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-rw-r--r--Tremolo/dsp.c306
1 files changed, 306 insertions, 0 deletions
diff --git a/Tremolo/dsp.c b/Tremolo/dsp.c
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index 0000000..affe73c
--- /dev/null
+++ b/Tremolo/dsp.c
@@ -0,0 +1,306 @@
+/********************************************************************
+ * *
+ * THIS FILE IS PART OF THE OggVorbis 'TREMOR' CODEC SOURCE CODE. *
+ * *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
+ * *
+ * THE OggVorbis 'TREMOR' SOURCE CODE IS (C) COPYRIGHT 1994-2003 *
+ * BY THE Xiph.Org FOUNDATION http://www.xiph.org/ *
+ * *
+ ********************************************************************
+
+ function: PCM data vector blocking, windowing and dis/reassembly
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include "ogg.h"
+#include "mdct.h"
+#include "ivorbiscodec.h"
+#include "codec_internal.h"
+#include "misc.h"
+#include "window_lookup.h"
+
+int vorbis_dsp_restart(vorbis_dsp_state *v){
+ if(!v)return -1;
+ {
+ vorbis_info *vi=v->vi;
+ codec_setup_info *ci;
+
+ if(!vi)return -1;
+ ci=vi->codec_setup;
+ if(!ci)return -1;
+
+ v->out_end=-1;
+ v->out_begin=-1;
+
+ v->granulepos=-1;
+ v->sequence=-1;
+ v->sample_count=-1;
+ }
+ return 0;
+}
+
+int vorbis_dsp_init(vorbis_dsp_state *v,vorbis_info *vi){
+ int i;
+
+ codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+
+ v->vi=vi;
+
+ v->work=(ogg_int32_t **)_ogg_malloc(vi->channels*sizeof(*v->work));
+ v->mdctright=(ogg_int32_t **)_ogg_malloc(vi->channels*sizeof(*v->mdctright));
+ for(i=0;i<vi->channels;i++){
+ v->work[i]=(ogg_int32_t *)_ogg_calloc(1,(ci->blocksizes[1]>>1)*
+ sizeof(*v->work[i]));
+ v->mdctright[i]=(ogg_int32_t *)_ogg_calloc(1,(ci->blocksizes[1]>>2)*
+ sizeof(*v->mdctright[i]));
+ }
+
+ v->lW=0; /* previous window size */
+ v->W=0; /* current window size */
+
+ vorbis_dsp_restart(v);
+ return 0;
+}
+
+vorbis_dsp_state *vorbis_dsp_create(vorbis_info *vi){
+ vorbis_dsp_state *v=_ogg_calloc(1,sizeof(*v));
+ vorbis_dsp_init(v,vi);
+ return v;
+}
+
+void vorbis_dsp_clear(vorbis_dsp_state *v){
+ int i;
+ if(v){
+ vorbis_info *vi=v->vi;
+
+ if(v->work){
+ for(i=0;i<vi->channels;i++)
+ if(v->work[i])_ogg_free(v->work[i]);
+ _ogg_free(v->work);
+ }
+ if(v->mdctright){
+ for(i=0;i<vi->channels;i++)
+ if(v->mdctright[i])_ogg_free(v->mdctright[i]);
+ _ogg_free(v->mdctright);
+ }
+ }
+}
+
+void vorbis_dsp_destroy(vorbis_dsp_state *v){
+ vorbis_dsp_clear(v);
+ _ogg_free(v);
+}
+
+static LOOKUP_T *_vorbis_window(int left){
+ switch(left){
+ case 32:
+ return vwin64;
+ case 64:
+ return vwin128;
+ case 128:
+ return vwin256;
+ case 256:
+ return vwin512;
+ case 512:
+ return vwin1024;
+ case 1024:
+ return vwin2048;
+ case 2048:
+ return vwin4096;
+#ifndef LIMIT_TO_64kHz
+ case 4096:
+ return vwin8192;
+#endif
+ default:
+ return(0);
+ }
+}
+
+/* pcm==0 indicates we just want the pending samples, no more */
+int vorbis_dsp_pcmout(vorbis_dsp_state *v,ogg_int16_t *pcm,int samples){
+ vorbis_info *vi=v->vi;
+ codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+ if(v->out_begin>-1 && v->out_begin<v->out_end){
+ int n=v->out_end-v->out_begin;
+ if(pcm){
+ int i;
+ if(n>samples)n=samples;
+ for(i=0;i<vi->channels;i++)
+ mdct_unroll_lap(ci->blocksizes[0],ci->blocksizes[1],
+ v->lW,v->W,v->work[i],v->mdctright[i],
+ _vorbis_window(ci->blocksizes[0]>>1),
+ _vorbis_window(ci->blocksizes[1]>>1),
+ pcm+i,vi->channels,
+ v->out_begin,v->out_begin+n);
+ }
+ return(n);
+ }
+ return(0);
+}
+
+int vorbis_dsp_read(vorbis_dsp_state *v,int s){
+ if(s && v->out_begin+s>v->out_end)return(OV_EINVAL);
+ v->out_begin+=s;
+ return(0);
+}
+
+long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op){
+ codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+ oggpack_buffer opb;
+ int mode;
+ int modebits=0;
+ int v=ci->modes;
+
+ oggpack_readinit(&opb,op->packet);
+
+ /* Check the packet type */
+ if(oggpack_read(&opb,1)!=0){
+ /* Oops. This is not an audio data packet */
+ return(OV_ENOTAUDIO);
+ }
+
+ while(v>1){
+ modebits++;
+ v>>=1;
+ }
+
+ /* read our mode and pre/post windowsize */
+ mode=oggpack_read(&opb,modebits);
+ if(mode==-1)return(OV_EBADPACKET);
+ return(ci->blocksizes[ci->mode_param[mode].blockflag]);
+}
+
+
+static int ilog(ogg_uint32_t v){
+ int ret=0;
+ if(v)--v;
+ while(v){
+ ret++;
+ v>>=1;
+ }
+ return(ret);
+}
+
+int vorbis_dsp_synthesis(vorbis_dsp_state *vd,ogg_packet *op,int decodep){
+ vorbis_info *vi=vd->vi;
+ codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+ int mode,i;
+
+ oggpack_readinit(&vd->opb,op->packet);
+
+ /* Check the packet type */
+ if(oggpack_read(&vd->opb,1)!=0){
+ /* Oops. This is not an audio data packet */
+ return OV_ENOTAUDIO ;
+ }
+
+ /* read our mode and pre/post windowsize */
+ mode=oggpack_read(&vd->opb,ilog(ci->modes));
+ if(mode==-1 || mode>=ci->modes) return OV_EBADPACKET;
+
+ /* shift information we still need from last window */
+ vd->lW=vd->W;
+ vd->W=ci->mode_param[mode].blockflag;
+ for(i=0;i<vi->channels;i++)
+ mdct_shift_right(ci->blocksizes[vd->lW],vd->work[i],vd->mdctright[i]);
+
+ if(vd->W){
+ int temp;
+ oggpack_read(&vd->opb,1);
+ temp=oggpack_read(&vd->opb,1);
+ if(temp==-1) return OV_EBADPACKET;
+ }
+
+ /* packet decode and portions of synthesis that rely on only this block */
+ if(decodep){
+ mapping_inverse(vd,ci->map_param+ci->mode_param[mode].mapping);
+
+ if(vd->out_begin==-1){
+ vd->out_begin=0;
+ vd->out_end=0;
+ }else{
+ vd->out_begin=0;
+ vd->out_end=ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+ }
+ }
+
+ /* track the frame number... This is for convenience, but also
+ making sure our last packet doesn't end with added padding.
+
+ This is not foolproof! It will be confused if we begin
+ decoding at the last page after a seek or hole. In that case,
+ we don't have a starting point to judge where the last frame
+ is. For this reason, vorbisfile will always try to make sure
+ it reads the last two marked pages in proper sequence */
+
+ /* if we're out of sequence, dump granpos tracking until we sync back up */
+ if(vd->sequence==-1 || vd->sequence+1 != op->packetno-3){
+ /* out of sequence; lose count */
+ vd->granulepos=-1;
+ vd->sample_count=-1;
+ }
+
+ vd->sequence=op->packetno;
+ vd->sequence=vd->sequence-3;
+
+ if(vd->sample_count==-1){
+ vd->sample_count=0;
+ }else{
+ vd->sample_count+=
+ ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+ }
+
+ if(vd->granulepos==-1){
+ if(op->granulepos!=-1){ /* only set if we have a
+ position to set to */
+
+ vd->granulepos=op->granulepos;
+
+ /* is this a short page? */
+ if(vd->sample_count>vd->granulepos){
+ /* corner case; if this is both the first and last audio page,
+ then spec says the end is cut, not beginning */
+ if(op->e_o_s){
+ /* trim the end */
+ /* no preceeding granulepos; assume we started at zero (we'd
+ have to in a short single-page stream) */
+ /* granulepos could be -1 due to a seek, but that would result
+ in a long coun t, not short count */
+
+ vd->out_end-=(int)(vd->sample_count-vd->granulepos);
+ }else{
+ /* trim the beginning */
+ vd->out_begin+=(int)(vd->sample_count-vd->granulepos);
+ if(vd->out_begin>vd->out_end)
+ vd->out_begin=vd->out_end;
+ }
+
+ }
+
+ }
+ }else{
+ vd->granulepos+=
+ ci->blocksizes[vd->lW]/4+ci->blocksizes[vd->W]/4;
+ if(op->granulepos!=-1 && vd->granulepos!=op->granulepos){
+
+ if(vd->granulepos>op->granulepos){
+ long extra=(long)(vd->granulepos-op->granulepos);
+
+ if(extra)
+ if(op->e_o_s){
+ /* partial last frame. Strip the extra samples off */
+ vd->out_end-=extra;
+ } /* else {Shouldn't happen *unless* the bitstream is out of
+ spec. Either way, believe the bitstream } */
+ } /* else {Shouldn't happen *unless* the bitstream is out of
+ spec. Either way, believe the bitstream } */
+ vd->granulepos=op->granulepos;
+ }
+ }
+
+ return(0);
+}
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