* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
- * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002 *
- * by the XIPHOPHORUS Company http://www.xiph.org/ *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2015 *
+ * by the Xiph.Org Foundation https://xiph.org/ *
* *
********************************************************************
function: PCM data vector blocking, windowing and dis/reassembly
- last mod: $Id: block.c,v 1.59 2002/02/28 04:12:48 xiphmont Exp $
Handle windowing, overlap-add, etc of the PCM vectors. This is made
more amusing by Vorbis' current two allowed block sizes.
-
+
********************************************************************/
#include <stdio.h>
#include "registry.h"
#include "misc.h"
-static int ilog2(unsigned int v){
- int ret=0;
- while(v>1){
- ret++;
- v>>=1;
- }
- return(ret);
-}
-
/* pcm accumulator examples (not exhaustive):
<-------------- lW ---------------->
:.....''' |_____--- '''......| | \_______|
:.................|__________________|_______|__|______|
|<------ Sl ------>| > Sr < |endW
- |beginSl |endSl | |endSr
+ |beginSl |endSl | |endSr
|beginW |endlW |beginSr
- |< lW >|
+ |< lW >|
<--------------- W ---------------->
| | .. ______________ |
| | ' `/ | ---_ |
- |___.'___/`. | ---_____|
+ |___.'___/`. | ---_____|
|_______|__|_______|_________________|
| >|Sl|< |<------ Sr ----->|endW
| | |endSl |beginSr |endSr
- |beginW | |endlW
+ |beginW | |endlW
mult[0] |beginSl mult[n]
<-------------- lW ----------------->
- |<--W-->|
-: .............. ___ | |
-: .''' |`/ \ | |
-:.....''' |/`....\|...|
-:.........................|___|___|___|
- |Sl |Sr |endW
+ |<--W-->|
+: .............. ___ | |
+: .''' |`/ \ | |
+:.....''' |/`....\|...|
+:.........................|___|___|___|
+ |Sl |Sr |endW
| | |endSr
| |beginSr
| |endSl
- |beginSl
- |beginW
+ |beginSl
+ |beginW
*/
/* block abstraction setup *********************************************/
#endif
int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb){
+ int i;
memset(vb,0,sizeof(*vb));
vb->vd=v;
vb->localalloc=0;
if(v->analysisp){
vorbis_block_internal *vbi=
vb->internal=_ogg_calloc(1,sizeof(vorbis_block_internal));
- oggpack_writeinit(&vb->opb);
vbi->ampmax=-9999;
- vbi->packet_markers=_ogg_malloc(vorbis_bitrate_maxmarkers()*
- sizeof(*vbi->packet_markers));
+
+ for(i=0;i<PACKETBLOBS;i++){
+ if(i==PACKETBLOBS/2){
+ vbi->packetblob[i]=&vb->opb;
+ }else{
+ vbi->packetblob[i]=
+ _ogg_calloc(1,sizeof(oggpack_buffer));
+ }
+ oggpack_writeinit(vbi->packetblob[i]);
+ }
}
-
+
return(0);
}
}
int vorbis_block_clear(vorbis_block *vb){
- if(vb->vd)
- if(vb->vd->analysisp)
- oggpack_writeclear(&vb->opb);
+ int i;
+ vorbis_block_internal *vbi=vb->internal;
+
_vorbis_block_ripcord(vb);
if(vb->localstore)_ogg_free(vb->localstore);
- if(vb->internal){
- vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
- if(vbi->packet_markers)_ogg_free(vbi->packet_markers);
-
- _ogg_free(vb->internal);
+ if(vbi){
+ for(i=0;i<PACKETBLOBS;i++){
+ oggpack_writeclear(vbi->packetblob[i]);
+ if(i!=PACKETBLOBS/2)_ogg_free(vbi->packetblob[i]);
+ }
+ _ogg_free(vbi);
}
-
memset(vb,0,sizeof(*vb));
return(0);
}
static int _vds_shared_init(vorbis_dsp_state *v,vorbis_info *vi,int encp){
int i;
codec_setup_info *ci=vi->codec_setup;
- backend_lookup_state *b=NULL;
+ private_state *b=NULL;
+ int hs;
+
+ if(ci==NULL||
+ ci->modes<=0||
+ ci->blocksizes[0]<64||
+ ci->blocksizes[1]<ci->blocksizes[0]){
+ return 1;
+ }
+ hs=ci->halfrate_flag;
memset(v,0,sizeof(*v));
b=v->backend_state=_ogg_calloc(1,sizeof(*b));
v->vi=vi;
- b->modebits=ilog2(ci->modes);
+ b->modebits=ov_ilog(ci->modes-1);
b->transform[0]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[0]));
b->transform[1]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[1]));
b->transform[0][0]=_ogg_calloc(1,sizeof(mdct_lookup));
b->transform[1][0]=_ogg_calloc(1,sizeof(mdct_lookup));
- mdct_init(b->transform[0][0],ci->blocksizes[0]);
- mdct_init(b->transform[1][0],ci->blocksizes[1]);
+ mdct_init(b->transform[0][0],ci->blocksizes[0]>>hs);
+ mdct_init(b->transform[1][0],ci->blocksizes[1]>>hs);
/* Vorbis I uses only window type 0 */
- b->window[0]=_vorbis_window(0,ci->blocksizes[0]/2);
- b->window[1]=_vorbis_window(0,ci->blocksizes[1]/2);
+ /* note that the correct computation below is technically:
+ b->window[0]=ov_ilog(ci->blocksizes[0]-1)-6;
+ b->window[1]=ov_ilog(ci->blocksizes[1]-1)-6;
+ but since blocksizes are always powers of two,
+ the below is equivalent.
+ */
+ b->window[0]=ov_ilog(ci->blocksizes[0])-7;
+ b->window[1]=ov_ilog(ci->blocksizes[1])-7;
if(encp){ /* encode/decode differ here */
+
+ /* analysis always needs an fft */
+ drft_init(&b->fft_look[0],ci->blocksizes[0]);
+ drft_init(&b->fft_look[1],ci->blocksizes[1]);
+
/* finish the codebooks */
if(!ci->fullbooks){
ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
for(i=0;i<ci->books;i++)
- vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
+ vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
}
+
+ b->psy=_ogg_calloc(ci->psys,sizeof(*b->psy));
+ for(i=0;i<ci->psys;i++){
+ _vp_psy_init(b->psy+i,
+ ci->psy_param[i],
+ &ci->psy_g_param,
+ ci->blocksizes[ci->psy_param[i]->blockflag]/2,
+ vi->rate);
+ }
+
v->analysisp=1;
}else{
/* finish the codebooks */
if(!ci->fullbooks){
ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
for(i=0;i<ci->books;i++){
- vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]);
- /* decode codebooks are now standalone after init */
- vorbis_staticbook_destroy(ci->book_param[i]);
- ci->book_param[i]=NULL;
+ if(ci->book_param[i]==NULL)
+ goto abort_books;
+ if(vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]))
+ goto abort_books;
+ /* decode codebooks are now standalone after init */
+ vorbis_staticbook_destroy(ci->book_param[i]);
+ ci->book_param[i]=NULL;
}
}
}
v->pcm_current=v->centerW;
- /* initialize all the mapping/backend lookups */
- b->mode=_ogg_calloc(ci->modes,sizeof(*b->mode));
- for(i=0;i<ci->modes;i++){
- int mapnum=ci->mode_param[i]->mapping;
- int maptype=ci->map_type[mapnum];
- b->mode[i]=_mapping_P[maptype]->look(v,ci->mode_param[i],
- ci->map_param[mapnum]);
- }
+ /* initialize all the backend lookups */
+ b->flr=_ogg_calloc(ci->floors,sizeof(*b->flr));
+ b->residue=_ogg_calloc(ci->residues,sizeof(*b->residue));
- return(0);
+ for(i=0;i<ci->floors;i++)
+ b->flr[i]=_floor_P[ci->floor_type[i]]->
+ look(v,ci->floor_param[i]);
+
+ for(i=0;i<ci->residues;i++)
+ b->residue[i]=_residue_P[ci->residue_type[i]]->
+ look(v,ci->residue_param[i]);
+
+ return 0;
+ abort_books:
+ for(i=0;i<ci->books;i++){
+ if(ci->book_param[i]!=NULL){
+ vorbis_staticbook_destroy(ci->book_param[i]);
+ ci->book_param[i]=NULL;
+ }
+ }
+ vorbis_dsp_clear(v);
+ return -1;
}
/* arbitrary settings and spec-mandated numbers get filled in here */
int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi){
- backend_lookup_state *b=NULL;
+ private_state *b=NULL;
- _vds_shared_init(v,vi,1);
+ if(_vds_shared_init(v,vi,1))return 1;
b=v->backend_state;
b->psy_g_look=_vp_global_look(vi);
vorbis_bitrate_init(vi,&b->bms);
+ /* compressed audio packets start after the headers
+ with sequence number 3 */
+ v->sequence=3;
+
return(0);
}
if(v){
vorbis_info *vi=v->vi;
codec_setup_info *ci=(vi?vi->codec_setup:NULL);
- backend_lookup_state *b=v->backend_state;
+ private_state *b=v->backend_state;
if(b){
- if(b->window[0])
- _ogg_free(b->window[0]);
- if(b->window[1])
- _ogg_free(b->window[1]);
-
+
if(b->ve){
- _ve_envelope_clear(b->ve);
- _ogg_free(b->ve);
+ _ve_envelope_clear(b->ve);
+ _ogg_free(b->ve);
}
if(b->transform[0]){
- mdct_clear(b->transform[0][0]);
- _ogg_free(b->transform[0][0]);
- _ogg_free(b->transform[0]);
+ mdct_clear(b->transform[0][0]);
+ _ogg_free(b->transform[0][0]);
+ _ogg_free(b->transform[0]);
}
if(b->transform[1]){
- mdct_clear(b->transform[1][0]);
- _ogg_free(b->transform[1][0]);
- _ogg_free(b->transform[1]);
+ mdct_clear(b->transform[1][0]);
+ _ogg_free(b->transform[1][0]);
+ _ogg_free(b->transform[1]);
+ }
+
+ if(b->flr){
+ if(ci)
+ for(i=0;i<ci->floors;i++)
+ _floor_P[ci->floor_type[i]]->
+ free_look(b->flr[i]);
+ _ogg_free(b->flr);
+ }
+ if(b->residue){
+ if(ci)
+ for(i=0;i<ci->residues;i++)
+ _residue_P[ci->residue_type[i]]->
+ free_look(b->residue[i]);
+ _ogg_free(b->residue);
+ }
+ if(b->psy){
+ if(ci)
+ for(i=0;i<ci->psys;i++)
+ _vp_psy_clear(b->psy+i);
+ _ogg_free(b->psy);
}
+
if(b->psy_g_look)_vp_global_free(b->psy_g_look);
vorbis_bitrate_clear(&b->bms);
+
+ drft_clear(&b->fft_look[0]);
+ drft_clear(&b->fft_look[1]);
+
}
-
+
if(v->pcm){
- for(i=0;i<vi->channels;i++)
- if(v->pcm[i])_ogg_free(v->pcm[i]);
+ if(vi)
+ for(i=0;i<vi->channels;i++)
+ if(v->pcm[i])_ogg_free(v->pcm[i]);
_ogg_free(v->pcm);
if(v->pcmret)_ogg_free(v->pcmret);
}
- /* free mode lookups; these are actually vorbis_look_mapping structs */
- if(ci){
- for(i=0;i<ci->modes;i++){
- int mapnum=ci->mode_param[i]->mapping;
- int maptype=ci->map_type[mapnum];
- if(b && b->mode)_mapping_P[maptype]->free_look(b->mode[i]);
- }
- }
-
if(b){
- if(b->mode)_ogg_free(b->mode);
-
/* free header, header1, header2 */
if(b->header)_ogg_free(b->header);
if(b->header1)_ogg_free(b->header1);
if(b->header2)_ogg_free(b->header2);
_ogg_free(b);
}
-
+
memset(v,0,sizeof(*v));
}
}
float **vorbis_analysis_buffer(vorbis_dsp_state *v, int vals){
int i;
vorbis_info *vi=v->vi;
- backend_lookup_state *b=v->backend_state;
+ private_state *b=v->backend_state;
/* free header, header1, header2 */
if(b->header)_ogg_free(b->header);b->header=NULL;
/* Do we have enough storage space for the requested buffer? If not,
expand the PCM (and envelope) storage */
-
+
if(v->pcm_current+vals>=v->pcm_storage){
v->pcm_storage=v->pcm_current+vals*2;
-
+
for(i=0;i<vi->channels;i++){
v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
}
for(i=0;i<vi->channels;i++)
v->pcmret[i]=v->pcm[i]+v->pcm_current;
-
+
return(v->pcmret);
}
static void _preextrapolate_helper(vorbis_dsp_state *v){
int i;
- int order=32;
+ int order=16;
float *lpc=alloca(order*sizeof(*lpc));
float *work=alloca(v->pcm_current*sizeof(*work));
long j;
for(i=0;i<v->vi->channels;i++){
/* need to run the extrapolation in reverse! */
for(j=0;j<v->pcm_current;j++)
- work[j]=v->pcm[i][v->pcm_current-j-1];
-
+ work[j]=v->pcm[i][v->pcm_current-j-1];
+
/* prime as above */
vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
-
+
+#if 0
+ if(v->vi->channels==2){
+ if(i==0)
+ _analysis_output("predataL",0,work,v->pcm_current-v->centerW,0,0,0);
+ else
+ _analysis_output("predataR",0,work,v->pcm_current-v->centerW,0,0,0);
+ }else{
+ _analysis_output("predata",0,work,v->pcm_current-v->centerW,0,0,0);
+ }
+#endif
+
/* run the predictor filter */
vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
- order,
- work+v->pcm_current-v->centerW,
- v->centerW);
+ order,
+ work+v->pcm_current-v->centerW,
+ v->centerW);
for(j=0;j<v->pcm_current;j++)
- v->pcm[i][v->pcm_current-j-1]=work[j];
+ v->pcm[i][v->pcm_current-j-1]=work[j];
}
}
int vorbis_analysis_wrote(vorbis_dsp_state *v, int vals){
vorbis_info *vi=v->vi;
codec_setup_info *ci=vi->codec_setup;
- /*backend_lookup_state *b=v->backend_state;*/
if(vals<=0){
int order=32;
_preextrapolate_helper(v);
/* We're encoding the end of the stream. Just make sure we have
- [at least] a full block of zeroes at the end. */
+ [at least] a few full blocks of zeroes at the end. */
/* actually, we don't want zeroes; that could drop a large
amplitude off a cliff, creating spread spectrum noise that will
suck to encode. Extrapolate for the sake of cleanliness. */
- vorbis_analysis_buffer(v,ci->blocksizes[1]*2);
+ vorbis_analysis_buffer(v,ci->blocksizes[1]*3);
v->eofflag=v->pcm_current;
- v->pcm_current+=ci->blocksizes[1]*2;
+ v->pcm_current+=ci->blocksizes[1]*3;
for(i=0;i<vi->channels;i++){
if(v->eofflag>order*2){
- /* extrapolate with LPC to fill in */
- long n;
+ /* extrapolate with LPC to fill in */
+ long n;
- /* make a predictor filter */
- n=v->eofflag;
- if(n>ci->blocksizes[1])n=ci->blocksizes[1];
- vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
+ /* make a predictor filter */
+ n=v->eofflag;
+ if(n>ci->blocksizes[1])n=ci->blocksizes[1];
+ vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
- /* run the predictor filter */
- vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
- v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
+ /* run the predictor filter */
+ vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
+ v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
}else{
- /* not enough data to extrapolate (unlikely to happen due to
+ /* not enough data to extrapolate (unlikely to happen due to
guarding the overlap, but bulletproof in case that
assumtion goes away). zeroes will do. */
- memset(v->pcm[i]+v->eofflag,0,
- (v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
+ memset(v->pcm[i]+v->eofflag,0,
+ (v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
}
}
int i;
vorbis_info *vi=v->vi;
codec_setup_info *ci=vi->codec_setup;
- backend_lookup_state *b=v->backend_state;
+ private_state *b=v->backend_state;
vorbis_look_psy_global *g=b->psy_g_look;
- vorbis_info_psy_global *gi=&ci->psy_g_param;
long beginW=v->centerW-ci->blocksizes[v->W]/2,centerNext;
vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
/* By our invariant, we have lW, W and centerW set. Search for
the next boundary so we can determine nW (the next window size)
which lets us compute the shape of the current block's window */
-
- if(ci->blocksizes[0]<ci->blocksizes[1]){
+
+ /* we do an envelope search even on a single blocksize; we may still
+ be throwing more bits at impulses, and envelope search handles
+ marking impulses too. */
+ {
long bp=_ve_envelope_search(v);
- if(bp==-1)return(0); /* not enough data currently to search for a
- full long block */
- v->nW=bp;
+ if(bp==-1){
+
+ if(v->eofflag==0)return(0); /* not enough data currently to search for a
+ full long block */
+ v->nW=0;
+ }else{
+
+ if(ci->blocksizes[0]==ci->blocksizes[1])
+ v->nW=0;
+ else
+ v->nW=bp;
+ }
+ }
- }else
- v->nW=0;
-
centerNext=v->centerW+ci->blocksizes[v->W]/4+ci->blocksizes[v->nW]/4;
{
}
-
+
/* fill in the block. Note that for a short window, lW and nW are *short*
regardless of actual settings in the stream */
_vorbis_block_ripcord(vb);
- if(v->W){
- vb->lW=v->lW;
- vb->W=v->W;
- vb->nW=v->nW;
- }else{
- vb->lW=0;
- vb->W=v->W;
- vb->nW=0;
- }
+ vb->lW=v->lW;
+ vb->W=v->W;
+ vb->nW=v->nW;
if(v->W){
- if(!v->lW || !v->nW)
+ if(!v->lW || !v->nW){
vbi->blocktype=BLOCKTYPE_TRANSITION;
- else
+ /*fprintf(stderr,"-");*/
+ }else{
vbi->blocktype=BLOCKTYPE_LONG;
+ /*fprintf(stderr,"_");*/
+ }
}else{
- if(_ve_envelope_mark(v))
+ if(_ve_envelope_mark(v)){
vbi->blocktype=BLOCKTYPE_IMPULSE;
- else
+ /*fprintf(stderr,"|");*/
+
+ }else{
vbi->blocktype=BLOCKTYPE_PADDING;
+ /*fprintf(stderr,".");*/
+
+ }
}
-
+
vb->vd=v;
vb->sequence=v->sequence++;
vb->granulepos=v->granulepos;
vb->pcmend=ci->blocksizes[v->W];
-
+
/* copy the vectors; this uses the local storage in vb */
/* this tracks 'strongest peak' for later psychoacoustics */
if(vbi->ampmax>g->ampmax)g->ampmax=vbi->ampmax;
g->ampmax=_vp_ampmax_decay(g->ampmax,v);
vbi->ampmax=g->ampmax;
-
+
vb->pcm=_vorbis_block_alloc(vb,sizeof(*vb->pcm)*vi->channels);
vbi->pcmdelay=_vorbis_block_alloc(vb,sizeof(*vbi->pcmdelay)*vi->channels);
for(i=0;i<vi->channels;i++){
_vorbis_block_alloc(vb,(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
memcpy(vbi->pcmdelay[i],v->pcm[i],(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
vb->pcm[i]=vbi->pcmdelay[i]+beginW;
-
- /* before we added the delay
+
+ /* before we added the delay
vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i]));
memcpy(vb->pcm[i],v->pcm[i]+beginW,ci->blocksizes[v->W]*sizeof(*vb->pcm[i]));
*/
-
+
}
-
+
/* handle eof detection: eof==0 means that we've not yet received EOF
eof>0 marks the last 'real' sample in pcm[]
eof<0 'no more to do'; doesn't get here */
/* advance storage vectors and clean up */
{
- int new_centerNext=ci->blocksizes[1]/2+gi->delaycache;
+ int new_centerNext=ci->blocksizes[1]/2;
int movementW=centerNext-new_centerNext;
if(movementW>0){
_ve_envelope_shift(b->ve,movementW);
v->pcm_current-=movementW;
-
+
for(i=0;i<vi->channels;i++)
- memmove(v->pcm[i],v->pcm[i]+movementW,
- v->pcm_current*sizeof(*v->pcm[i]));
-
-
+ memmove(v->pcm[i],v->pcm[i]+movementW,
+ v->pcm_current*sizeof(*v->pcm[i]));
+
+
v->lW=v->W;
v->W=v->nW;
v->centerW=new_centerNext;
-
+
if(v->eofflag){
- v->eofflag-=movementW;
- /* do not add padding to end of stream! */
- if(v->centerW>=v->eofflag){
- v->granulepos+=movementW-(v->centerW-v->eofflag);
- }else{
- v->granulepos+=movementW;
- }
+ v->eofflag-=movementW;
+ if(v->eofflag<=0)v->eofflag=-1;
+ /* do not add padding to end of stream! */
+ if(v->centerW>=v->eofflag){
+ v->granulepos+=movementW-(v->centerW-v->eofflag);
+ }else{
+ v->granulepos+=movementW;
+ }
}else{
- v->granulepos+=movementW;
+ v->granulepos+=movementW;
}
}
}
return(1);
}
-int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){
- _vds_shared_init(v,vi,0);
+int vorbis_synthesis_restart(vorbis_dsp_state *v){
+ vorbis_info *vi=v->vi;
+ codec_setup_info *ci;
+ int hs;
+
+ if(!v->backend_state)return -1;
+ if(!vi)return -1;
+ ci=vi->codec_setup;
+ if(!ci)return -1;
+ hs=ci->halfrate_flag;
+
+ v->centerW=ci->blocksizes[1]>>(hs+1);
+ v->pcm_current=v->centerW>>hs;
v->pcm_returned=-1;
v->granulepos=-1;
v->sequence=-1;
+ v->eofflag=0;
+ ((private_state *)(v->backend_state))->sample_count=-1;
return(0);
}
+int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){
+ if(_vds_shared_init(v,vi,0)){
+ vorbis_dsp_clear(v);
+ return 1;
+ }
+ vorbis_synthesis_restart(v);
+ return 0;
+}
+
/* Unlike in analysis, the window is only partially applied for each
block. The time domain envelope is not yet handled at the point of
calling (as it relies on the previous block). */
int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb){
vorbis_info *vi=v->vi;
codec_setup_info *ci=vi->codec_setup;
+ private_state *b=v->backend_state;
+ int hs=ci->halfrate_flag;
int i,j;
if(!vb)return(OV_EINVAL);
if(v->pcm_current>v->pcm_returned && v->pcm_returned!=-1)return(OV_EINVAL);
-
+
v->lW=v->W;
v->W=vb->W;
v->nW=-1;
-
- if(v->sequence+1 != vb->sequence)v->granulepos=-1; /* out of sequence;
- lose count */
+
+ if((v->sequence==-1)||
+ (v->sequence+1 != vb->sequence)){
+ v->granulepos=-1; /* out of sequence; lose count */
+ b->sample_count=-1;
+ }
+
v->sequence=vb->sequence;
-
- if(vb->pcm){ /* not pcm to process if vorbis_synthesis_trackonly
- was called on block */
- int n=ci->blocksizes[v->W]/2;
- int n0=ci->blocksizes[0]/2;
- int n1=ci->blocksizes[1]/2;
-
+
+ if(vb->pcm){ /* no pcm to process if vorbis_synthesis_trackonly
+ was called on block */
+ int n=ci->blocksizes[v->W]>>(hs+1);
+ int n0=ci->blocksizes[0]>>(hs+1);
+ int n1=ci->blocksizes[1]>>(hs+1);
+
int thisCenter;
int prevCenter;
-
+
v->glue_bits+=vb->glue_bits;
v->time_bits+=vb->time_bits;
v->floor_bits+=vb->floor_bits;
v->res_bits+=vb->res_bits;
-
+
if(v->centerW){
thisCenter=n1;
prevCenter=0;
}else{
- thisCenter=0;
- prevCenter=n1;
+ thisCenter=0;
+ prevCenter=n1;
}
-
+
/* v->pcm is now used like a two-stage double buffer. We don't want
to have to constantly shift *or* adjust memory usage. Don't
accept a new block until the old is shifted out */
-
- /* overlap/add PCM */
-
+
for(j=0;j<vi->channels;j++){
/* the overlap/add section */
if(v->lW){
- if(v->W){
- /* large/large */
- float *pcm=v->pcm[j]+prevCenter;
- float *p=vb->pcm[j];
- for(i=0;i<n1;i++)
- pcm[i]+=p[i];
- }else{
- /* large/small */
- float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
- float *p=vb->pcm[j];
- for(i=0;i<n0;i++)
- pcm[i]+=p[i];
- }
+ if(v->W){
+ /* large/large */
+ const float *w=_vorbis_window_get(b->window[1]-hs);
+ float *pcm=v->pcm[j]+prevCenter;
+ float *p=vb->pcm[j];
+ for(i=0;i<n1;i++)
+ pcm[i]=pcm[i]*w[n1-i-1] + p[i]*w[i];
+ }else{
+ /* large/small */
+ const float *w=_vorbis_window_get(b->window[0]-hs);
+ float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
+ float *p=vb->pcm[j];
+ for(i=0;i<n0;i++)
+ pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+ }
}else{
- if(v->W){
- /* small/large */
- float *pcm=v->pcm[j]+prevCenter;
- float *p=vb->pcm[j]+n1/2-n0/2;
- for(i=0;i<n0;i++)
- pcm[i]+=p[i];
- for(;i<n1/2+n0/2;i++)
- pcm[i]=p[i];
- }else{
- /* small/small */
- float *pcm=v->pcm[j]+prevCenter;
- float *p=vb->pcm[j];
- for(i=0;i<n0;i++)
- pcm[i]+=p[i];
- }
+ if(v->W){
+ /* small/large */
+ const float *w=_vorbis_window_get(b->window[0]-hs);
+ float *pcm=v->pcm[j]+prevCenter;
+ float *p=vb->pcm[j]+n1/2-n0/2;
+ for(i=0;i<n0;i++)
+ pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+ for(;i<n1/2+n0/2;i++)
+ pcm[i]=p[i];
+ }else{
+ /* small/small */
+ const float *w=_vorbis_window_get(b->window[0]-hs);
+ float *pcm=v->pcm[j]+prevCenter;
+ float *p=vb->pcm[j];
+ for(i=0;i<n0;i++)
+ pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+ }
}
-
+
/* the copy section */
{
- float *pcm=v->pcm[j]+thisCenter;
- float *p=vb->pcm[j]+n;
- for(i=0;i<n;i++)
- pcm[i]=p[i];
+ float *pcm=v->pcm[j]+thisCenter;
+ float *p=vb->pcm[j]+n;
+ for(i=0;i<n;i++)
+ pcm[i]=p[i];
}
}
-
+
if(v->centerW)
v->centerW=0;
else
v->centerW=n1;
-
+
/* deal with initial packet state; we do this using the explicit
pcm_returned==-1 flag otherwise we're sensitive to first block
being short or long */
-
+
if(v->pcm_returned==-1){
v->pcm_returned=thisCenter;
v->pcm_current=thisCenter;
}else{
v->pcm_returned=prevCenter;
v->pcm_current=prevCenter+
- ci->blocksizes[v->lW]/4+
- ci->blocksizes[v->W]/4;
+ ((ci->blocksizes[v->lW]/4+
+ ci->blocksizes[v->W]/4)>>hs);
}
-
+
}
/* track the frame number... This is for convenience, but also
making sure our last packet doesn't end with added padding. If
the last packet is partial, the number of samples we'll have to
return will be past the vb->granulepos.
-
+
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(b->sample_count==-1){
+ b->sample_count=0;
+ }else{
+ b->sample_count+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
+ }
+
if(v->granulepos==-1){
if(vb->granulepos!=-1){ /* only set if we have a position to set to */
+
v->granulepos=vb->granulepos;
+
+ /* is this a short page? */
+ if(b->sample_count>v->granulepos){
+ /* corner case; if this is both the first and last audio page,
+ then spec says the end is cut, not beginning */
+ long extra=b->sample_count-vb->granulepos;
+
+ /* we use ogg_int64_t for granule positions because a
+ uint64 isn't universally available. Unfortunately,
+ that means granposes can be 'negative' and result in
+ extra being negative */
+ if(extra<0)
+ extra=0;
+
+ if(vb->eofflag){
+ /* trim the end */
+ /* no preceding 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 count, not short count */
+
+ /* Guard against corrupt/malicious frames that set EOP and
+ a backdated granpos; don't rewind more samples than we
+ actually have */
+ if(extra > (v->pcm_current - v->pcm_returned)<<hs)
+ extra = (v->pcm_current - v->pcm_returned)<<hs;
+
+ v->pcm_current-=extra>>hs;
+ }else{
+ /* trim the beginning */
+ v->pcm_returned+=extra>>hs;
+ if(v->pcm_returned>v->pcm_current)
+ v->pcm_returned=v->pcm_current;
+ }
+
+ }
+
}
}else{
v->granulepos+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
if(vb->granulepos!=-1 && v->granulepos!=vb->granulepos){
-
+
if(v->granulepos>vb->granulepos){
- long extra=v->granulepos-vb->granulepos;
-
- if(vb->eofflag){
- /* partial last frame. Strip the extra samples off */
- v->pcm_current-=extra;
- }else if(vb->sequence == 1){
- /* ^^^ argh, this can be 1 from seeking! */
-
-
- /* partial first frame. Discard extra leading samples */
- v->pcm_returned+=extra;
- if(v->pcm_returned>v->pcm_current)
- v->pcm_returned=v->pcm_current;
-
- } /* else {Shouldn't happen *unless* the bitstream is out of
- spec. Either way, believe the bitstream } */
+ long extra=v->granulepos-vb->granulepos;
+
+ if(extra)
+ if(vb->eofflag){
+ /* partial last frame. Strip the extra samples off */
+
+ /* Guard against corrupt/malicious frames that set EOP and
+ a backdated granpos; don't rewind more samples than we
+ actually have */
+ if(extra > (v->pcm_current - v->pcm_returned)<<hs)
+ extra = (v->pcm_current - v->pcm_returned)<<hs;
+
+ /* we use ogg_int64_t for granule positions because a
+ uint64 isn't universally available. Unfortunately,
+ that means granposes can be 'negative' and result in
+ extra being negative */
+ if(extra<0)
+ extra=0;
+
+ v->pcm_current-=extra>>hs;
+ } /* 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 } */
+ spec. Either way, believe the bitstream } */
v->granulepos=vb->granulepos;
}
}
-
+
/* Update, cleanup */
-
+
if(vb->eofflag)v->eofflag=1;
return(0);
-
+
}
/* pcm==NULL indicates we just want the pending samples, no more */
int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm){
vorbis_info *vi=v->vi;
+
if(v->pcm_returned>-1 && v->pcm_returned<v->pcm_current){
if(pcm){
int i;
for(i=0;i<vi->channels;i++)
- v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+ v->pcmret[i]=v->pcm[i]+v->pcm_returned;
*pcm=v->pcmret;
}
return(v->pcm_current-v->pcm_returned);
return(0);
}
+/* intended for use with a specific vorbisfile feature; we want access
+ to the [usually synthetic/postextrapolated] buffer and lapping at
+ the end of a decode cycle, specifically, a half-short-block worth.
+ This funtion works like pcmout above, except it will also expose
+ this implicit buffer data not normally decoded. */
+int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm){
+ vorbis_info *vi=v->vi;
+ codec_setup_info *ci=vi->codec_setup;
+ int hs=ci->halfrate_flag;
+
+ int n=ci->blocksizes[v->W]>>(hs+1);
+ int n0=ci->blocksizes[0]>>(hs+1);
+ int n1=ci->blocksizes[1]>>(hs+1);
+ int i,j;
+
+ if(v->pcm_returned<0)return 0;
+
+ /* our returned data ends at pcm_returned; because the synthesis pcm
+ buffer is a two-fragment ring, that means our data block may be
+ fragmented by buffering, wrapping or a short block not filling
+ out a buffer. To simplify things, we unfragment if it's at all
+ possibly needed. Otherwise, we'd need to call lapout more than
+ once as well as hold additional dsp state. Opt for
+ simplicity. */
+
+ /* centerW was advanced by blockin; it would be the center of the
+ *next* block */
+ if(v->centerW==n1){
+ /* the data buffer wraps; swap the halves */
+ /* slow, sure, small */
+ for(j=0;j<vi->channels;j++){
+ float *p=v->pcm[j];
+ for(i=0;i<n1;i++){
+ float temp=p[i];
+ p[i]=p[i+n1];
+ p[i+n1]=temp;
+ }
+ }
+
+ v->pcm_current-=n1;
+ v->pcm_returned-=n1;
+ v->centerW=0;
+ }
+
+ /* solidify buffer into contiguous space */
+ if((v->lW^v->W)==1){
+ /* long/short or short/long */
+ for(j=0;j<vi->channels;j++){
+ float *s=v->pcm[j];
+ float *d=v->pcm[j]+(n1-n0)/2;
+ for(i=(n1+n0)/2-1;i>=0;--i)
+ d[i]=s[i];
+ }
+ v->pcm_returned+=(n1-n0)/2;
+ v->pcm_current+=(n1-n0)/2;
+ }else{
+ if(v->lW==0){
+ /* short/short */
+ for(j=0;j<vi->channels;j++){
+ float *s=v->pcm[j];
+ float *d=v->pcm[j]+n1-n0;
+ for(i=n0-1;i>=0;--i)
+ d[i]=s[i];
+ }
+ v->pcm_returned+=n1-n0;
+ v->pcm_current+=n1-n0;
+ }
+ }
+
+ if(pcm){
+ int i;
+ for(i=0;i<vi->channels;i++)
+ v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+ *pcm=v->pcmret;
+ }
+
+ return(n1+n-v->pcm_returned);
+
+}
+
+const float *vorbis_window(vorbis_dsp_state *v,int W){
+ vorbis_info *vi=v->vi;
+ codec_setup_info *ci=vi->codec_setup;
+ int hs=ci->halfrate_flag;
+ private_state *b=v->backend_state;
+
+ if(b->window[W]-1<0)return NULL;
+ return _vorbis_window_get(b->window[W]-hs);
+}
projects / platform / upstream / libvorbis.git / blobdiff