int main(int argc,char *argv[]){
int eos=0;
float nonz=0.f;
float acc=0.f;
float tot=0.f;
float ampmax=-9999,newmax;
float local_ampmax[2];
int framesize=2048;
float ampmax_att_per_sec=-6.;
float *pcm[2],*out[2],*window,*flr[2],*mask[2],*work[2];
signed char *buffer,*buffer2;
mdct_lookup m_look;
drft_lookup f_look;
vorbis_look_psy p_look;
vorbis_look_psy_global *pg_look;
vorbis_look_floor *floor_look;
vorbis_info vi;
long i,j,k;
int ath=0;
int decayp=0;
argv++;
while(*argv){
if(*argv[0]=='-'){
/* option */
if(argv[0][1]=='v'){
noisy=0;
}
}else
if(*argv[0]=='+'){
/* option */
if(argv[0][1]=='v'){
noisy=1;
}
}else
framesize=atoi(argv[0]);
argv++;
}
vi.channels=2;
vi.codec_setup=&codec_setup0;
pcm[0]=_ogg_malloc(framesize*sizeof(float));
pcm[1]=_ogg_malloc(framesize*sizeof(float));
out[0]=_ogg_calloc(framesize/2,sizeof(float));
out[1]=_ogg_calloc(framesize/2,sizeof(float));
work[0]=_ogg_calloc(framesize,sizeof(float));
work[1]=_ogg_calloc(framesize,sizeof(float));
flr[0]=_ogg_calloc(framesize/2,sizeof(float));
flr[1]=_ogg_calloc(framesize/2,sizeof(float));
buffer=_ogg_malloc(framesize*4);
buffer2=buffer+framesize*2;
window=_vorbis_window_create(0,framesize,framesize/2,framesize/2);
mdct_init(&m_look,framesize);
drft_init(&f_look,framesize);
_vp_psy_init(&p_look,&_psy_set0,&_psy_set0G,framesize/2,44100);
pg_look=_vp_global_look(&vi);
floor_look=_floor_P[1]->look(NULL,NULL,&_floor_set0);
/* we cheat on the WAV header; we just bypass 44 bytes and never
verify that it matches 16bit/stereo/44.1kHz. */
fread(buffer,1,44,stdin);
fwrite(buffer,1,44,stdout);
memset(buffer,0,framesize*2);
analysis("window",0,window,framesize,0,0);
fprintf(stderr,"Processing for frame size %d...\n",framesize);
while(!eos){
long bytes=fread(buffer2,1,framesize*2,stdin);
if(bytes<framesize*2)
memset(buffer2+bytes,0,framesize*2-bytes);
if(bytes!=0){
int nonzero[2];
/* uninterleave samples */
for(i=0;i<framesize;i++){
pcm[0][i]=((buffer[i*4+1]<<8)|
(0x00ff&(int)buffer[i*4]))/32768.f;
pcm[1][i]=((buffer[i*4+3]<<8)|
(0x00ff&(int)buffer[i*4+2]))/32768.f;
}
{
float secs=framesize/44100.;
ampmax+=secs*ampmax_att_per_sec;
if(ampmax<-9999)ampmax=-9999;
}
for(i=0;i<2;i++){
float scale=4.f/framesize;
float *fft=work[i];
float *mdct=pcm[i];
float *logmdct=mdct+framesize/2;
analysis("pre",frameno+i,pcm[i],framesize,0,0);
/* fft and mdct transforms */
for(j=0;j<framesize;j++)
fft[j]=pcm[i][j]*=window[j];
drft_forward(&f_look,fft);
local_ampmax[i]=-9999.f;
fft[0]*=scale;
fft[0]=todB(fft);
for(j=1;j<framesize-1;j+=2){
float temp=scale*FAST_HYPOT(fft[j],fft[j+1]);
temp=fft[(j+1)>>1]=todB(&temp);
if(temp>local_ampmax[i])local_ampmax[i]=temp;
}
if(local_ampmax[i]>ampmax)ampmax=local_ampmax[i];
mdct_forward(&m_look,pcm[i],mdct);
for(j=0;j<framesize/2;j++)
logmdct[j]=todB(mdct+j);
analysis("mdct",frameno+i,logmdct,framesize/2,1,0);
analysis("fft",frameno+i,fft,framesize/2,1,0);
}
for(i=0;i<2;i++){
float amp;
float *fft=work[i];
float *logmax=fft;
float *mdct=pcm[i];
float *logmdct=mdct+framesize/2;
float *mask=fft+framesize/2;
/* floor psychoacoustics */
_vp_compute_mask(&p_look,
pg_look,
i,
fft,
logmdct,
mask,
ampmax,
local_ampmax[i],
framesize/2);
analysis("mask",frameno+i,mask,framesize/2,1,0);
{
vorbis_block vb;
vorbis_dsp_state vd;
memset(&vd,0,sizeof(vd));
vd.vi=&vi;
vb.vd=&vd;
vb.pcmend=framesize;
/* floor quantization/application */
nonzero[i]=_floor_P[1]->forward(&vb,floor_look,
mdct,
logmdct,
mask,
logmax,
flr[i]);
}
_vp_remove_floor(&p_look,
pg_look,
logmdct,
mdct,
flr[i],
pcm[i],
local_ampmax[i]);
for(j=0;j<framesize/2;j++)
if(fabs(pcm[i][j])>1500)
fprintf(stderr,"%ld ",frameno+i);
analysis("res",frameno+i,pcm[i],framesize/2,1,0);
analysis("codedflr",frameno+i,flr[i],framesize/2,1,1);
}
/* residue prequantization */
_vp_partition_prequant(&p_look,
&vi,
pcm,
nonzero);
for(i=0;i<2;i++)
analysis("quant",frameno+i,pcm[i],framesize/2,1,0);
/* channel coupling / stereo quantization */
_vp_couple(&p_look,
&mapping_info,
pcm,
nonzero);
for(i=0;i<2;i++)
analysis("coupled",frameno+i,pcm[i],framesize/2,1,0);
/* decoupling */
for(i=mapping_info.coupling_steps-1;i>=0;i--){
float *pcmM=pcm[mapping_info.coupling_mag[i]];
float *pcmA=pcm[mapping_info.coupling_ang[i]];
for(j=0;j<framesize/2;j++){
float mag=pcmM[j];
float ang=pcmA[j];
if(mag>0)
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag-ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag+ang;
}
else
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag+ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag-ang;
}
}
}
for(i=0;i<2;i++)
analysis("decoupled",frameno+i,pcm[i],framesize/2,1,0);
for(i=0;i<2;i++){
float amp;
for(j=0;j<framesize/2;j++)
pcm[i][j]*=flr[i][j];
analysis("final",frameno+i,pcm[i],framesize/2,1,1);
/* take it back to time */
mdct_backward(&m_look,pcm[i],pcm[i]);
for(j=0;j<framesize/2;j++)
out[i][j]+=pcm[i][j]*window[j];
analysis("out",frameno+i,out[i],framesize/2,0,0);
}
/* write data. Use the part of buffer we're about to shift out */
for(i=0;i<2;i++){
char *ptr=buffer+i*2;
float *mono=out[i];
int flag=0;
for(j=0;j<framesize/2;j++){
int val=mono[j]*32767.;
/* might as well guard against clipping */
if(val>32767){
if(!flag)fprintf(stderr,"clipping in frame %ld ",frameno+i);
flag=1;
val=32767;
}
if(val<-32768){
if(!flag)fprintf(stderr,"clipping in frame %ld ",frameno+i);
flag=1;
val=-32768;
}
ptr[0]=val&0xff;
ptr[1]=(val>>8)&0xff;
ptr+=4;
}
}
fprintf(stderr,"*");
fwrite(buffer,1,framesize*2,stdout);
memmove(buffer,buffer2,framesize*2);
for(i=0;i<2;i++){
for(j=0,k=framesize/2;j<framesize/2;j++,k++)
out[i][j]=pcm[i][k]*window[k];
}
frameno+=2;
}else
static int _ve_amp(envelope_lookup *ve,
vorbis_info_psy_global *gi,
float *data,
envelope_band *bands,
envelope_filter_state *filters,
long pos){
long n=ve->winlength;
int ret=0;
long i,j;
float decay;
/* we want to have a 'minimum bar' for energy, else we're just
basing blocks on quantization noise that outweighs the signal
itself (for low power signals) */
float minV=ve->minenergy;
float *vec=alloca(n*sizeof(*vec));
/* stretch is used to gradually lengthen the number of windows
considered prevoius-to-potential-trigger */
int stretch=max(VE_MINSTRETCH,ve->stretch/2);
float penalty=gi->stretch_penalty-(ve->stretch/2-VE_MINSTRETCH);
if(penalty<0.f)penalty=0.f;
if(penalty>gi->stretch_penalty)penalty=gi->stretch_penalty;
/*_analysis_output_always("lpcm",seq2,data,n,0,0,
totalshift+pos*ve->searchstep);*/
/* window and transform */
for(i=0;i<n;i++)
vec[i]=data[i]*ve->mdct_win[i];
mdct_forward(&ve->mdct,vec,vec);
/*_analysis_output_always("mdct",seq2,vec,n/2,0,1,0); */
/* near-DC spreading function; this has nothing to do with
psychoacoustics, just sidelobe leakage and window size */
{
float temp=vec[0]*vec[0]+.7*vec[1]*vec[1]+.2*vec[2]*vec[2];
int ptr=filters->nearptr;
/* the accumulation is regularly refreshed from scratch to avoid
floating point creep */
if(ptr==0){
decay=filters->nearDC_acc=filters->nearDC_partialacc+temp;
filters->nearDC_partialacc=temp;
}else{
decay=filters->nearDC_acc+=temp;
filters->nearDC_partialacc+=temp;
}
filters->nearDC_acc-=filters->nearDC[ptr];
filters->nearDC[ptr]=temp;
decay*=(1./(VE_NEARDC+1));
filters->nearptr++;
if(filters->nearptr>=VE_NEARDC)filters->nearptr=0;
decay=todB(&decay)*.5-15.f;
}
/* perform spreading and limiting, also smooth the spectrum. yes,
the MDCT results in all real coefficients, but it still *behaves*
like real/imaginary pairs */
for(i=0;i<n/2;i+=2){
float val=vec[i]*vec[i]+vec[i+1]*vec[i+1];
val=todB(&val)*.5f;
if(val<decay)val=decay;
if(val<minV)val=minV;
vec[i>>1]=val;
decay-=8.;
}
/*_analysis_output_always("spread",seq2++,vec,n/4,0,0,0);*/
/* perform preecho/postecho triggering by band */
for(j=0;j<VE_BANDS;j++){
float acc=0.;
float valmax,valmin;
/* accumulate amplitude */
for(i=0;i<bands[j].end;i++)
acc+=vec[i+bands[j].begin]*bands[j].window[i];
acc*=bands[j].total;
/* convert amplitude to delta */
{
int p,this=filters[j].ampptr;
float postmax,postmin,premax=-99999.f,premin=99999.f;
p=this;
p--;
if(p<0)p+=VE_AMP;
postmax=max(acc,filters[j].ampbuf[p]);
postmin=min(acc,filters[j].ampbuf[p]);
for(i=0;i<stretch;i++){
p--;
if(p<0)p+=VE_AMP;
premax=max(premax,filters[j].ampbuf[p]);
premin=min(premin,filters[j].ampbuf[p]);
}
valmin=postmin-premin;
valmax=postmax-premax;
/*filters[j].markers[pos]=valmax;*/
filters[j].ampbuf[this]=acc;
filters[j].ampptr++;
if(filters[j].ampptr>=VE_AMP)filters[j].ampptr=0;
}
/* look at min/max, decide trigger */
if(valmax>gi->preecho_thresh[j]+penalty){
ret|=1;
ret|=4;
}
if(valmin<gi->postecho_thresh[j]-penalty)ret|=2;
}
return(ret);
}
static int mapping0_forward(vorbis_block *vb){
vorbis_dsp_state *vd=vb->vd;
vorbis_info *vi=vd->vi;
codec_setup_info *ci=vi->codec_setup;
private_state *b=vb->vd->backend_state;
vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
int n=vb->pcmend;
int i,j,k;
int *nonzero = alloca(sizeof(*nonzero)*vi->channels);
float **gmdct = _vorbis_block_alloc(vb,vi->channels*sizeof(*gmdct));
int **iwork = _vorbis_block_alloc(vb,vi->channels*sizeof(*iwork));
int ***floor_posts = _vorbis_block_alloc(vb,vi->channels*sizeof(*floor_posts));
float global_ampmax=vbi->ampmax;
float *local_ampmax=alloca(sizeof(*local_ampmax)*vi->channels);
int blocktype=vbi->blocktype;
int modenumber=vb->W;
vorbis_info_mapping0 *info=ci->map_param[modenumber];
vorbis_look_psy *psy_look=b->psy+blocktype+(vb->W?2:0);
vb->mode=modenumber;
for(i=0;i<vi->channels;i++){
float scale=4.f/n;
float scale_dB;
float *pcm =vb->pcm[i];
float *logfft =pcm;
iwork[i]=_vorbis_block_alloc(vb,n/2*sizeof(**iwork));
gmdct[i]=_vorbis_block_alloc(vb,n/2*sizeof(**gmdct));
scale_dB=todB(&scale) + .345; /* + .345 is a hack; the original
todB estimation used on IEEE 754
compliant machines had a bug that
returned dB values about a third
of a decibel too high. The bug
was harmless because tunings
implicitly took that into
account. However, fixing the bug
in the estimator requires
changing all the tunings as well.
For now, it's easier to sync
things back up here, and
recalibrate the tunings in the
next major model upgrade. */
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("pcmL",seq,pcm,n,0,0,total-n/2);
else
_analysis_output("pcmR",seq,pcm,n,0,0,total-n/2);
}else{
_analysis_output("pcm",seq,pcm,n,0,0,total-n/2);
}
#endif
/* window the PCM data */
_vorbis_apply_window(pcm,b->window,ci->blocksizes,vb->lW,vb->W,vb->nW);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("windowedL",seq,pcm,n,0,0,total-n/2);
else
_analysis_output("windowedR",seq,pcm,n,0,0,total-n/2);
}else{
_analysis_output("windowed",seq,pcm,n,0,0,total-n/2);
}
#endif
/* transform the PCM data */
/* only MDCT right now.... */
mdct_forward(b->transform[vb->W][0],pcm,gmdct[i]);
/* FFT yields more accurate tonal estimation (not phase sensitive) */
drft_forward(&b->fft_look[vb->W],pcm);
logfft[0]=scale_dB+todB(pcm) + .345; /* + .345 is a hack; the
original todB estimation used on
IEEE 754 compliant machines had a
bug that returned dB values about
a third of a decibel too high.
The bug was harmless because
tunings implicitly took that into
account. However, fixing the bug
in the estimator requires
changing all the tunings as well.
For now, it's easier to sync
things back up here, and
recalibrate the tunings in the
next major model upgrade. */
local_ampmax[i]=logfft[0];
for(j=1;j<n-1;j+=2){
float temp=pcm[j]*pcm[j]+pcm[j+1]*pcm[j+1];
temp=logfft[(j+1)>>1]=scale_dB+.5f*todB(&temp) + .345; /* +
.345 is a hack; the original todB
estimation used on IEEE 754
compliant machines had a bug that
returned dB values about a third
of a decibel too high. The bug
was harmless because tunings
implicitly took that into
account. However, fixing the bug
in the estimator requires
changing all the tunings as well.
For now, it's easier to sync
things back up here, and
recalibrate the tunings in the
next major model upgrade. */
if(temp>local_ampmax[i])local_ampmax[i]=temp;
}
if(local_ampmax[i]>0.f)local_ampmax[i]=0.f;
if(local_ampmax[i]>global_ampmax)global_ampmax=local_ampmax[i];
#if 0
if(vi->channels==2){
if(i==0){
_analysis_output("fftL",seq,logfft,n/2,1,0,0);
}else{
_analysis_output("fftR",seq,logfft,n/2,1,0,0);
}
}else{
_analysis_output("fft",seq,logfft,n/2,1,0,0);
}
#endif
}
{
float *noise = _vorbis_block_alloc(vb,n/2*sizeof(*noise));
float *tone = _vorbis_block_alloc(vb,n/2*sizeof(*tone));
for(i=0;i<vi->channels;i++){
/* the encoder setup assumes that all the modes used by any
specific bitrate tweaking use the same floor */
int submap=info->chmuxlist[i];
/* the following makes things clearer to *me* anyway */
float *mdct =gmdct[i];
float *logfft =vb->pcm[i];
float *logmdct =logfft+n/2;
float *logmask =logfft;
vb->mode=modenumber;
floor_posts[i]=_vorbis_block_alloc(vb,PACKETBLOBS*sizeof(**floor_posts));
memset(floor_posts[i],0,sizeof(**floor_posts)*PACKETBLOBS);
for(j=0;j<n/2;j++)
logmdct[j]=todB(mdct+j) + .345; /* + .345 is a hack; the original
todB estimation used on IEEE 754
compliant machines had a bug that
returned dB values about a third
of a decibel too high. The bug
was harmless because tunings
implicitly took that into
account. However, fixing the bug
in the estimator requires
changing all the tunings as well.
For now, it's easier to sync
things back up here, and
recalibrate the tunings in the
next major model upgrade. */
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("mdctL",seq,logmdct,n/2,1,0,0);
else
_analysis_output("mdctR",seq,logmdct,n/2,1,0,0);
}else{
_analysis_output("mdct",seq,logmdct,n/2,1,0,0);
}
#endif
/* first step; noise masking. Not only does 'noise masking'
give us curves from which we can decide how much resolution
to give noise parts of the spectrum, it also implicitly hands
us a tonality estimate (the larger the value in the
'noise_depth' vector, the more tonal that area is) */
_vp_noisemask(psy_look,
logmdct,
noise); /* noise does not have by-frequency offset
bias applied yet */
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("noiseL",seq,noise,n/2,1,0,0);
else
_analysis_output("noiseR",seq,noise,n/2,1,0,0);
}else{
_analysis_output("noise",seq,noise,n/2,1,0,0);
}
#endif
/* second step: 'all the other crap'; all the stuff that isn't
computed/fit for bitrate management goes in the second psy
vector. This includes tone masking, peak limiting and ATH */
_vp_tonemask(psy_look,
logfft,
tone,
global_ampmax,
local_ampmax[i]);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("toneL",seq,tone,n/2,1,0,0);
else
_analysis_output("toneR",seq,tone,n/2,1,0,0);
}else{
_analysis_output("tone",seq,tone,n/2,1,0,0);
}
#endif
/* third step; we offset the noise vectors, overlay tone
masking. We then do a floor1-specific line fit. If we're
performing bitrate management, the line fit is performed
multiple times for up/down tweakage on demand. */
#if 0
{
float aotuv[psy_look->n];
#endif
_vp_offset_and_mix(psy_look,
noise,
tone,
1,
logmask,
mdct,
logmdct);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("aotuvM1_L",seq,aotuv,psy_look->n,1,1,0);
else
_analysis_output("aotuvM1_R",seq,aotuv,psy_look->n,1,1,0);
}else{
_analysis_output("aotuvM1",seq,aotuv,psy_look->n,1,1,0);
}
}
#endif
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("mask1L",seq,logmask,n/2,1,0,0);
else
_analysis_output("mask1R",seq,logmask,n/2,1,0,0);
}else{
_analysis_output("mask1",seq,logmask,n/2,1,0,0);
}
#endif
/* this algorithm is hardwired to floor 1 for now; abort out if
we're *not* floor1. This won't happen unless someone has
broken the encode setup lib. Guard it anyway. */
if(ci->floor_type[info->floorsubmap[submap]]!=1)return(-1);
floor_posts[i][PACKETBLOBS/2]=
floor1_fit(vb,b->flr[info->floorsubmap[submap]],
logmdct,
logmask);
/* are we managing bitrate? If so, perform two more fits for
later rate tweaking (fits represent hi/lo) */
if(vorbis_bitrate_managed(vb) && floor_posts[i][PACKETBLOBS/2]){
/* higher rate by way of lower noise curve */
_vp_offset_and_mix(psy_look,
noise,
tone,
2,
logmask,
mdct,
logmdct);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("mask2L",seq,logmask,n/2,1,0,0);
else
_analysis_output("mask2R",seq,logmask,n/2,1,0,0);
}else{
_analysis_output("mask2",seq,logmask,n/2,1,0,0);
}
#endif
floor_posts[i][PACKETBLOBS-1]=
floor1_fit(vb,b->flr[info->floorsubmap[submap]],
logmdct,
logmask);
/* lower rate by way of higher noise curve */
_vp_offset_and_mix(psy_look,
noise,
tone,
0,
logmask,
mdct,
logmdct);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("mask0L",seq,logmask,n/2,1,0,0);
else
_analysis_output("mask0R",seq,logmask,n/2,1,0,0);
}else{
_analysis_output("mask0",seq,logmask,n/2,1,0,0);
}
#endif
floor_posts[i][0]=
floor1_fit(vb,b->flr[info->floorsubmap[submap]],
logmdct,
logmask);
/* we also interpolate a range of intermediate curves for
intermediate rates */
for(k=1;k<PACKETBLOBS/2;k++)
floor_posts[i][k]=
floor1_interpolate_fit(vb,b->flr[info->floorsubmap[submap]],
floor_posts[i][0],
floor_posts[i][PACKETBLOBS/2],
k*65536/(PACKETBLOBS/2));
for(k=PACKETBLOBS/2+1;k<PACKETBLOBS-1;k++)
floor_posts[i][k]=
floor1_interpolate_fit(vb,b->flr[info->floorsubmap[submap]],
floor_posts[i][PACKETBLOBS/2],
floor_posts[i][PACKETBLOBS-1],
(k-PACKETBLOBS/2)*65536/(PACKETBLOBS/2));
}
}
}
vbi->ampmax=global_ampmax;
/*
the next phases are performed once for vbr-only and PACKETBLOB
times for bitrate managed modes.
1) encode actual mode being used
2) encode the floor for each channel, compute coded mask curve/res
3) normalize and couple.
4) encode residue
5) save packet bytes to the packetblob vector
*/
/* iterate over the many masking curve fits we've created */
{
int **couple_bundle=alloca(sizeof(*couple_bundle)*vi->channels);
int *zerobundle=alloca(sizeof(*zerobundle)*vi->channels);
for(k=(vorbis_bitrate_managed(vb)?0:PACKETBLOBS/2);
k<=(vorbis_bitrate_managed(vb)?PACKETBLOBS-1:PACKETBLOBS/2);
k++){
oggpack_buffer *opb=vbi->packetblob[k];
/* start out our new packet blob with packet type and mode */
/* Encode the packet type */
oggpack_write(opb,0,1);
/* Encode the modenumber */
/* Encode frame mode, pre,post windowsize, then dispatch */
oggpack_write(opb,modenumber,b->modebits);
if(vb->W){
oggpack_write(opb,vb->lW,1);
oggpack_write(opb,vb->nW,1);
}
/* encode floor, compute masking curve, sep out residue */
for(i=0;i<vi->channels;i++){
int submap=info->chmuxlist[i];
int *ilogmask=iwork[i];
nonzero[i]=floor1_encode(opb,vb,b->flr[info->floorsubmap[submap]],
floor_posts[i][k],
ilogmask);
#if 0
{
char buf[80];
sprintf(buf,"maskI%c%d",i?'R':'L',k);
float work[n/2];
for(j=0;j<n/2;j++)
work[j]=FLOOR1_fromdB_LOOKUP[iwork[i][j]];
_analysis_output(buf,seq,work,n/2,1,1,0);
}
#endif
}
/* our iteration is now based on masking curve, not prequant and
coupling. Only one prequant/coupling step */
/* quantize/couple */
/* incomplete implementation that assumes the tree is all depth
one, or no tree at all */
_vp_couple_quantize_normalize(k,
&ci->psy_g_param,
psy_look,
info,
gmdct,
iwork,
nonzero,
ci->psy_g_param.sliding_lowpass[vb->W][k],
vi->channels);
#if 0
for(i=0;i<vi->channels;i++){
char buf[80];
sprintf(buf,"res%c%d",i?'R':'L',k);
float work[n/2];
for(j=0;j<n/2;j++)
work[j]=iwork[i][j];
_analysis_output(buf,seq,work,n/2,1,0,0);
}
#endif
/* classify and encode by submap */
for(i=0;i<info->submaps;i++){
int ch_in_bundle=0;
long **classifications;
int resnum=info->residuesubmap[i];
for(j=0;j<vi->channels;j++){
if(info->chmuxlist[j]==i){
zerobundle[ch_in_bundle]=0;
if(nonzero[j])zerobundle[ch_in_bundle]=1;
couple_bundle[ch_in_bundle++]=iwork[j];
}
}
classifications=_residue_P[ci->residue_type[resnum]]->
class(vb,b->residue[resnum],couple_bundle,zerobundle,ch_in_bundle);
ch_in_bundle=0;
for(j=0;j<vi->channels;j++)
if(info->chmuxlist[j]==i)
couple_bundle[ch_in_bundle++]=iwork[j];
_residue_P[ci->residue_type[resnum]]->
forward(opb,vb,b->residue[resnum],
couple_bundle,zerobundle,ch_in_bundle,classifications,i);
}
/* ok, done encoding. Next protopacket. */
}
}
#if 0
seq++;
total+=ci->blocksizes[vb->W]/4+ci->blocksizes[vb->nW]/4;
#endif
return(0);
}
static int mapping0_forward(vorbis_block *vb,vorbis_look_mapping *l){
vorbis_dsp_state *vd=vb->vd;
vorbis_info *vi=vd->vi;
codec_setup_info *ci=vi->codec_setup;
backend_lookup_state *b=vb->vd->backend_state;
vorbis_look_mapping0 *look=(vorbis_look_mapping0 *)l;
vorbis_info_mapping0 *info=look->map;
vorbis_info_mode *mode=look->mode;
vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
int n=vb->pcmend;
int i,j;
float *window=b->window[vb->W][vb->lW][vb->nW][mode->windowtype];
int *nonzero=alloca(sizeof(int)*vi->channels);
float *work=_vorbis_block_alloc(vb,n*sizeof(float));
float global_ampmax=vbi->ampmax;
float *local_ampmax=alloca(sizeof(float)*vi->channels);
int blocktype;
/* we differentiate between short and long block types to help the
masking engine; the window shapes also matter.
impulse block (a short block in which an impulse occurs)
padding block (a short block that pads between a transitional
long block and an impulse block, or vice versa)
transition block (the wqeird one; a long block with the transition
window; affects bass/midrange response and that must be
accounted for in masking)
long block (run of the mill long block)
*/
if(vb->W){
if(!vb->lW || !vb->nW)
blocktype=BLOCKTYPE_TRANSITION;
else
blocktype=BLOCKTYPE_LONG;
}else{
/* right now we're missing the infrastructure to distingush the
two short types */
blocktype=BLOCKTYPE_IMPULSE;
}
for(i=0;i<vi->channels;i++){
float scale=4.f/n;
/* the following makes things clearer to *me* anyway */
float *pcm =vb->pcm[i];
float *fft =work;
float *logfft =pcm+n/2;
/*float *res =pcm;
float *mdct =pcm;
float *codedflr=pcm+n/2;
float *logmax =work;
float *logmask =work+n/2;*/
/* window the PCM data */
for(j=0;j<n;j++)
fft[j]=pcm[j]*=window[j];
/* transform the PCM data */
/* only MDCT right now.... */
mdct_forward(b->transform[vb->W][0],pcm,pcm);
/* FFT yields more accurate tonal estimation (not phase sensitive) */
drft_forward(&look->fft_look,fft);
fft[0]*=scale;
logfft[0]=todB(fft);
local_ampmax[i]=logfft[0];
for(j=1;j<n-1;j+=2){
float temp=scale*FAST_HYPOT(fft[j],fft[j+1]);
temp=logfft[(j+1)>>1]=todB(&temp);
if(temp>local_ampmax[i])local_ampmax[i]=temp;
}
if(local_ampmax[i]>global_ampmax)global_ampmax=local_ampmax[i];
_analysis_output("fft",seq+i,logfft,n/2,1,0);
}
for(i=0;i<vi->channels;i++){
int submap=info->chmuxlist[i];
/* the following makes things clearer to *me* anyway */
float *mdct =vb->pcm[i];
float *res =mdct;
float *codedflr=mdct+n/2;
float *logfft =mdct+n/2;
float *logmdct =work;
float *logmax =mdct+n/2;
float *logmask =work+n/2;
for(j=0;j<n/2;j++)
logmdct[j]=todB(mdct+j);
_analysis_output("mdct",seq+i,logmdct,n/2,1,0);
/* perform psychoacoustics; do masking */
_vp_compute_mask(look->psy_look[blocktype],
b->psy_g_look,
i,
logfft, /* -> logmax */
logmdct,
logmask,
global_ampmax,
local_ampmax[i],
ci->blocksizes[vb->lW]/2);
_analysis_output("mask",seq+i,logmask,n/2,1,0);
/* perform floor encoding */
nonzero[i]=look->floor_func[submap]->
forward(vb,look->floor_look[submap],
mdct,
logmdct,
logmask,
logmax,
codedflr);
_analysis_output("mdct2",seq+i,mdct,n/2,1,1);
_vp_remove_floor(look->psy_look[blocktype],
b->psy_g_look,
logmdct,
mdct,
codedflr,
res,
local_ampmax[i]);
/*for(j=0;j<n/2;j++)
if(fabs(res[j])>1200){
analysis_noisy=1;
fprintf(stderr,"%ld ",seq+i);
}*/
_analysis_output("res",seq+i,res,n/2,1,0);
_analysis_output("codedflr",seq+i,codedflr,n/2,1,1);
}
vbi->ampmax=global_ampmax;
/* partition based prequantization and channel coupling */
/* Steps in prequant and coupling:
down-couple/down-quantize from perfect residue -> quantized vector
classify by this first quantized vector
do{
encode quantized vector; add encoded values to 'so-far' vector
more? [not yet at bitrate/not yet at target]
yes{
down-couple/down-quantize from perfect-'so-far' ->
quantized vector; when subtracting coupling,
account for +/- out-of-phase component
}no{
break
}
}
done.
quantization in each iteration is done (after circular normalization
in coupling) using a by-iteration quantization granule value.
*/
{
float **pcm=vb->pcm;
float **quantized=alloca(sizeof(float*)*vi->channels);
float **sofar=alloca(sizeof(float*)*vi->channels);
long ***classifications=alloca(sizeof(long**)*info->submaps);
float ***pcmbundle=alloca(sizeof(float **)*info->submaps);
float ***sobundle=alloca(sizeof(float **)*info->submaps);
int **zerobundle=alloca(sizeof(int *)*info->submaps);
int *chbundle=alloca(sizeof(int)*info->submaps);
int chcounter=0;
/* play a little loose with this abstraction */
int quant_passes=look->psy_look[blocktype]->vi->coupling_passes;
int stopflag=0;
for(i=0;i<vi->channels;i++){
quantized[i]=pcm[i]+n/2;
sofar[i]=_vorbis_block_alloc(vb,n/2*sizeof(float));
memset(sofar[i],0,sizeof(float)*n/2);
}
pcmbundle[0]=alloca(sizeof(float *)*vi->channels);
sobundle[0]=alloca(sizeof(float *)*vi->channels);
zerobundle[0]=alloca(sizeof(int)*vi->channels);
/* initial down-quantized coupling */
if(info->coupling_steps==0){
/* this assumes all or nothing coupling right now. it should pass
through any channels left uncoupled, but it doesn't do that now */
for(i=0;i<vi->channels;i++){
float *lpcm=pcm[i];
float *lqua=quantized[i];
for(j=0;j<n/2;j++)
lqua[j]=lpcm[j];
}
}else{
_vp_quantize_couple(look->psy_look[blocktype],
info,
pcm,
sofar,
quantized,
nonzero,
0);
}
for(i=0;i<vi->channels;i++)
_analysis_output("quant",seq+i,quantized[i],n/2,1,0);
/* classify, by submap */
for(i=0;i<info->submaps;i++){
int ch_in_bundle=0;
pcmbundle[i]=pcmbundle[0]+chcounter;
sobundle[i]=sobundle[0]+chcounter;
zerobundle[i]=zerobundle[0]+chcounter;
for(j=0;j<vi->channels;j++){
if(info->chmuxlist[j]==i){
if(nonzero[j])
zerobundle[i][ch_in_bundle]=1;
else
zerobundle[i][ch_in_bundle]=0;
pcmbundle[i][ch_in_bundle]=quantized[j];
sobundle[i][ch_in_bundle++]=sofar[j];
}
}
chbundle[i]=ch_in_bundle;
chcounter+=ch_in_bundle;
classifications[i]=look->residue_func[i]->
class(vb,look->residue_look[i],pcmbundle[i],zerobundle[i],chbundle[i]);
}
/* actual encoding loop */
for(i=0;!stopflag;){
/* perform residue encoding of this pass's quantized residue
vector, according residue mapping */
for(j=0;j<info->submaps;j++)
look->residue_func[j]->
forward(vb,look->residue_look[j],
pcmbundle[j],sobundle[j],zerobundle[j],chbundle[j],
i,classifications[j]);
i++;
/* bitrate management decision hook; the following if() is where
we tell progressive encoding to halt, right now it just
avoids falling off the edge */
if(i>=quant_passes /* || yadda yadda */)stopflag=1;
if(!stopflag){
/* down-couple/down-quantize from perfect-'so-far' ->
new quantized vector */
if(info->coupling_steps==0){
/* this assumes all or nothing coupling right now. it should pass
through any channels left uncoupled, but it doesn't do that now */
for(i=0;i<vi->channels;i++){
float *lpcm=pcm[i];
float *lsof=sofar[i];
float *lqua=quantized[i];
for(j=0;j<n/2;j++)
lqua[j]=lpcm[j]-lsof[j];
}
}else{
_vp_quantize_couple(look->psy_look[blocktype],
info,
pcm,
sofar,
quantized,
nonzero,
i);
}
}
/* steady as she goes */
}
seq+=vi->channels;
}
look->lastframe=vb->sequence;
return(0);
}