -15, 11, -6, 1, 9,
+ -6, -3, 11, -18, 28, -23, 8, 6, -2, -1, 0, -5, 18, -23, 24, -28,
+ 27, -12, -1, -2, 6, 4, -14, 19, -29, 42, -40, 22, -6, -3, 14, -16,
+ 4, 8, -9, 13, -22, 17, 5, -24, 37, -34, 20, -4, -11, 19, -20, 21,
+ -16, 1, 17, -17, 5, -9, 25, -31, 31, -40, 32, -11, -7, 4, -4, 13,
+ -23, 29, -40, 51, -49, 20, 3, -20, 38, -47, 34, -21, 14, -9, 4, -13,
+ 20, -29, 37, -57, 59, -48, 23, -5, -7, 5, -6, -5, 13, -17, 9, -4,
+ -12, 22, -19, -8, 10, -5, -7, 8, -18, 19, -13, 5, -12, 6, -10, 18,
+ -40, 66, -91, 82, -56, 17, 19, -51, 56, -45, 30, -19, 5, -6, 7, -9,
+ -4, -2, -5, 13, -19, 1, 20, -28, 13, -6, 0, -17, 33, -48, 38, -16,
+ -20, 53, -75, 75, -61, 30, -12, -9, 2, 5, -9, -7, 15, -17, 9, 4,
+ -46, 70, -85, 86, -67, 29, 18, -60, 76, -70, 29, -8, -1, 1, 14, -58,
+ 93, -118, 127, -125, 85, -36, -2, 17, -39, 46, -61, 70, -99, 122, -120, 56,
+ 15, -85, 116, -101, 30, 43, -108, 127, -122, 72, -38, -3, 20, -31, 29, -37,
+ 42, -50, 63, -72, 40, -7, -46, 66, -82, 53, -25, 1, 1, -1, -3, 3,
+ -24, 24, -34, 27, -24, 36, -49, 37, -38, 35, -42, 13, -4, -14, 14, -29,
+ 42, -58, 58, -60, 41, -39, 27, -35, 27, -32, 25, -37, 43, -50, 32, -20,
+ -4, 4, -14, 16, -43, 66, -93, 79, -58, 27, -11, -21, 34, -40, 24, -7,
+ -25, 31, -29, 19, -18, -2, 1, -6, 4, -31, 47, -63, 62, -64, 32, 2,
+ -29, 23, -21, 19, -41, 59, -76, 62, -52, 26, -11, -1, -10, 6, -10, -7,
+ 17, -42, 34, -12, -34, 73, -106, 92, -61, 26, -24, 8, -7, -2, 3, -24,
+ 31, -44, 27, -9, -22, 38, -43, 15, -4, -3, -2, -12, 5, 1, -7, 7,
+ -25, 17, -17, 4, -15, 27, -40, 34, -18, 6, -13, 12, -3, -9, -6, 8,
+ -18, 18, -36, 54, -55, 30, -25, 26, -41, 38, -49, 48, -38, 8, 16, -40,
+ 47, -52, 34, -12, -4, 2, -13, -1, -9, -21, -32, -43, -55, -58, -40, -8,
+ 15, 29, 38, 44, 59, 63, 51, 20, -20, -47, -61, -58, -48, -42, -43, -35,
+ -15, 15, 41, 55, 61, 62, 70, 67, 45, 11, -31, -62, -80, -78, -71, -62,
+ -54, -39, -15, 18, 50, 68, 79, 83, 92, 88, 57, 16, -31, -71, -94, -95,
+ -90, -83, -71, -55, -28, 7, 49, 77, 93, 100, 109, 107, 78, 36, -20, -71,
+ -99, -104, -100, -92, -83, -69, -43, -5, 42, 79, 100, 108, 114, 112, 90, 51,
+ -9, -65, -100, -110, -105, -95, -87, -76, -52, -11, 35, 75, 100, 107, 111, 106,
+ 90, 52, -4, -60, -97, -109, -99, -87, -80, -65, -35, 5, 43, 75, 91, 95,
+ 93, 87, 71, 31, -19, -64, -93, -99, -88, -76, -66, -44, -9, 28, 59, 79,
+ 84, 80, 71, 59, 35, -5, -45, -76, -96, -95, -77, -59, -39, -9, 28, 61,
+ 86, 98, 93, 74, 52, 28, -6, -43, -72, -94, -108, -99, -77, -52, -20, 19,
+ 56, 84, 106, 115, 106, 79, 45, 3, -40, -74, -93, -108, -112, -100, -79, -46,
+ -1, 45, 78, 100, 113, 118, 108, 81, 39, -19, -67, -93, -103, -108, -104, -94,
+ -74, -32, 19, 61, 88, 102, 108, 109, 100, 73, 21, -44, -87, -102, -104, -97,
+ -90, -78, -49, 0, 46, 78, 90, 92, 89, 87, 77, 43, -15, -72, -101, -104,
+ -93, -79, -69, -47, -3, 47, 84, 96, 89, 74, 65, 58, 37, -7, -61, -99,
+ -110, -101, -82, -62, -38, 2, 49, 92, 112, 106, 82, 59, 41, 18, -19, -61,
+ -95, -111, -110, -98, -74, -39, 6, 54, 99, 122, 123, 104, 79, 50, 12, -32,
+ -72, -100, -111, -110, -106, -90, -54, -4, 49, 99, 123, 126, 113, 95, 69, 24,
+ -27, -72, -102, -114, -109, -104, -94, -64, -16, 36, 87, 116, 121, 111, 97, 76,
+ 36, -15, -61, -94, -110, -108, -98, -86, -60, -19, 29, 77, 104, 109, 99, 86,
+ 67, 35, -7, -47, -82, -98, -101, -92, -77, -48, -14, 25, 67, 92, 95, 83,
+ 66, 45, 18, -9, -37, -62, -77, -81, -78, -54, -16, 0, 0, 0, 12, -10,
+ 16, -14, -18, 17, -31, -2, 3, -19, -8, 17, 28, 37, 31, -1, -25, -16,
+ -47, -13, -43, -22, 6, 34, 41, 49, 41, 27, 6, -33, -62, -54, -59, -32,
+ -13, 29, 53, 73, 60, 54, 16, -19, -58, -77, -89, -42, -46, 17, 55, 78,
+ 86, 81, 33, -3, -35, -84, -110, -75, -70, 0, 39, 78, 92, 105, 70, 13,
+ -15, -82, -107, -103, -84, -41, 20, 81, 85, 127, 86, 39, -4, -63, -103, -99,
+ -107, -69, 2, 68, 82, 127, 84, 58, 28, -44, -88, -104, -108, -83, -13, 29,
+ 69, 116, 87, 77, 40, -16, -65, -89, -104, -89, -40, -1, 47, 96, 90, 85,
+ 52, 9, -38, -62, -96, -95, -62, -22, 21, 73, 78, 86, 65, 35, -15, -41,
+ -80, -92, -69, -48, 1, 43, 65, 80, 73, 49, 8, -23, -59, -75, -74, -60,
+ -22, 17, 49, 69, 65, 60, 25, -1, -36, -59, -70, -61, -40, -9, 30, 52,
+ 57, 63, 37, 17, -12, -41, -61, -57, -50, -30, 9, 30, 49, 60, 47, 28,
+ 6, -20, -49, -51, -53, -42, -11, 13, 34, 53, 50, 34, 18, -3, -35, -39,
+ -52, -49, -23, 0, 18, 41, 47, 36, 31, 10, -21, -28, -43, -49, -31, -13,
+ 2, 28, 39, 35, 38, 19, -7, -18, -33, -42, -36, -26, -12, 19, 30, 34,
+ 36, 22, 6, -5, -25, -38, -36, -29, -17, 8, 19, 26, 35, 28, 12, 3,
+ -17, -30, -29, -29, -22, -3, 9, 19, 33, 26, 14, 8, -9, -18, -22, -28,
+ -25, -9, 3, 15, 22, 20, 18, 12, 1, -11, -22, -24, -21, -13, -3, 8,
+ 11, 20, 21, 12, 5, -6, -19, -16, -17, -17, -7, 3, 8, 20, 17, 7,
+ 7, -2, -11, -13, -16, -17, -5, 2, 5, 12, 13, 8, 6, 3, -9, -11,
+ -9, -14, -5, 1, -3, 6, 11, 6, 7, 2, -6, -4, -4, -10, -6, -3,
+ -4, 5, 8, 2, 5, 4, -2, 0, -3, -9, -4, -3, -3, 2, 3, 0,
+ 6, 4, 0, -1, -2, -2, 0, -4, 8, -1, 9, 14, 53, 57, 33, -4,
+ -20, -14, -31, -71, -112, -43, 30, -22, -102, -74, 38, 90, 35, -22, 20, 99,
+ 96, 22, -42, -5, 48, 16, -74, -97, -51, 10, -21, -80, -68, 24, 77, 43,
+ -9, 20, 90, 99, 32, -28, -8, 41, 18, -63, -97, -55, -2, -18, -74, -67,
+ 11, 69, 44, -1, 17, 83, 98, 40, -20, -8, 35, 20, -55, -95, -60, -8,
+ -19, -69, -68, 3, 61, 45, 1, 15, 78, 99, 46, -14, -8, 33, 22, -48,
+ -93, -62, -13, -19, -67, -70, -5, 56, 44, 3, 13, 73, 99, 51, -10, -7,
+ 32, 25, -42, -91, -66, -15, -18, -65, -73, -11, 51, 44, 3, 9, 69, 100,
+ 55, -6, -8, 32, 30, -36, -89, -68, -17, -17, -64, -77, -18, 47, 45, 3,
+ 5, 64, 100, 60, -2, -8, 32, 34, -30, -87, -70, -19, -14, -63, -80, -25,
+ 43, 45, 3, 1, 59, 100, 66, 1, -9, 31, 39, -23, -84, -74, -20, -12,
+ -59, -84, -32, 38, 46, 3, -3, 53, 101, 71, 6, -10, 31, 43, -14, -81,
+ -76, -24, -9, -57, -86, -41, 33, 46, 5, -9, 47, 99, 77, 9, -11, 28,
+ 49, -6, -76, -79, -25, -9, -50, -91, -46, 9, -41, 2, 6, -33, 33, -4,
+ -1, 34, -93, 60, -49, 68, -21, 41, -38, -38, 27, -87, 127, -34, 38, -3,
+ -34, -2, -85, 79, -24, 4, 75, -23, 11, -70, 1, -6, -6, 63, -48, 76,
+ -71, 2, -7, -15, 46, -32, 48, -25, -5, 10, -31, 18, -23, 5, 6, 17,
+ 11, 3, 24, -46, -21, -1, -5, 31, 1, 36, -23, -11, -4, -29, 14, -5,
+ 29, 9, -2, -2, -20, -2, -15, 13, 7, 4, 15, -11, 4, -13, -1, 1,
+ -13, 15, -6, 9, 9, -2, -4, -11, -2, -11, 7, 12, 9, 5, -6, -6,
+ -16, -2, 3, 6, 13, 1, 7, -14, -3, -11, -1, 8, 0, 18, -3, -3,
+ -6, -13, 0, 4, 2, 9, 2, 2, -6, -4, -1, -8, 2, 4, 2, 7,
+ -4, 2, 1, -7, -3, -2, 0, 2, 8, -2, 2, -1, -7, -3, -1, 3,
+ 4, 3, 3, -3, -4, -4, 0, 2, 1, 5, -1, -2, 1, -4, -1, 1,
+ 0, 2, 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, 94, 127, 124, 127, 127, 114, 73, 35, -11, -52, -98, -118, -125, -128,
+ -128, -128, -128, -128, -128, -128, -124, -86, -47, 0, 40, 87, 106, 116, 119, 125,
+ 37, -7, 122, 72, -57, -71, -88, -28, 48, 127, 36, -30, -53, -126, -16, 71,
+ 92, 51, -1, -83, -128, 14, 43, 83, 86, -14, -106, -87, -7, 28, 1, 122,
+ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+ 0, 0, 0, 0
+};
+
+const EAS_U32 eas_sampleLengths[] =
+{
+ 9922, 9338, 6793, 5242, 3057, 2818, 2806, 1835,
+ 1603, 1262, 1227, 1168, 1132, 1132, 994, 907,
+ 817, 801, 583, 418, 402, 387, 387, 357,
+ 347, 212, 167, 40, 40, 32, 30
+};
+
+const EAS_U32 eas_sampleOffsets[] =
+{
+ 0x00000000, 0x000026c2, 0x00004b3c, 0x000065c5, 0x00007a3f, 0x00008630, 0x00009132, 0x00009c28,
+ 0x0000a353, 0x0000a996, 0x0000ae84, 0x0000b34f, 0x0000b7df, 0x0000bc4b, 0x0000c0b7, 0x0000c499,
+ 0x0000c824, 0x0000cb55, 0x0000ce76, 0x0000d0bd, 0x0000d25f, 0x0000d3f1, 0x0000d574, 0x0000d6f7,
+ 0x0000d85c, 0x0000d9b7, 0x0000da8b, 0x0000db32, 0x0000db5a, 0x0000db82, 0x0000dba2
+};
+
+/*----------------------------------------------------------------------------
+ * S_EAS (hybrid)
+ *----------------------------------------------------------------------------
+*/
+const S_EAS easSoundLib =
+{
+ 0x01534145,
+ 0x00105622,
+ eas_banks,
+ eas_programs,
+ eas_regions,
+ eas_articulations,
+ eas_sampleLengths,
+ eas_sampleOffsets,
+ eas_samples,
+ eas_fmRegions,
+ 1,
+ 1,
+ 61,
+ 53,
+ 31,
+ 128
+}; /* end S_EAS */
+
+/*----------------------------------------------------------------------------
+ * Statistics
+ *
+ * Number of banks: 1
+ * Number of programs: 1
+ * Number of regions: 61
+ * Number of articulations: 53
+ * Number of samples: 31
+ * Size of sample pool: 56276
+ *----------------------------------------------------------------------------
+*/
+/* end C:\Sonic\Trunk\EASLib\WTLibrary\hybrid_22khz_mcu.c */
|