* 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 1994-2001 *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2003 *
* by the XIPHOPHORUS Company http://www.xiph.org/ *
-
+ * *
********************************************************************
function: PCM data vector blocking, windowing and dis/reassembly
- last mod: $Id: block.c,v 1.51 2001/12/12 09:45:24 xiphmont Exp $
+ last mod: $Id: block.c,v 1.75 2003/09/02 04:39:26 xiphmont Exp $
Handle windowing, overlap-add, etc of the PCM vectors. This is made
more amusing by Vorbis' current two allowed block sizes.
static int ilog2(unsigned int v){
int ret=0;
- while(v>1){
+ if(v)--v;
+ while(v){
ret++;
v>>=1;
}
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));
}
return(0);
_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);
-
+ if(vb->internal)
_ogg_free(vb->internal);
- }
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) return 1;
+ hs=ci->halfrate_flag;
memset(v,0,sizeof(*v));
b=v->backend_state=_ogg_calloc(1,sizeof(*b));
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]);
-
- b->window[0][0][0]=_ogg_calloc(VI_WINDOWB,sizeof(*b->window[0][0][0]));
- b->window[0][0][1]=b->window[0][0][0];
- b->window[0][1][0]=b->window[0][0][0];
- b->window[0][1][1]=b->window[0][0][0];
- b->window[1][0][0]=_ogg_calloc(VI_WINDOWB,sizeof(*b->window[1][0][0]));
- b->window[1][0][1]=_ogg_calloc(VI_WINDOWB,sizeof(*b->window[1][0][1]));
- b->window[1][1][0]=_ogg_calloc(VI_WINDOWB,sizeof(*b->window[1][1][0]));
- b->window[1][1][1]=_ogg_calloc(VI_WINDOWB,sizeof(*b->window[1][1][1]));
-
- for(i=0;i<VI_WINDOWB;i++){
- b->window[0][0][0][i]=
- _vorbis_window(i,ci->blocksizes[0],ci->blocksizes[0]/2,ci->blocksizes[0]/2);
- b->window[1][0][0][i]=
- _vorbis_window(i,ci->blocksizes[1],ci->blocksizes[0]/2,ci->blocksizes[0]/2);
- b->window[1][0][1][i]=
- _vorbis_window(i,ci->blocksizes[1],ci->blocksizes[0]/2,ci->blocksizes[1]/2);
- b->window[1][1][0][i]=
- _vorbis_window(i,ci->blocksizes[1],ci->blocksizes[1]/2,ci->blocksizes[0]/2);
- b->window[1][1][1][i]=
- _vorbis_window(i,ci->blocksizes[1],ci->blocksizes[1]/2,ci->blocksizes[1]/2);
- }
+ 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]=ilog2(ci->blocksizes[0])-6;
+ b->window[1]=ilog2(ci->blocksizes[1])-6;
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 */
- b->fullbooks=_ogg_calloc(ci->books,sizeof(*b->fullbooks));
- for(i=0;i<ci->books;i++)
- vorbis_book_init_encode(b->fullbooks+i,ci->book_param[i]);
+ 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]);
+ }
+
+ 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 */
- b->fullbooks=_ogg_calloc(ci->books,sizeof(*b->fullbooks));
- for(i=0;i<ci->books;i++)
- vorbis_book_init_decode(b->fullbooks+i,ci->book_param[i]);
+ 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;
+ }
+ }
}
- /* initialize the storage vectors to a decent size greater than the
- minimum */
-
- v->pcm_storage=8192; /* we'll assume later that we have
- a minimum of twice the blocksize of
- accumulated samples in analysis */
+ /* initialize the storage vectors. blocksize[1] is small for encode,
+ but the correct size for decode */
+ v->pcm_storage=ci->blocksizes[1];
v->pcm=_ogg_malloc(vi->channels*sizeof(*v->pcm));
v->pcmret=_ogg_malloc(vi->channels*sizeof(*v->pcmret));
{
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;
}
/* 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);
}
void vorbis_dsp_clear(vorbis_dsp_state *v){
- int i,j,k;
+ int i;
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][0][0]){
- for(i=0;i<VI_WINDOWB;i++)
- if(b->window[0][0][0][i])_ogg_free(b->window[0][0][0][i]);
- _ogg_free(b->window[0][0][0]);
- for(j=0;j<2;j++)
- for(k=0;k<2;k++){
- for(i=0;i<VI_WINDOWB;i++)
- if(b->window[1][j][k][i])_ogg_free(b->window[1][j][k][i]);
- _ogg_free(b->window[1][j][k]);
- }
- }
-
if(b->ve){
_ve_envelope_clear(b->ve);
_ogg_free(b->ve);
_ogg_free(b->transform[1][0]);
_ogg_free(b->transform[1]);
}
+
+ if(b->flr){
+ 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){
+ 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){
+ 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){
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]);
- }
- /* free codebooks */
- for(i=0;i<ci->books;i++)
- if(b && b->fullbooks)vorbis_book_clear(b->fullbooks+i);
- }
-
if(b){
- if(b->mode)_ogg_free(b->mode);
- if(b->fullbooks)_ogg_free(b->fullbooks);
-
/* free header, header1, header2 */
if(b->header)_ogg_free(b->header);
if(b->header1)_ogg_free(b->header1);
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;
return(v->pcmret);
}
-static int seq=0;
static void _preextrapolate_helper(vorbis_dsp_state *v){
int i;
int order=32;
for(j=0;j<v->pcm_current;j++)
work[j]=v->pcm[i][v->pcm_current-j-1];
- _analysis_output("preextrap",seq,v->pcm[i],v->pcm_current,0,0);
- _analysis_output("workextrap",seq,work,v->pcm_current,0,0);
-
/* prime as above */
vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
- _analysis_output("lpc",seq,lpc,order,0,0);
/* run the predictor filter */
vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
work+v->pcm_current-v->centerW,
v->centerW);
- _analysis_output("extrap",seq,work,v->pcm_current,0,0);
-
-
for(j=0;j<v->pcm_current;j++)
v->pcm[i][v->pcm_current-j-1]=work[j];
- _analysis_output("postextrap",seq++,v->pcm[i],v->pcm_current,0,0);
}
}
}
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){
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;
- //v->nW=0;
+ 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;
{
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->sequence=v->sequence++;
vb->granulepos=v->granulepos;
vb->pcmend=ci->blocksizes[v->W];
-
/* copy the vectors; this uses the local storage in vb */
- {
- vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
-
- /* this tracks 'strongest peak' for later psychoacoustics */
- /* moved to the global psy state; clean this mess up */
- 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++){
- vbi->pcmdelay[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
- 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]));
- */
- }
+ /* this tracks 'strongest peak' for later psychoacoustics */
+ /* moved to the global psy state; clean this mess up */
+ 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++){
+ vbi->pcmdelay[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
+ 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
/* 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){
v->W=v->nW;
v->centerW=new_centerNext;
- v->sequence++;
-
if(v->eofflag){
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);
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)) 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;
- /* Shift out any PCM that we returned previously */
- /* centerW is currently the center of the last block added */
-
- if(v->centerW>ci->blocksizes[1]/2 &&
- /* Quick additional hack; to avoid *alot* of shifts, use an
- oversized buffer. This increases memory usage, but doesn't make
- much difference wrt L1/L2 cache pressure. */
- v->pcm_returned>8192){
-
- /* don't shift too much; we need to have a minimum PCM buffer of
- 1/2 long block */
-
- int shiftPCM=v->centerW-ci->blocksizes[1]/2;
- shiftPCM=(v->pcm_returned<shiftPCM?v->pcm_returned:shiftPCM);
-
- v->pcm_current-=shiftPCM;
- v->centerW-=shiftPCM;
- v->pcm_returned-=shiftPCM;
+ if(!vb)return(OV_EINVAL);
+ if(v->pcm_current>v->pcm_returned && v->pcm_returned!=-1)return(OV_EINVAL);
- if(shiftPCM){
- int i;
- for(i=0;i<vi->channels;i++)
- memmove(v->pcm[i],v->pcm[i]+shiftPCM,
- v->pcm_current*sizeof(*v->pcm[i]));
- }
- }
-
v->lW=v->W;
v->W=vb->W;
v->nW=-1;
-
- 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->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;
-
- {
- int sizeW=ci->blocksizes[v->W];
- int centerW=v->centerW+ci->blocksizes[v->lW]/4+sizeW/4;
- int beginW=centerW-sizeW/2;
- int endW=beginW+sizeW;
- int beginSl;
- int endSl;
- int i,j;
-
- /* Do we have enough PCM/mult storage for the block? */
- if(endW>v->pcm_storage){
- /* expand the storage */
- v->pcm_storage=endW+ci->blocksizes[1];
-
- for(i=0;i<vi->channels;i++)
- v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
- }
-
- /* overlap/add PCM */
-
- switch(v->W){
- case 0:
- beginSl=0;
- endSl=ci->blocksizes[0]/2;
- break;
- case 1:
- beginSl=ci->blocksizes[1]/4-ci->blocksizes[v->lW]/4;
- endSl=beginSl+ci->blocksizes[v->lW]/2;
- break;
- default:
- return(-1);
+
+ 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;
}
-
+
+ /* 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 */
+
for(j=0;j<vi->channels;j++){
- static int seq=0;
- float *pcm=v->pcm[j]+beginW;
- float *p=vb->pcm[j];
-
/* the overlap/add section */
- for(i=beginSl;i<endSl;i++)
- pcm[i]+=p[i];
- /* the remaining section */
- for(;i<sizeW;i++)
- pcm[i]=p[i];
-
- //_analysis_output("lapped",seq,pcm,sizeW,0,0);
- //_analysis_output("buffered",seq++,v->pcm[j],sizeW+beginW,0,0);
-
+ if(v->lW){
+ if(v->W){
+ /* large/large */
+ 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 */
+ 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 *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 */
+ 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];
+ }
}
-
+
+ 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)>>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 */
+ if(vb->eofflag){
+ /* 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 count, not short count */
+
+ v->pcm_current-=(b->sample_count-v->granulepos)>>hs;
+ }else{
+ /* trim the beginning */
+ v->pcm_returned+=(b->sample_count-v->granulepos)>>hs;
+ if(v->pcm_returned>v->pcm_current)
+ v->pcm_returned=v->pcm_current;
+ }
- if(v->pcm_returned==-1)
- v->pcm_returned=centerW;
-
- /* 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(v->granulepos==-1)
- if(vb->granulepos==-1){
- v->granulepos=0;
- }else{
- v->granulepos=vb->granulepos;
}
- else{
- v->granulepos+=(centerW-v->centerW);
- if(vb->granulepos!=-1 && v->granulepos!=vb->granulepos){
- if(v->granulepos>vb->granulepos){
- long extra=v->granulepos-vb->granulepos;
+ }
+ }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(extra)
if(vb->eofflag){
/* partial last frame. Strip the extra samples off */
- centerW-=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>centerW)v->pcm_returned=centerW;
-
- }
-
- }/* else{ Shouldn't happen *unless* the bitstream is out of
- spec. Either way, believe the bitstream } */
- v->granulepos=vb->granulepos;
- }
+ 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 } */
+ v->granulepos=vb->granulepos;
}
-
- /* Update, cleanup */
-
- v->centerW=centerW;
- v->pcm_current=endW;
-
- if(vb->eofflag)v->eofflag=1;
}
-
+
+ /* 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->centerW){
+
+ 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;
*pcm=v->pcmret;
}
- return(v->centerW-v->pcm_returned);
+ return(v->pcm_current-v->pcm_returned);
}
return(0);
}
-int vorbis_synthesis_read(vorbis_dsp_state *v,int bytes){
- if(bytes && v->pcm_returned+bytes>v->centerW)return(OV_EINVAL);
- v->pcm_returned+=bytes;
+int vorbis_synthesis_read(vorbis_dsp_state *v,int n){
+ if(n && v->pcm_returned+n>v->pcm_current)return(OV_EINVAL);
+ v->pcm_returned+=n;
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);
+
+}
+
+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