diff --git a/src/add-ons/media/plugins/vorbis/Jamfile b/src/add-ons/media/plugins/vorbis/Jamfile new file mode 100644 index 0000000000..231f7593a6 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/Jamfile @@ -0,0 +1,14 @@ +SubDir OBOS_TOP src add-ons media plugins vorbis ; + +UsePrivateHeaders media ; + +SubDirHdrs $(SUBDIR) libvorbis vorbis ; + +Addon vorbis_decoder : media plugins : + vorbisCodecPlugin.cpp + : false : libvorbis.a +; + +LinkSharedOSLibs vorbis : be libmedia.so ; + +SubInclude OBOS_TOP src add-ons media plugins vorbis libvorbis ; diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/Jamfile b/src/add-ons/media/plugins/vorbis/libvorbis/Jamfile new file mode 100644 index 0000000000..aab22b8ccd --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/Jamfile @@ -0,0 +1,29 @@ +SubDir OBOS_TOP src add-ons media plugins vorbis_codec libvorbis ; + +SubDirHdrs [ FDirName $(SUBDIR) .. .. ogg_multistream libogg ] ; + +SubDirCcFlags -DPACKAGE=\"libvorbis\" -DVERSION=\"1.0\" -DHAVE_ALLOCA_H=1 -DHAVE_ALLOCA=1 -DHAVE_SQRTF=1 -DHAVE_LOGF=1 -DHAVE_EXPF=1 -DHAVE_ACOSF=1 -DHAVE_ATANF=1 -DHAVE_FREXPF=1 -DHAVE_RINTF=1 -DHAVE_FLOORF=1 ; +SubDirCcFlags -O20 -D__NO_MATH_INLINES -fsigned-char -DUSE_MEMORY_H -DPIC ; + +StaticLibrary vorbis : + analysis.c + bitrate.c + block.c + codebook.c + envelope.c + floor0.c + floor1.c + info.c + lookup.c + lpc.c + lsp.c + mapping0.c + mdct.c + psy.c + registry.c + res0.c + sharedbook.c + smallft.c + synthesis.c + window.c +; diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/analysis.c b/src/add-ons/media/plugins/vorbis/libvorbis/analysis.c new file mode 100644 index 0000000000..7193b6a364 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/analysis.c @@ -0,0 +1,119 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: single-block PCM analysis mode dispatch + last mod: $Id: analysis.c,v 1.1 2003/12/13 20:17:55 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "registry.h" +#include "scales.h" +#include "os.h" +#include "misc.h" + +int analysis_noisy=1; + +/* decides between modes, dispatches to the appropriate mapping. */ +int vorbis_analysis(vorbis_block *vb, ogg_packet *op){ + int ret; + + vb->glue_bits=0; + vb->time_bits=0; + vb->floor_bits=0; + vb->res_bits=0; + + /* first things first. Make sure encode is ready */ + oggpack_reset(&vb->opb); + + /* we only have one mapping type (0), and we let the mapping code + itself figure out what soft mode to use. This allows easier + bitrate management */ + + if((ret=_mapping_P[0]->forward(vb))) + return(ret); + + if(op){ + if(vorbis_bitrate_managed(vb)) + /* The app is using a bitmanaged mode... but not using the + bitrate management interface. */ + return(OV_EINVAL); + + op->packet=oggpack_get_buffer(&vb->opb); + op->bytes=oggpack_bytes(&vb->opb); + op->b_o_s=0; + op->e_o_s=vb->eofflag; + op->granulepos=vb->granulepos; + op->packetno=vb->sequence; /* for sake of completeness */ + } + return(0); +} + +/* there was no great place to put this.... */ +void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){ + int j; + FILE *of; + char buffer[80]; + + /* if(i==5870){*/ + sprintf(buffer,"%s_%d.m",base,i); + of=fopen(buffer,"w"); + + if(!of)perror("failed to open data dump file"); + + for(j=0;j +#include "scales.h" +int main(){ + int i; + double rate; + for(i=64;i<32000;i*=2){ + rate=48000.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=44100.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=32000.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=22050.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=16000.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=11025.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + rate=8000.f; + fprintf(stderr,"rate=%gHz, block=%d, f(1)=%.2gHz bark(1)=%.2g (of %.2g)\n\n", + rate,i,rate/2 / (i/2),toBARK(rate/2 /(i/2)),toBARK(rate/2)); + + + } + { + float i; + int j; + for(i=0.,j=0;i<28;i+=1,j++){ + fprintf(stderr,"(%d) bark=%f %gHz (%d of 128)\n", + j,i,fromBARK(i),(int)(fromBARK(i)/22050.*128.)); + } + } + return(0); +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.c b/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.c new file mode 100644 index 0000000000..be366479d9 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.c @@ -0,0 +1,531 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: bitrate tracking and management + last mod: $Id: bitrate.c,v 1.1 2003/12/13 20:17:55 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "os.h" +#include "misc.h" +#include "bitrate.h" + + +static long BINBYTES(bitrate_manager_state *bm,long pos,long bin){ + int bins=bm->queue_bins; + return(bm->queue_binned[pos*bins+bin]); +} + +#define LIMITBYTES(pos,bin) (bm->minmax_binstack[(pos)*bins*2+((bin)+bins)]) + +static long LACING_ADJUST(long bytes){ + int addto=bytes/255+1; + return(bytes+addto); +} + +static int floater_interpolate(bitrate_manager_state *bm,vorbis_info *vi, + double desired_rate){ + int bin=rint(bm->avgfloat); + double lobitrate,hibitrate; + + + lobitrate=(double)(bm->avg_binacc[bin]*8)/bm->avg_sampleacc*vi->rate; + while(lobitrate>desired_rate && bin>0){ + bin--; + lobitrate=(double)(bm->avg_binacc[bin]*8)/bm->avg_sampleacc*vi->rate; + } + + if(bin+1queue_bins){ + hibitrate=(double)(bm->avg_binacc[bin+1]*8)/bm->avg_sampleacc*vi->rate; + if(fabs(hibitrate-desired_rate) < fabs(lobitrate-desired_rate))bin++; + } + return(bin); +} + +/* try out a new limit */ +static long limit_sum(bitrate_manager_state *bm,int limit){ + int i=bm->minmax_stackptr; + long acc=bm->minmax_acctotal; + long bins=bm->queue_bins; + + acc-=LIMITBYTES(i,0); + acc+=LIMITBYTES(i,limit); + + while(i-->0){ + if(bm->minmax_limitstack[i]<=limit)break; + acc-=LIMITBYTES(i,bm->minmax_limitstack[i]); + acc+=LIMITBYTES(i,limit); + } + return(acc); +} + +/* compute bitrate tracking setup, allocate circular packet size queue */ +void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bm){ + int i; + codec_setup_info *ci=vi->codec_setup; + bitrate_manager_info *bi=&ci->bi; + long maxlatency; + + memset(bm,0,sizeof(*bm)); + + if(bi){ + + bm->avg_sampledesired=bi->queue_avg_time*vi->rate; + bm->avg_centerdesired=bi->queue_avg_time*vi->rate*bi->queue_avg_center; + bm->minmax_sampledesired=bi->queue_minmax_time*vi->rate; + + /* first find the max possible needed queue size */ + maxlatency=max(bm->avg_sampledesired-bm->avg_centerdesired, + bm->minmax_sampledesired)+bm->avg_centerdesired; + + if(maxlatency>0 && + (bi->queue_avgmin>0 || bi->queue_avgmax>0 || bi->queue_hardmax>0 || + bi->queue_hardmin>0)){ + long maxpackets=maxlatency/(ci->blocksizes[0]>>1)+3; + long bins=PACKETBLOBS; + + bm->queue_size=maxpackets; + bm->queue_bins=bins; + bm->queue_binned=_ogg_calloc(maxpackets,bins*sizeof(*bm->queue_binned)); + bm->queue_actual=_ogg_calloc(maxpackets,sizeof(*bm->queue_actual)); + + if((bi->queue_avgmin>0 || bi->queue_avgmax>0) && + bi->queue_avg_time>0){ + + bm->avg_binacc=_ogg_calloc(bins,sizeof(*bm->avg_binacc)); + bm->avgfloat=PACKETBLOBS/2; + + }else{ + bm->avg_tail= -1; + } + + if((bi->queue_hardmin>0 || bi->queue_hardmax>0) && + bi->queue_minmax_time>0){ + + bm->minmax_binstack=_ogg_calloc((bins*2+1)*bins*2, + sizeof(*bm->minmax_binstack)); + bm->minmax_posstack=_ogg_calloc((bins*2+1), + sizeof(*bm->minmax_posstack)); + bm->minmax_limitstack=_ogg_calloc((bins*2+1), + sizeof(*bm->minmax_limitstack)); + }else{ + bm->minmax_tail= -1; + } + + /* space for the packet queueing */ + bm->packetbuffers=_ogg_calloc(maxpackets,sizeof(*bm->packetbuffers)); + bm->packets=_ogg_calloc(maxpackets,sizeof(*bm->packets)); + for(i=0;ipacketbuffers+i); + + }else{ + bm->packetbuffers=_ogg_calloc(1,sizeof(*bm->packetbuffers)); + bm->packets=_ogg_calloc(1,sizeof(*bm->packets)); + oggpack_writeinit(bm->packetbuffers); + + } + } +} + +void vorbis_bitrate_clear(bitrate_manager_state *bm){ + int i; + if(bm){ + if(bm->queue_binned)_ogg_free(bm->queue_binned); + if(bm->queue_actual)_ogg_free(bm->queue_actual); + if(bm->avg_binacc)_ogg_free(bm->avg_binacc); + if(bm->minmax_binstack)_ogg_free(bm->minmax_binstack); + if(bm->minmax_posstack)_ogg_free(bm->minmax_posstack); + if(bm->minmax_limitstack)_ogg_free(bm->minmax_limitstack); + + if(bm->packetbuffers){ + if(bm->queue_size==0){ + oggpack_writeclear(bm->packetbuffers); + }else{ + for(i=0;iqueue_size;i++) + oggpack_writeclear(bm->packetbuffers+i); + } + _ogg_free(bm->packetbuffers); + } + if(bm->packets)_ogg_free(bm->packets); + + memset(bm,0,sizeof(*bm)); + } +} + +int vorbis_bitrate_managed(vorbis_block *vb){ + vorbis_dsp_state *vd=vb->vd; + backend_lookup_state *b=vd->backend_state; + bitrate_manager_state *bm=&b->bms; + + if(bm->queue_binned)return(1); + return(0); +} + +/* finish taking in the block we just processed */ +int vorbis_bitrate_addblock(vorbis_block *vb){ + int i; + vorbis_block_internal *vbi=vb->internal; + vorbis_dsp_state *vd=vb->vd; + backend_lookup_state *b=vd->backend_state; + bitrate_manager_state *bm=&b->bms; + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + bitrate_manager_info *bi=&ci->bi; + int eofflag=vb->eofflag; + int head=bm->queue_head; + int next_head=head+1; + int bins=bm->queue_bins; + int minmax_head,new_minmax_head; + + ogg_uint32_t *head_ptr; + oggpack_buffer temp; + + if(!bm->queue_binned){ + oggpack_buffer temp; + /* not a bitrate managed stream, but for API simplicity, we'll + buffer one packet to keep the code path clean */ + + if(bm->queue_head)return(-1); /* one has been submitted without + being claimed */ + bm->queue_head++; + + bm->packets[0].packet=oggpack_get_buffer(&vb->opb); + bm->packets[0].bytes=oggpack_bytes(&vb->opb); + bm->packets[0].b_o_s=0; + bm->packets[0].e_o_s=vb->eofflag; + bm->packets[0].granulepos=vb->granulepos; + bm->packets[0].packetno=vb->sequence; /* for sake of completeness */ + + memcpy(&temp,bm->packetbuffers,sizeof(vb->opb)); + memcpy(bm->packetbuffers,&vb->opb,sizeof(vb->opb)); + memcpy(&vb->opb,&temp,sizeof(vb->opb)); + + return(0); + } + + /* add encoded packet to head */ + if(next_head>=bm->queue_size)next_head=0; + head_ptr=bm->queue_binned+bins*head; + + /* is there room to add a block? In proper use of the API, this will + never come up... but guard it anyway */ + if(next_head==bm->avg_tail || next_head==bm->minmax_tail)return(-1); + + /* add the block to the toplevel queue */ + bm->queue_head=next_head; + bm->queue_actual[head]=(vb->W?0x80000000UL:0); + + /* buffer packet fields */ + bm->packets[head].packet=oggpack_get_buffer(&vb->opb); + bm->packets[head].bytes=oggpack_bytes(&vb->opb); + bm->packets[head].b_o_s=0; + bm->packets[head].e_o_s=vb->eofflag; + bm->packets[head].granulepos=vb->granulepos; + bm->packets[head].packetno=vb->sequence; /* for sake of completeness */ + + /* swap packet buffers */ + memcpy(&temp,bm->packetbuffers+head,sizeof(vb->opb)); + memcpy(bm->packetbuffers+head,&vb->opb,sizeof(vb->opb)); + memcpy(&vb->opb,&temp,sizeof(vb->opb)); + + /* save markers */ + head_ptr[0]=vbi->packetblob_markers[0]; + for(i=1;ipacketblob_markers[i]-vbi->packetblob_markers[i-1]; + } + + if(bm->avg_binacc) + new_minmax_head=minmax_head=bm->avg_center; + else + new_minmax_head=minmax_head=head; + + /* the average tracking queue is updated first; its results (if it's + in use) are taken into account by the min/max limiter (if min/max + is in use) */ + if(bm->avg_binacc){ + unsigned long desired_center=bm->avg_centerdesired; + if(eofflag)desired_center=0; + + /* update the avg head */ + for(i=0;iavg_binacc[i]+=LACING_ADJUST(head_ptr[i]); + bm->avg_sampleacc+=ci->blocksizes[vb->W]>>1; + bm->avg_centeracc+=ci->blocksizes[vb->W]>>1; + + if(bm->avg_sampleacc>bm->avg_sampledesired || eofflag){ + + /* update the avg center */ + if(bm->avg_centeracc>desired_center){ + /* choose the new average floater */ + int samples=ci->blocksizes[vb->W]>>1; + double upper=floater_interpolate(bm,vi,bi->queue_avgmax); + double lower=floater_interpolate(bm,vi,bi->queue_avgmin); + double new=PACKETBLOBS/2.,slew; + int bin; + + if(uppernew)new=lower; + + slew=(new-bm->avgfloat)/samples*vi->rate; + + if(slewavgfloat_downslew_max) + new=bm->avgfloat+bi->avgfloat_downslew_max/vi->rate*samples; + if(slew>bi->avgfloat_upslew_max) + new=bm->avgfloat+bi->avgfloat_upslew_max/vi->rate*samples; + + bm->avgfloat=new; + /* apply the average floater to new blocks */ + bin=rint(bm->avgfloat); + + /*fprintf(stderr,"%d ",bin);*/ + + while(bm->avg_centeracc>desired_center){ + samples=ci->blocksizes[bm->queue_actual[bm->avg_center]& + 0x80000000UL?1:0]>>1; + + bm->queue_actual[bm->avg_center]|=bin; + + bm->avg_centeracc-=samples; + bm->avg_center++; + if(bm->avg_center>=bm->queue_size)bm->avg_center=0; + } + new_minmax_head=bm->avg_center; + + } + + /* update the avg tail if needed */ + while(bm->avg_sampleacc>bm->avg_sampledesired){ + int samples= + ci->blocksizes[bm->queue_actual[bm->avg_tail]&0x80000000UL?1:0]>>1; + for(i=0;iqueue_bins;i++) + bm->avg_binacc[i]-=LACING_ADJUST(bm->queue_binned[bins*bm->avg_tail+i]); + bm->avg_sampleacc-=samples; + bm->avg_tail++; + if(bm->avg_tail>=bm->queue_size)bm->avg_tail=0; + } + + + } + }else{ + /* if we're not using an average tracker, the 'float' is nailed to + the avgfloat_initial value. It needs to be set for the min/max + to deal properly */ + long bin=PACKETBLOBS/2; + bm->queue_actual[head]|=bin; + new_minmax_head=next_head; + } + + /* update the min/max queues and enforce limits */ + if(bm->minmax_binstack){ + unsigned long sampledesired=eofflag?0:bm->minmax_sampledesired; + + /* add to stack recent */ + while(minmax_head!=new_minmax_head){ + unsigned int i; + int samples=ci->blocksizes[bm->queue_actual[minmax_head]& + 0x80000000UL?1:0]>>1; + int actual=bm->queue_actual[minmax_head]&0x7fffffffUL; + + for(i=0;i<(unsigned int)bins;i++){ + bm->minmax_binstack[bm->minmax_stackptr*bins*2+bins+i]+= + LACING_ADJUST(BINBYTES(bm,minmax_head, + actual>i?actual:i)); + + bm->minmax_binstack[bm->minmax_stackptr*bins*2+i]+= + LACING_ADJUST(BINBYTES(bm,minmax_head, + actualminmax_posstack[bm->minmax_stackptr]=minmax_head; /* not one + past + like + typical */ + bm->minmax_limitstack[bm->minmax_stackptr]=0; + bm->minmax_sampleacc+=samples; + bm->minmax_acctotal+= + LACING_ADJUST(BINBYTES(bm,minmax_head,actual)); + + minmax_head++; + if(minmax_head>=bm->queue_size)minmax_head=0; + + + } + + /* check limits, enforce changes */ + if(bm->minmax_sampleacc>sampledesired){ + double bitrate=(double)(bm->minmax_acctotal*8)/ + bm->minmax_sampleacc*vi->rate; + int limit=0; + + if((bi->queue_hardmax>0 && bitrate>bi->queue_hardmax) || + (bi->queue_hardmin>0 && bitratequeue_hardmin)){ + int newstack; + int stackctr; + long bitsum=bm->minmax_acctotal*8; + + bitrate=(double)bitsum/bm->minmax_sampleacc*vi->rate; + + /* we're off rate. Iteratively try out new hard floater + limits until we find one that brings us inside. Here's + where we see the whole point of the limit stacks. */ + if(bi->queue_hardmax>0 && bitrate>bi->queue_hardmax){ + for(limit=-1;limit>-bins+1;limit--){ + long bitsum=limit_sum(bm,limit)*8; + bitrate=(double)bitsum/bm->minmax_sampleacc*vi->rate; + if(bitrate<=bi->queue_hardmax)break; + } + }else if(bitratequeue_hardmin){ + for(limit=1;limitminmax_sampleacc*vi->rate; + if(bitrate>=bi->queue_hardmin)break; + } + if(bitrate>bi->queue_hardmax)limit--; + } + + /* trace the limit backward, stop when we see a lower limit */ + newstack=bm->minmax_stackptr-1; + while(newstack>=0){ + if(bm->minmax_limitstack[newstack]minmax_stackptr; + while(stackctr>newstack){ + bm->minmax_acctotal-= + LIMITBYTES(stackctr,bm->minmax_limitstack[stackctr]); + bm->minmax_acctotal+=LIMITBYTES(stackctr,limit); + + if(stackctrminmax_stackptr) + for(i=0;iminmax_binstack[stackctr*bins*2+i]+= + bm->minmax_binstack[(stackctr+1)*bins*2+i]; + + stackctr--; + } + stackctr++; + bm->minmax_posstack[stackctr]=bm->minmax_posstack[bm->minmax_stackptr]; + bm->minmax_limitstack[stackctr]=limit; + + /* set up new blank stack entry */ + stackctr++; + bm->minmax_stackptr=stackctr; + memset(&bm->minmax_binstack[stackctr*bins*2], + 0, + sizeof(*bm->minmax_binstack)*bins*2); + bm->minmax_limitstack[stackctr]=0; + bm->minmax_posstack[stackctr]=-1; + + } + } + + /* remove from tail */ + while(bm->minmax_sampleacc>sampledesired){ + int samples= + ci->blocksizes[bm->queue_actual[bm->minmax_tail]&0x80000000UL?1:0]>>1; + int actual=bm->queue_actual[bm->minmax_tail]&0x7fffffffUL; + + for(i=0;iminmax_binstack[bins+i]-= /* always comes off the stack bottom */ + LACING_ADJUST(BINBYTES(bm,bm->minmax_tail, + actual>i? + actual:i)); + bm->minmax_binstack[i]-= + LACING_ADJUST(BINBYTES(bm,bm->minmax_tail, + actualminmax_limitstack[0]>actual) + actual=bm->minmax_limitstack[0]; + if(bins+bm->minmax_limitstack[0]minmax_limitstack[0]; + + bm->minmax_acctotal-=LACING_ADJUST(BINBYTES(bm,bm->minmax_tail,actual)); + bm->minmax_sampleacc-=samples; + + /* revise queue_actual to reflect the limit */ + bm->queue_actual[bm->minmax_tail]&=0x80000000UL; + bm->queue_actual[bm->minmax_tail]|=actual; + + if(bm->minmax_tail==bm->minmax_posstack[0]){ + /* the stack becomes a FIFO; the first data has fallen off */ + memmove(bm->minmax_binstack,bm->minmax_binstack+bins*2, + sizeof(*bm->minmax_binstack)*bins*2*bm->minmax_stackptr); + memmove(bm->minmax_posstack,bm->minmax_posstack+1, + sizeof(*bm->minmax_posstack)*bm->minmax_stackptr); + memmove(bm->minmax_limitstack,bm->minmax_limitstack+1, + sizeof(*bm->minmax_limitstack)*bm->minmax_stackptr); + bm->minmax_stackptr--; + } + + bm->minmax_tail++; + if(bm->minmax_tail>=bm->queue_size)bm->minmax_tail=0; + + } + + + bm->last_to_flush=bm->minmax_tail; + }else{ + bm->last_to_flush=bm->avg_center; + } + if(eofflag) + bm->last_to_flush=bm->queue_head; + return(0); +} + +int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,ogg_packet *op){ + backend_lookup_state *b=vd->backend_state; + bitrate_manager_state *bm=&b->bms; + + if(bm->queue_size==0){ + if(bm->queue_head==0)return(0); + + memcpy(op,bm->packets,sizeof(*op)); + bm->queue_head=0; + + }else{ + + if(bm->next_to_flush==bm->last_to_flush)return(0); + + { + long bin=bm->queue_actual[bm->next_to_flush]&0x7fffffff,i; + long bins=bm->queue_bins; + ogg_uint32_t *markers=bm->queue_binned+bins*bm->next_to_flush; + long bytes=markers[bin]; + + memcpy(op,bm->packets+bm->next_to_flush,sizeof(*op)); + + /* we have [PACKETBLOBS] possible packets all squished together in + the buffer, in sequence. count in to number [bin] */ + for(i=0;ipacket+=markers[i]; + op->bytes=bytes; + + } + + bm->next_to_flush++; + if(bm->next_to_flush>=bm->queue_size)bm->next_to_flush=0; + + } + + return(1); +} diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.h b/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.h new file mode 100644 index 0000000000..c4f0d70be3 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/bitrate.h @@ -0,0 +1,84 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: bitrate tracking and management + last mod: $Id: bitrate.h,v 1.1 2003/12/13 20:17:55 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_BITRATE_H_ +#define _V_BITRATE_H_ + +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "os.h" + +/* encode side bitrate tracking */ +typedef struct bitrate_manager_state { + ogg_uint32_t *queue_binned; + ogg_uint32_t *queue_actual; + int queue_size; + + int queue_head; + int queue_bins; + + long *avg_binacc; + int avg_center; + int avg_tail; + ogg_uint32_t avg_centeracc; + ogg_uint32_t avg_sampleacc; + ogg_uint32_t avg_sampledesired; + ogg_uint32_t avg_centerdesired; + + long *minmax_binstack; + long *minmax_posstack; + long *minmax_limitstack; + long minmax_stackptr; + + long minmax_acctotal; + int minmax_tail; + ogg_uint32_t minmax_sampleacc; + ogg_uint32_t minmax_sampledesired; + + int next_to_flush; + int last_to_flush; + + double avgfloat; + + /* unfortunately, we need to hold queued packet data somewhere */ + oggpack_buffer *packetbuffers; + ogg_packet *packets; + +} bitrate_manager_state; + +typedef struct bitrate_manager_info{ + /* detailed bitrate management setup */ + double queue_avg_time; + double queue_avg_center; + double queue_minmax_time; + double queue_hardmin; + double queue_hardmax; + double queue_avgmin; + double queue_avgmax; + + double avgfloat_downslew_max; + double avgfloat_upslew_max; + +} bitrate_manager_info; + +extern void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bs); +extern void vorbis_bitrate_clear(bitrate_manager_state *bs); +extern int vorbis_bitrate_managed(vorbis_block *vb); +extern int vorbis_bitrate_addblock(vorbis_block *vb); +extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, ogg_packet *op); + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/block.c b/src/add-ons/media/plugins/vorbis/libvorbis/block.c new file mode 100644 index 0000000000..376014ef99 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/block.c @@ -0,0 +1,833 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: PCM data vector blocking, windowing and dis/reassembly + last mod: $Id: block.c,v 1.1 2003/12/13 20:17:55 shatty Exp $ + + Handle windowing, overlap-add, etc of the PCM vectors. This is made + more amusing by Vorbis' current two allowed block sizes. + + ********************************************************************/ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" + +#include "window.h" +#include "mdct.h" +#include "lpc.h" +#include "registry.h" +#include "misc.h" + +static int ilog2(unsigned int v){ + int ret=0; + if(v)--v; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +/* pcm accumulator examples (not exhaustive): + + <-------------- lW ----------------> + <--------------- W ----------------> +: .....|..... _______________ | +: .''' | '''_--- | |\ | +:.....''' |_____--- '''......| | \_______| +:.................|__________________|_______|__|______| + |<------ Sl ------>| > Sr < |endW + |beginSl |endSl | |endSr + |beginW |endlW |beginSr + + + |< lW >| + <--------------- W ----------------> + | | .. ______________ | + | | ' `/ | ---_ | + |___.'___/`. | ---_____| + |_______|__|_______|_________________| + | >|Sl|< |<------ Sr ----->|endW + | | |endSl |beginSr |endSr + |beginW | |endlW + mult[0] |beginSl mult[n] + + <-------------- lW -----------------> + |<--W-->| +: .............. ___ | | +: .''' |`/ \ | | +:.....''' |/`....\|...| +:.........................|___|___|___| + |Sl |Sr |endW + | | |endSr + | |beginSr + | |endSl + |beginSl + |beginW +*/ + +/* block abstraction setup *********************************************/ + +#ifndef WORD_ALIGN +#define WORD_ALIGN 8 +#endif + +int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb){ + memset(vb,0,sizeof(*vb)); + vb->vd=v; + vb->localalloc=0; + vb->localstore=NULL; + if(v->analysisp){ + vorbis_block_internal *vbi= + vb->internal=_ogg_calloc(1,sizeof(vorbis_block_internal)); + oggpack_writeinit(&vb->opb); + vbi->ampmax=-9999; + } + + return(0); +} + +void *_vorbis_block_alloc(vorbis_block *vb,long bytes){ + bytes=(bytes+(WORD_ALIGN-1)) & ~(WORD_ALIGN-1); + if(bytes+vb->localtop>vb->localalloc){ + /* can't just _ogg_realloc... there are outstanding pointers */ + if(vb->localstore){ + struct alloc_chain *link=_ogg_malloc(sizeof(*link)); + vb->totaluse+=vb->localtop; + link->next=vb->reap; + link->ptr=vb->localstore; + vb->reap=link; + } + /* highly conservative */ + vb->localalloc=bytes; + vb->localstore=_ogg_malloc(vb->localalloc); + vb->localtop=0; + } + { + void *ret=(void *)(((char *)vb->localstore)+vb->localtop); + vb->localtop+=bytes; + return ret; + } +} + +/* reap the chain, pull the ripcord */ +void _vorbis_block_ripcord(vorbis_block *vb){ + /* reap the chain */ + struct alloc_chain *reap=vb->reap; + while(reap){ + struct alloc_chain *next=reap->next; + _ogg_free(reap->ptr); + memset(reap,0,sizeof(*reap)); + _ogg_free(reap); + reap=next; + } + /* consolidate storage */ + if(vb->totaluse){ + vb->localstore=_ogg_realloc(vb->localstore,vb->totaluse+vb->localalloc); + vb->localalloc+=vb->totaluse; + vb->totaluse=0; + } + + /* pull the ripcord */ + vb->localtop=0; + vb->reap=NULL; +} + +int vorbis_block_clear(vorbis_block *vb){ + if(vb->vd) + if(vb->vd->analysisp) + oggpack_writeclear(&vb->opb); + _vorbis_block_ripcord(vb); + if(vb->localstore)_ogg_free(vb->localstore); + + if(vb->internal) + _ogg_free(vb->internal); + + memset(vb,0,sizeof(*vb)); + return(0); +} + +/* Analysis side code, but directly related to blocking. Thus it's + here and not in analysis.c (which is for analysis transforms only). + The init is here because some of it is shared */ + +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; + + memset(v,0,sizeof(*v)); + b=v->backend_state=_ogg_calloc(1,sizeof(*b)); + + v->vi=vi; + b->modebits=ilog2(ci->modes); + + b->transform[0]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[0])); + b->transform[1]=_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[1])); + + /* MDCT is tranform 0 */ + + 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]); + + /* 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); + + 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;ibooks;i++) + vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]); + } + + b->psy=_ogg_calloc(ci->psys,sizeof(*b->psy)); + for(i=0;ipsys;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;ibooks;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. 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)); + { + int i; + for(i=0;ichannels;i++) + v->pcm[i]=_ogg_calloc(v->pcm_storage,sizeof(*v->pcm[i])); + } + + /* all 1 (large block) or 0 (small block) */ + /* explicitly set for the sake of clarity */ + v->lW=0; /* previous window size */ + v->W=0; /* current window size */ + + /* all vector indexes */ + v->centerW=ci->blocksizes[1]/2; + + v->pcm_current=v->centerW; + + /* initialize all the backend lookups */ + b->flr=_ogg_calloc(ci->floors,sizeof(*b->flr)); + b->residue=_ogg_calloc(ci->residues,sizeof(*b->residue)); + + for(i=0;ifloors;i++) + b->flr[i]=_floor_P[ci->floor_type[i]]-> + look(v,ci->floor_param[i]); + + for(i=0;iresidues;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; + + _vds_shared_init(v,vi,1); + b=v->backend_state; + b->psy_g_look=_vp_global_look(vi); + + /* Initialize the envelope state storage */ + b->ve=_ogg_calloc(1,sizeof(*b->ve)); + _ve_envelope_init(b->ve,vi); + + vorbis_bitrate_init(vi,&b->bms); + + return(0); +} + +void vorbis_dsp_clear(vorbis_dsp_state *v){ + 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; + + 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); + } + + if(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]); + } + + if(b->flr){ + for(i=0;ifloors;i++) + _floor_P[ci->floor_type[i]]-> + free_look(b->flr[i]); + _ogg_free(b->flr); + } + if(b->residue){ + for(i=0;iresidues;i++) + _residue_P[ci->residue_type[i]]-> + free_look(b->residue[i]); + _ogg_free(b->residue); + } + if(b->psy){ + for(i=0;ipsys;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;ichannels;i++) + if(v->pcm[i])_ogg_free(v->pcm[i]); + _ogg_free(v->pcm); + if(v->pcmret)_ogg_free(v->pcmret); + } + + if(b){ + /* 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; + + /* free header, header1, header2 */ + if(b->header)_ogg_free(b->header);b->header=NULL; + if(b->header1)_ogg_free(b->header1);b->header1=NULL; + if(b->header2)_ogg_free(b->header2);b->header2=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;ichannels;i++){ + v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i])); + } + } + + for(i=0;ichannels;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; + float *lpc=alloca(order*sizeof(*lpc)); + float *work=alloca(v->pcm_current*sizeof(*work)); + long j; + v->preextrapolate=1; + + if(v->pcm_current-v->centerW>order*2){ /* safety */ + for(i=0;ivi->channels;i++){ + /* need to run the extrapolation in reverse! */ + for(j=0;jpcm_current;j++) + 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); + + /* run the predictor filter */ + vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order, + order, + work+v->pcm_current-v->centerW, + v->centerW); + + for(j=0;jpcm_current;j++) + v->pcm[i][v->pcm_current-j-1]=work[j]; + + } + } +} + + +/* call with val<=0 to set eof */ + +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; + int i; + float *lpc=alloca(order*sizeof(*lpc)); + + /* if it wasn't done earlier (very short sample) */ + if(!v->preextrapolate) + _preextrapolate_helper(v); + + /* We're encoding the end of the stream. Just make sure we have + [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]*3); + v->eofflag=v->pcm_current; + v->pcm_current+=ci->blocksizes[1]*3; + + for(i=0;ichannels;i++){ + if(v->eofflag>order*2){ + /* 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); + + /* 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 + 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])); + + } + } + }else{ + + if(v->pcm_current+vals>v->pcm_storage) + return(OV_EINVAL); + + v->pcm_current+=vals; + + /* we may want to reverse extrapolate the beginning of a stream + too... in case we're beginning on a cliff! */ + /* clumsy, but simple. It only runs once, so simple is good. */ + if(!v->preextrapolate && v->pcm_current-v->centerW>ci->blocksizes[1]) + _preextrapolate_helper(v); + + } + return(0); +} + +/* do the deltas, envelope shaping, pre-echo and determine the size of + the next block on which to continue analysis */ +int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb){ + int i; + vorbis_info *vi=v->vi; + codec_setup_info *ci=vi->codec_setup; + backend_lookup_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; + + /* check to see if we're started... */ + if(!v->preextrapolate)return(0); + + /* check to see if we're done... */ + if(v->eofflag==-1)return(0); + + /* 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 */ + + /* 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){ + + 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; + } + } + + centerNext=v->centerW+ci->blocksizes[v->W]/4+ci->blocksizes[v->nW]/4; + + { + /* center of next block + next block maximum right side. */ + + long blockbound=centerNext+ci->blocksizes[v->nW]/2; + if(v->pcm_currentlW=v->lW; + vb->W=v->W; + vb->nW=v->nW; + + if(v->W){ + if(!v->lW || !v->nW){ + vbi->blocktype=BLOCKTYPE_TRANSITION; + /*fprintf(stderr,"-");*/ + }else{ + vbi->blocktype=BLOCKTYPE_LONG; + /*fprintf(stderr,"_");*/ + } + }else{ + if(_ve_envelope_mark(v)){ + vbi->blocktype=BLOCKTYPE_IMPULSE; + /*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 */ + /* 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;ichannels;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 + eof>0 marks the last 'real' sample in pcm[] + eof<0 'no more to do'; doesn't get here */ + + if(v->eofflag){ + if(v->centerW>=v->eofflag){ + v->eofflag=-1; + vb->eofflag=1; + return(1); + } + } + + /* advance storage vectors and clean up */ + { + 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;ichannels;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; + 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; + } + } + } + + /* done */ + return(1); +} + +int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){ + _vds_shared_init(v,vi,0); + + v->pcm_returned=-1; + v->granulepos=-1; + v->sequence=-1; + + 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; + 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 */ + 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; + + 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 */ + + /* overlap/add PCM */ + + for(j=0;jchannels;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;ipcm[j]+prevCenter+n1/2-n0/2; + float *p=vb->pcm[j]; + for(i=0;iW){ + /* small/large */ + float *pcm=v->pcm[j]+prevCenter; + float *p=vb->pcm[j]+n1/2-n0/2; + for(i=0;ipcm[j]+prevCenter; + float *p=vb->pcm[j]; + for(i=0;ipcm[j]+thisCenter; + float *p=vb->pcm[j]+n; + for(i=0;icenterW) + 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; + } + + } + + /* 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){ /* only set if we have a position to set to */ + 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(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 } */ + } /* else {Shouldn't happen *unless* the bitstream is out of + 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_returnedpcm_current){ + if(pcm){ + int i; + for(i=0;ichannels;i++) + v->pcmret[i]=v->pcm[i]+v->pcm_returned; + *pcm=v->pcmret; + } + return(v->pcm_current-v->pcm_returned); + } + return(0); +} + +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); +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/codebook.c b/src/add-ons/media/plugins/vorbis/libvorbis/codebook.c new file mode 100644 index 0000000000..97dd3e12f4 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/codebook.c @@ -0,0 +1,615 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: basic codebook pack/unpack/code/decode operations + last mod: $Id: codebook.c,v 1.1 2003/12/13 20:17:55 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codebook.h" +#include "scales.h" +#include "misc.h" +#include "os.h" + +/* packs the given codebook into the bitstream **************************/ + +int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *opb){ + long i,j; + int ordered=0; + + /* first the basic parameters */ + oggpack_write(opb,0x564342,24); + oggpack_write(opb,c->dim,16); + oggpack_write(opb,c->entries,24); + + /* pack the codewords. There are two packings; length ordered and + length random. Decide between the two now. */ + + for(i=1;ientries;i++) + if(c->lengthlist[i-1]==0 || c->lengthlist[i]lengthlist[i-1])break; + if(i==c->entries)ordered=1; + + if(ordered){ + /* length ordered. We only need to say how many codewords of + each length. The actual codewords are generated + deterministically */ + + long count=0; + oggpack_write(opb,1,1); /* ordered */ + oggpack_write(opb,c->lengthlist[0]-1,5); /* 1 to 32 */ + + for(i=1;ientries;i++){ + long this=c->lengthlist[i]; + long last=c->lengthlist[i-1]; + if(this>last){ + for(j=last;jentries-count)); + count=i; + } + } + } + oggpack_write(opb,i-count,_ilog(c->entries-count)); + + }else{ + /* length random. Again, we don't code the codeword itself, just + the length. This time, though, we have to encode each length */ + oggpack_write(opb,0,1); /* unordered */ + + /* algortihmic mapping has use for 'unused entries', which we tag + here. The algorithmic mapping happens as usual, but the unused + entry has no codeword. */ + for(i=0;ientries;i++) + if(c->lengthlist[i]==0)break; + + if(i==c->entries){ + oggpack_write(opb,0,1); /* no unused entries */ + for(i=0;ientries;i++) + oggpack_write(opb,c->lengthlist[i]-1,5); + }else{ + oggpack_write(opb,1,1); /* we have unused entries; thus we tag */ + for(i=0;ientries;i++){ + if(c->lengthlist[i]==0){ + oggpack_write(opb,0,1); + }else{ + oggpack_write(opb,1,1); + oggpack_write(opb,c->lengthlist[i]-1,5); + } + } + } + } + + /* is the entry number the desired return value, or do we have a + mapping? If we have a mapping, what type? */ + oggpack_write(opb,c->maptype,4); + switch(c->maptype){ + case 0: + /* no mapping */ + break; + case 1:case 2: + /* implicitly populated value mapping */ + /* explicitly populated value mapping */ + + if(!c->quantlist){ + /* no quantlist? error */ + return(-1); + } + + /* values that define the dequantization */ + oggpack_write(opb,c->q_min,32); + oggpack_write(opb,c->q_delta,32); + oggpack_write(opb,c->q_quant-1,4); + oggpack_write(opb,c->q_sequencep,1); + + { + int quantvals; + switch(c->maptype){ + case 1: + /* a single column of (c->entries/c->dim) quantized values for + building a full value list algorithmically (square lattice) */ + quantvals=_book_maptype1_quantvals(c); + break; + case 2: + /* every value (c->entries*c->dim total) specified explicitly */ + quantvals=c->entries*c->dim; + break; + default: /* NOT_REACHABLE */ + quantvals=-1; + } + + /* quantized values */ + for(i=0;iquantlist[i]),c->q_quant); + + } + break; + default: + /* error case; we don't have any other map types now */ + return(-1); + } + + return(0); +} + +/* unpacks a codebook from the packet buffer into the codebook struct, + readies the codebook auxiliary structures for decode *************/ +int vorbis_staticbook_unpack(oggpack_buffer *opb,static_codebook *s){ + long i,j; + memset(s,0,sizeof(*s)); + s->allocedp=1; + + /* make sure alignment is correct */ + if(oggpack_read(opb,24)!=0x564342)goto _eofout; + + /* first the basic parameters */ + s->dim=oggpack_read(opb,16); + s->entries=oggpack_read(opb,24); + if(s->entries==-1)goto _eofout; + + /* codeword ordering.... length ordered or unordered? */ + switch((int)oggpack_read(opb,1)){ + case 0: + /* unordered */ + s->lengthlist=_ogg_malloc(sizeof(*s->lengthlist)*s->entries); + + /* allocated but unused entries? */ + if(oggpack_read(opb,1)){ + /* yes, unused entries */ + + for(i=0;ientries;i++){ + if(oggpack_read(opb,1)){ + long num=oggpack_read(opb,5); + if(num==-1)goto _eofout; + s->lengthlist[i]=num+1; + }else + s->lengthlist[i]=0; + } + }else{ + /* all entries used; no tagging */ + for(i=0;ientries;i++){ + long num=oggpack_read(opb,5); + if(num==-1)goto _eofout; + s->lengthlist[i]=num+1; + } + } + + break; + case 1: + /* ordered */ + { + long length=oggpack_read(opb,5)+1; + s->lengthlist=_ogg_malloc(sizeof(*s->lengthlist)*s->entries); + + for(i=0;ientries;){ + long num=oggpack_read(opb,_ilog(s->entries-i)); + if(num==-1)goto _eofout; + for(j=0;jentries;j++,i++) + s->lengthlist[i]=length; + length++; + } + } + break; + default: + /* EOF */ + return(-1); + } + + /* Do we have a mapping to unpack? */ + switch((s->maptype=oggpack_read(opb,4))){ + case 0: + /* no mapping */ + break; + case 1: case 2: + /* implicitly populated value mapping */ + /* explicitly populated value mapping */ + + s->q_min=oggpack_read(opb,32); + s->q_delta=oggpack_read(opb,32); + s->q_quant=oggpack_read(opb,4)+1; + s->q_sequencep=oggpack_read(opb,1); + + { + int quantvals=0; + switch(s->maptype){ + case 1: + quantvals=_book_maptype1_quantvals(s); + break; + case 2: + quantvals=s->entries*s->dim; + break; + } + + /* quantized values */ + s->quantlist=_ogg_malloc(sizeof(*s->quantlist)*quantvals); + for(i=0;iquantlist[i]=oggpack_read(opb,s->q_quant); + + if(quantvals&&s->quantlist[quantvals-1]==-1)goto _eofout; + } + break; + default: + goto _errout; + } + + /* all set */ + return(0); + + _errout: + _eofout: + vorbis_staticbook_clear(s); + return(-1); +} + +/* returns the number of bits ************************************************/ +int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b){ + oggpack_write(b,book->codelist[a],book->c->lengthlist[a]); + return(book->c->lengthlist[a]); +} + +/* One the encode side, our vector writers are each designed for a +specific purpose, and the encoder is not flexible without modification: + +The LSP vector coder uses a single stage nearest-match with no +interleave, so no step and no error return. This is specced by floor0 +and doesn't change. + +Residue0 encoding interleaves, uses multiple stages, and each stage +peels of a specific amount of resolution from a lattice (thus we want +to match by threshold, not nearest match). Residue doesn't *have* to +be encoded that way, but to change it, one will need to add more +infrastructure on the encode side (decode side is specced and simpler) */ + +/* floor0 LSP (single stage, non interleaved, nearest match) */ +/* returns entry number and *modifies a* to the quantization value *****/ +int vorbis_book_errorv(codebook *book,float *a){ + int dim=book->dim,k; + int best=_best(book,a,1); + for(k=0;kvaluelist+best*dim)[k]; + return(best); +} + +/* returns the number of bits and *modifies a* to the quantization value *****/ +int vorbis_book_encodev(codebook *book,int best,float *a,oggpack_buffer *b){ + int k,dim=book->dim; + for(k=0;kvaluelist+best*dim)[k]; + return(vorbis_book_encode(book,best,b)); +} + +/* the 'eliminate the decode tree' optimization actually requires the + codewords to be MSb first, not LSb. This is an annoying inelegancy + (and one of the first places where carefully thought out design + turned out to be wrong; Vorbis II and future Ogg codecs should go + to an MSb bitpacker), but not actually the huge hit it appears to + be. The first-stage decode table catches most words so that + bitreverse is not in the main execution path. */ + +static ogg_uint32_t bitreverse(ogg_uint32_t x){ + x= ((x>>16)&0x0000ffff) | ((x<<16)&0xffff0000); + x= ((x>> 8)&0x00ff00ff) | ((x<< 8)&0xff00ff00); + x= ((x>> 4)&0x0f0f0f0f) | ((x<< 4)&0xf0f0f0f0); + x= ((x>> 2)&0x33333333) | ((x<< 2)&0xcccccccc); + return((x>> 1)&0x55555555) | ((x<< 1)&0xaaaaaaaa); +} + +STIN long decode_packed_entry_number(codebook *book, oggpack_buffer *b){ + int read=book->dec_maxlength; + long lo,hi; + long lok = oggpack_look(b,book->dec_firsttablen); + + if (lok >= 0) { + long entry = book->dec_firsttable[lok]; + if(entry&0x80000000UL){ + lo=(entry>>15)&0x7fff; + hi=book->used_entries-(entry&0x7fff); + }else{ + oggpack_adv(b, book->dec_codelengths[entry-1]); + return(entry-1); + } + }else{ + lo=0; + hi=book->used_entries; + } + + lok = oggpack_look(b, read); + + while(lok<0 && read>1) + lok = oggpack_look(b, --read); + if(lok<0)return -1; + + /* bisect search for the codeword in the ordered list */ + { + ogg_uint32_t testword=bitreverse((ogg_uint32_t)lok); + + while(hi-lo>1){ + long p=(hi-lo)>>1; + long test=book->codelist[lo+p]>testword; + lo+=p&(test-1); + hi-=p&(-test); + } + + if(book->dec_codelengths[lo]<=read){ + oggpack_adv(b, book->dec_codelengths[lo]); + return(lo); + } + } + + oggpack_adv(b, read); + return(-1); +} + +/* Decode side is specced and easier, because we don't need to find + matches using different criteria; we simply read and map. There are + two things we need to do 'depending': + + We may need to support interleave. We don't really, but it's + convenient to do it here rather than rebuild the vector later. + + Cascades may be additive or multiplicitive; this is not inherent in + the codebook, but set in the code using the codebook. Like + interleaving, it's easiest to do it here. + addmul==0 -> declarative (set the value) + addmul==1 -> additive + addmul==2 -> multiplicitive */ + +/* returns the [original, not compacted] entry number or -1 on eof *********/ +long vorbis_book_decode(codebook *book, oggpack_buffer *b){ + long packed_entry=decode_packed_entry_number(book,b); + if(packed_entry>=0) + return(book->dec_index[packed_entry]); + + /* if there's no dec_index, the codebook unpacking isn't collapsed */ + return(packed_entry); +} + +/* returns 0 on OK or -1 on eof *************************************/ +long vorbis_book_decodevs_add(codebook *book,float *a,oggpack_buffer *b,int n){ + int step=n/book->dim; + long *entry = alloca(sizeof(*entry)*step); + float **t = alloca(sizeof(*t)*step); + int i,j,o; + + for (i = 0; i < step; i++) { + entry[i]=decode_packed_entry_number(book,b); + if(entry[i]==-1)return(-1); + t[i] = book->valuelist+entry[i]*book->dim; + } + for(i=0,o=0;idim;i++,o+=step) + for (j=0;jdim>8){ + for(i=0;ivaluelist+entry*book->dim; + for (j=0;jdim;) + a[i++]+=t[j++]; + } + }else{ + for(i=0;ivaluelist+entry*book->dim; + j=0; + switch((int)book->dim){ + case 8: + a[i++]+=t[j++]; + case 7: + a[i++]+=t[j++]; + case 6: + a[i++]+=t[j++]; + case 5: + a[i++]+=t[j++]; + case 4: + a[i++]+=t[j++]; + case 3: + a[i++]+=t[j++]; + case 2: + a[i++]+=t[j++]; + case 1: + a[i++]+=t[j++]; + case 0: + break; + } + } + } + return(0); +} + +long vorbis_book_decodev_set(codebook *book,float *a,oggpack_buffer *b,int n){ + int i,j,entry; + float *t; + + for(i=0;ivaluelist+entry*book->dim; + for (j=0;jdim;) + a[i++]=t[j++]; + } + return(0); +} + +long vorbis_book_decodevv_add(codebook *book,float **a,long offset,int ch, + oggpack_buffer *b,int n){ + long i,j,entry; + int chptr=0; + + for(i=offset/ch;i<(offset+n)/ch;){ + entry = decode_packed_entry_number(book,b); + if(entry==-1)return(-1); + { + const float *t = book->valuelist+entry*book->dim; + for (j=0;jdim;j++){ + a[chptr++][i]+=t[j]; + if(chptr==ch){ + chptr=0; + i++; + } + } + } + } + return(0); +} + +#ifdef _V_SELFTEST +/* Simple enough; pack a few candidate codebooks, unpack them. Code a + number of vectors through (keeping track of the quantized values), + and decode using the unpacked book. quantized version of in should + exactly equal out */ + +#include + +#include "vorbis/book/lsp20_0.vqh" +#include "vorbis/book/res0a_13.vqh" +#define TESTSIZE 40 + +float test1[TESTSIZE]={ + 0.105939f, + 0.215373f, + 0.429117f, + 0.587974f, + + 0.181173f, + 0.296583f, + 0.515707f, + 0.715261f, + + 0.162327f, + 0.263834f, + 0.342876f, + 0.406025f, + + 0.103571f, + 0.223561f, + 0.368513f, + 0.540313f, + + 0.136672f, + 0.395882f, + 0.587183f, + 0.652476f, + + 0.114338f, + 0.417300f, + 0.525486f, + 0.698679f, + + 0.147492f, + 0.324481f, + 0.643089f, + 0.757582f, + + 0.139556f, + 0.215795f, + 0.324559f, + 0.399387f, + + 0.120236f, + 0.267420f, + 0.446940f, + 0.608760f, + + 0.115587f, + 0.287234f, + 0.571081f, + 0.708603f, +}; + +float test3[TESTSIZE]={ + 0,1,-2,3,4,-5,6,7,8,9, + 8,-2,7,-1,4,6,8,3,1,-9, + 10,11,12,13,14,15,26,17,18,19, + 30,-25,-30,-1,-5,-32,4,3,-2,0}; + +static_codebook *testlist[]={&_vq_book_lsp20_0, + &_vq_book_res0a_13,NULL}; +float *testvec[]={test1,test3}; + +int main(){ + oggpack_buffer write; + oggpack_buffer read; + long ptr=0,i; + oggpack_writeinit(&write); + + fprintf(stderr,"Testing codebook abstraction...:\n"); + + while(testlist[ptr]){ + codebook c; + static_codebook s; + float *qv=alloca(sizeof(*qv)*TESTSIZE); + float *iv=alloca(sizeof(*iv)*TESTSIZE); + memcpy(qv,testvec[ptr],sizeof(*qv)*TESTSIZE); + memset(iv,0,sizeof(*iv)*TESTSIZE); + + fprintf(stderr,"\tpacking/coding %ld... ",ptr); + + /* pack the codebook, write the testvector */ + oggpack_reset(&write); + vorbis_book_init_encode(&c,testlist[ptr]); /* get it into memory + we can write */ + vorbis_staticbook_pack(testlist[ptr],&write); + fprintf(stderr,"Codebook size %ld bytes... ",oggpack_bytes(&write)); + for(i=0;i.000001){ + fprintf(stderr,"read (%g) != written (%g) at position (%ld)\n", + iv[i],qv[i],i); + exit(1); + } + + fprintf(stderr,"OK\n"); + ptr++; + } + + /* The above is the trivial stuff; now try unquantizing a log scale codebook */ + + exit(0); +} + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/codebook.h b/src/add-ons/media/plugins/vorbis/libvorbis/codebook.h new file mode 100644 index 0000000000..f71592b4e6 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/codebook.h @@ -0,0 +1,160 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: basic shared codebook operations + last mod: $Id: codebook.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_CODEBOOK_H_ +#define _V_CODEBOOK_H_ + +#include + +/* This structure encapsulates huffman and VQ style encoding books; it + doesn't do anything specific to either. + + valuelist/quantlist are nonNULL (and q_* significant) only if + there's entry->value mapping to be done. + + If encode-side mapping must be done (and thus the entry needs to be + hunted), the auxiliary encode pointer will point to a decision + tree. This is true of both VQ and huffman, but is mostly useful + with VQ. + +*/ + +typedef struct static_codebook{ + long dim; /* codebook dimensions (elements per vector) */ + long entries; /* codebook entries */ + long *lengthlist; /* codeword lengths in bits */ + + /* mapping ***************************************************************/ + int maptype; /* 0=none + 1=implicitly populated values from map column + 2=listed arbitrary values */ + + /* The below does a linear, single monotonic sequence mapping. */ + long q_min; /* packed 32 bit float; quant value 0 maps to minval */ + long q_delta; /* packed 32 bit float; val 1 - val 0 == delta */ + int q_quant; /* bits: 0 < quant <= 16 */ + int q_sequencep; /* bitflag */ + + long *quantlist; /* map == 1: (int)(entries^(1/dim)) element column map + map == 2: list of dim*entries quantized entry vals + */ + + /* encode helpers ********************************************************/ + struct encode_aux_nearestmatch *nearest_tree; + struct encode_aux_threshmatch *thresh_tree; + struct encode_aux_pigeonhole *pigeon_tree; + + int allocedp; +} static_codebook; + +/* this structures an arbitrary trained book to quickly find the + nearest cell match */ +typedef struct encode_aux_nearestmatch{ + /* pre-calculated partitioning tree */ + long *ptr0; + long *ptr1; + + long *p; /* decision points (each is an entry) */ + long *q; /* decision points (each is an entry) */ + long aux; /* number of tree entries */ + long alloc; +} encode_aux_nearestmatch; + +/* assumes a maptype of 1; encode side only, so that's OK */ +typedef struct encode_aux_threshmatch{ + float *quantthresh; + long *quantmap; + int quantvals; + int threshvals; +} encode_aux_threshmatch; + +typedef struct encode_aux_pigeonhole{ + float min; + float del; + + int mapentries; + int quantvals; + long *pigeonmap; + + long fittotal; + long *fitlist; + long *fitmap; + long *fitlength; +} encode_aux_pigeonhole; + +typedef struct codebook{ + long dim; /* codebook dimensions (elements per vector) */ + long entries; /* codebook entries */ + long used_entries; /* populated codebook entries */ + const static_codebook *c; + + /* for encode, the below are entry-ordered, fully populated */ + /* for decode, the below are ordered by bitreversed codeword and only + used entries are populated */ + float *valuelist; /* list of dim*entries actual entry values */ + ogg_uint32_t *codelist; /* list of bitstream codewords for each entry */ + + int *dec_index; /* only used if sparseness collapsed */ + char *dec_codelengths; + ogg_uint32_t *dec_firsttable; + int dec_firsttablen; + int dec_maxlength; + +} codebook; + +extern void vorbis_staticbook_clear(static_codebook *b); +extern void vorbis_staticbook_destroy(static_codebook *b); +extern int vorbis_book_init_encode(codebook *dest,const static_codebook *source); +extern int vorbis_book_init_decode(codebook *dest,const static_codebook *source); +extern void vorbis_book_clear(codebook *b); + +extern float *_book_unquantize(const static_codebook *b,int n,int *map); +extern float *_book_logdist(const static_codebook *b,float *vals); +extern float _float32_unpack(long val); +extern long _float32_pack(float val); +extern int _best(codebook *book, float *a, int step); +extern int _ilog(unsigned int v); +extern long _book_maptype1_quantvals(const static_codebook *b); + +extern int vorbis_book_besterror(codebook *book,float *a,int step,int addmul); +extern long vorbis_book_codeword(codebook *book,int entry); +extern long vorbis_book_codelen(codebook *book,int entry); + + + +extern int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *b); +extern int vorbis_staticbook_unpack(oggpack_buffer *b,static_codebook *c); + +extern int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b); +extern int vorbis_book_errorv(codebook *book, float *a); +extern int vorbis_book_encodev(codebook *book, int best,float *a, + oggpack_buffer *b); + +extern long vorbis_book_decode(codebook *book, oggpack_buffer *b); +extern long vorbis_book_decodevs_add(codebook *book, float *a, + oggpack_buffer *b,int n); +extern long vorbis_book_decodev_set(codebook *book, float *a, + oggpack_buffer *b,int n); +extern long vorbis_book_decodev_add(codebook *book, float *a, + oggpack_buffer *b,int n); +extern long vorbis_book_decodevv_add(codebook *book, float **a, + long off,int ch, + oggpack_buffer *b,int n); + + + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/codec_internal.h b/src/add-ons/media/plugins/vorbis/libvorbis/codec_internal.h new file mode 100644 index 0000000000..c3a2330eca --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/codec_internal.h @@ -0,0 +1,133 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: libvorbis codec headers + last mod: $Id: codec_internal.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_CODECI_H_ +#define _V_CODECI_H_ + +#include "envelope.h" +#include "codebook.h" + +#define BLOCKTYPE_IMPULSE 0 +#define BLOCKTYPE_PADDING 1 +#define BLOCKTYPE_TRANSITION 0 +#define BLOCKTYPE_LONG 1 + +#define PACKETBLOBS 15 + +typedef struct vorbis_block_internal{ + float **pcmdelay; /* this is a pointer into local storage */ + float ampmax; + int blocktype; + + ogg_uint32_t packetblob_markers[PACKETBLOBS]; +} vorbis_block_internal; + +typedef void vorbis_look_floor; +typedef void vorbis_look_residue; +typedef void vorbis_look_transform; + +/* mode ************************************************************/ +typedef struct { + int blockflag; + int windowtype; + int transformtype; + int mapping; +} vorbis_info_mode; + +typedef void vorbis_info_floor; +typedef void vorbis_info_residue; +typedef void vorbis_info_mapping; + +#include "psy.h" +#include "bitrate.h" + +typedef struct backend_lookup_state { + /* local lookup storage */ + envelope_lookup *ve; /* envelope lookup */ + float *window[2]; + vorbis_look_transform **transform[2]; /* block, type */ + drft_lookup fft_look[2]; + + int modebits; + vorbis_look_floor **flr; + vorbis_look_residue **residue; + vorbis_look_psy *psy; + vorbis_look_psy_global *psy_g_look; + + /* local storage, only used on the encoding side. This way the + application does not need to worry about freeing some packets' + memory and not others'; packet storage is always tracked. + Cleared next call to a _dsp_ function */ + unsigned char *header; + unsigned char *header1; + unsigned char *header2; + + bitrate_manager_state bms; + +} backend_lookup_state; + +/* codec_setup_info contains all the setup information specific to the + specific compression/decompression mode in progress (eg, + psychoacoustic settings, channel setup, options, codebook + etc). +*********************************************************************/ + +#include "highlevel.h" +typedef struct codec_setup_info { + + /* Vorbis supports only short and long blocks, but allows the + encoder to choose the sizes */ + + long blocksizes[2]; + + /* modes are the primary means of supporting on-the-fly different + blocksizes, different channel mappings (LR or M/A), + different residue backends, etc. Each mode consists of a + blocksize flag and a mapping (along with the mapping setup */ + + int modes; + int maps; + int floors; + int residues; + int books; + int psys; /* encode only */ + + vorbis_info_mode *mode_param[64]; + int map_type[64]; + vorbis_info_mapping *map_param[64]; + int floor_type[64]; + vorbis_info_floor *floor_param[64]; + int residue_type[64]; + vorbis_info_residue *residue_param[64]; + static_codebook *book_param[256]; + codebook *fullbooks; + + vorbis_info_psy *psy_param[4]; /* encode only */ + vorbis_info_psy_global psy_g_param; + + bitrate_manager_info bi; + highlevel_encode_setup hi; /* used only by vorbisenc.c. It's a + highly redundant structure, but + improves clarity of program flow. */ + +} codec_setup_info; + +extern vorbis_look_psy_global *_vp_global_look(vorbis_info *vi); +extern void _vp_global_free(vorbis_look_psy_global *look); + +#endif + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/envelope.c b/src/add-ons/media/plugins/vorbis/libvorbis/envelope.c new file mode 100644 index 0000000000..bd6537eb4d --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/envelope.c @@ -0,0 +1,383 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: PCM data envelope analysis + last mod: $Id: envelope.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" + +#include "os.h" +#include "scales.h" +#include "envelope.h" +#include "mdct.h" +#include "misc.h" + +void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi){ + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *gi=&ci->psy_g_param; + int ch=vi->channels; + int i,j; + int n=e->winlength=128; + e->searchstep=64; /* not random */ + + e->minenergy=gi->preecho_minenergy; + e->ch=ch; + e->storage=128; + e->cursor=ci->blocksizes[1]/2; + e->mdct_win=_ogg_calloc(n,sizeof(*e->mdct_win)); + mdct_init(&e->mdct,n); + + for(i=0;imdct_win[i]=sin(i/(n-1.)*M_PI); + e->mdct_win[i]*=e->mdct_win[i]; + } + + /* magic follows */ + e->band[0].begin=2; e->band[0].end=4; + e->band[1].begin=4; e->band[1].end=5; + e->band[2].begin=6; e->band[2].end=6; + e->band[3].begin=9; e->band[3].end=8; + e->band[4].begin=13; e->band[4].end=8; + e->band[5].begin=17; e->band[5].end=8; + e->band[6].begin=22; e->band[6].end=8; + + for(j=0;jband[j].end; + e->band[j].window=_ogg_malloc(n*sizeof(*e->band[0].window)); + for(i=0;iband[j].window[i]=sin((i+.5)/n*M_PI); + e->band[j].total+=e->band[j].window[i]; + } + e->band[j].total=1./e->band[j].total; + } + + e->filter=_ogg_calloc(VE_BANDS*ch,sizeof(*e->filter)); + e->mark=_ogg_calloc(e->storage,sizeof(*e->mark)); + +} + +void _ve_envelope_clear(envelope_lookup *e){ + int i; + mdct_clear(&e->mdct); + for(i=0;iband[i].window); + _ogg_free(e->mdct_win); + _ogg_free(e->filter); + _ogg_free(e->mark); + memset(e,0,sizeof(*e)); +} + +/* fairly straight threshhold-by-band based until we find something + that works better and isn't patented. */ + +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;imdct_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>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_AMP)filters[j].ampptr=0; + } + + /* look at min/max, decide trigger */ + if(valmax>gi->preecho_thresh[j]+penalty){ + ret|=1; + ret|=4; + } + if(valminpostecho_thresh[j]-penalty)ret|=2; + } + + return(ret); +} + +#if 0 +static int seq=0; +static ogg_int64_t totalshift=-1024; +#endif + +long _ve_envelope_search(vorbis_dsp_state *v){ + vorbis_info *vi=v->vi; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *gi=&ci->psy_g_param; + envelope_lookup *ve=((backend_lookup_state *)(v->backend_state))->ve; + long i,j; + + int first=ve->current/ve->searchstep; + int last=v->pcm_current/ve->searchstep-VE_WIN; + if(first<0)first=0; + + /* make sure we have enough storage to match the PCM */ + if(last+VE_WIN+VE_POST>ve->storage){ + ve->storage=last+VE_WIN+VE_POST; /* be sure */ + ve->mark=_ogg_realloc(ve->mark,ve->storage*sizeof(*ve->mark)); + } + + for(j=first;jstretch++; + if(ve->stretch>VE_MAXSTRETCH*2) + ve->stretch=VE_MAXSTRETCH*2; + + for(i=0;ich;i++){ + float *pcm=v->pcm[i]+ve->searchstep*(j); + ret|=_ve_amp(ve,gi,pcm,ve->band,ve->filter+i*VE_BANDS,j); + } + + ve->mark[j+VE_POST]=0; + if(ret&1){ + ve->mark[j]=1; + ve->mark[j+1]=1; + } + + if(ret&2){ + ve->mark[j]=1; + if(j>0)ve->mark[j-1]=1; + } + + if(ret&4)ve->stretch=-1; + } + + ve->current=last*ve->searchstep; + + { + long centerW=v->centerW; + long testW= + centerW+ + ci->blocksizes[v->W]/4+ + ci->blocksizes[1]/2+ + ci->blocksizes[0]/4; + + j=ve->cursor; + + while(jcurrent-(ve->searchstep)){/* account for postecho + working back one window */ + if(j>=testW)return(1); + + ve->cursor=j; + + if(ve->mark[j/ve->searchstep]){ + if(j>centerW){ + + #if 0 + if(j>ve->curmark){ + float *marker=alloca(v->pcm_current*sizeof(*marker)); + int l,m; + memset(marker,0,sizeof(*marker)*v->pcm_current); + fprintf(stderr,"mark! seq=%d, cursor:%fs time:%fs\n", + seq, + (totalshift+ve->cursor)/44100., + (totalshift+j)/44100.); + _analysis_output_always("pcmL",seq,v->pcm[0],v->pcm_current,0,0,totalshift); + _analysis_output_always("pcmR",seq,v->pcm[1],v->pcm_current,0,0,totalshift); + + _analysis_output_always("markL",seq,v->pcm[0],j,0,0,totalshift); + _analysis_output_always("markR",seq,v->pcm[1],j,0,0,totalshift); + + for(m=0;msearchstep]=ve->filter[m].markers[l]*.1; + _analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift); + } + + for(m=0;msearchstep]=ve->filter[m+VE_BANDS].markers[l]*.1; + _analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift); + } + + for(l=0;lsearchstep]=ve->mark[l]*.4; + _analysis_output_always("mark",seq,marker,v->pcm_current,0,0,totalshift); + + + seq++; + + } +#endif + + ve->curmark=j; + if(j>=testW)return(1); + return(0); + } + } + j+=ve->searchstep; + } + } + + return(-1); +} + +int _ve_envelope_mark(vorbis_dsp_state *v){ + envelope_lookup *ve=((backend_lookup_state *)(v->backend_state))->ve; + vorbis_info *vi=v->vi; + codec_setup_info *ci=vi->codec_setup; + long centerW=v->centerW; + long beginW=centerW-ci->blocksizes[v->W]/4; + long endW=centerW+ci->blocksizes[v->W]/4; + if(v->W){ + beginW-=ci->blocksizes[v->lW]/4; + endW+=ci->blocksizes[v->nW]/4; + }else{ + beginW-=ci->blocksizes[0]/4; + endW+=ci->blocksizes[0]/4; + } + + if(ve->curmark>=beginW && ve->curmarksearchstep; + long last=endW/ve->searchstep; + long i; + for(i=first;imark[i])return(1); + } + return(0); +} + +void _ve_envelope_shift(envelope_lookup *e,long shift){ + int smallsize=e->current/e->searchstep+VE_POST; /* adjust for placing marks + ahead of ve->current */ + int smallshift=shift/e->searchstep; + int i; + + memmove(e->mark,e->mark+smallshift,(smallsize-smallshift)*sizeof(*e->mark)); + + #if 0 + for(i=0;ich;i++) + memmove(e->filter[i].markers, + e->filter[i].markers+smallshift, + (1024-smallshift)*sizeof(*(*e->filter).markers)); + totalshift+=shift; + #endif + + e->current-=shift; + if(e->curmark>=0) + e->curmark-=shift; + e->cursor-=shift; +} + + + + + + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/envelope.h b/src/add-ons/media/plugins/vorbis/libvorbis/envelope.h new file mode 100644 index 0000000000..2cc9c6fbc4 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/envelope.h @@ -0,0 +1,81 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: PCM data envelope analysis and manipulation + last mod: $Id: envelope.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_ENVELOPE_ +#define _V_ENVELOPE_ + +#include "mdct.h" + +#define VE_PRE 16 +#define VE_WIN 4 +#define VE_POST 2 +#define VE_AMP (VE_PRE+VE_POST-1) + +#define VE_BANDS 7 +#define VE_NEARDC 15 + +#define VE_MINSTRETCH 2 /* a bit less than short block */ +#define VE_MAXSTRETCH 12 /* one-third full block */ + +typedef struct { + float ampbuf[VE_AMP]; + int ampptr; + + float nearDC[VE_NEARDC]; + float nearDC_acc; + float nearDC_partialacc; + int nearptr; + +} envelope_filter_state; + +typedef struct { + int begin; + int end; + float *window; + float total; +} envelope_band; + +typedef struct { + int ch; + int winlength; + int searchstep; + float minenergy; + + mdct_lookup mdct; + float *mdct_win; + + envelope_band band[VE_BANDS]; + envelope_filter_state *filter; + int stretch; + + int *mark; + + long storage; + long current; + long curmark; + long cursor; +} envelope_lookup; + +extern void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi); +extern void _ve_envelope_clear(envelope_lookup *e); +extern long _ve_envelope_search(vorbis_dsp_state *v); +extern void _ve_envelope_shift(envelope_lookup *e,long shift); +extern int _ve_envelope_mark(vorbis_dsp_state *v); + + +#endif + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/floor0.c b/src/add-ons/media/plugins/vorbis/libvorbis/floor0.c new file mode 100644 index 0000000000..bb021fa3ef --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/floor0.c @@ -0,0 +1,235 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: floor backend 0 implementation + last mod: $Id: floor0.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "registry.h" +#include "lpc.h" +#include "lsp.h" +#include "codebook.h" +#include "scales.h" +#include "misc.h" +#include "os.h" + +#include "misc.h" +#include + +typedef struct { + int ln; + int m; + int **linearmap; + int n[2]; + + vorbis_info_floor0 *vi; + lpc_lookup lpclook; + float *lsp_look; + + long bits; + long frames; +} vorbis_look_floor0; + + +/***********************************************/ + +static void floor0_free_info(vorbis_info_floor *i){ + vorbis_info_floor0 *info=(vorbis_info_floor0 *)i; + if(info){ + memset(info,0,sizeof(*info)); + _ogg_free(info); + } +} + +static void floor0_free_look(vorbis_look_floor *i){ + vorbis_look_floor0 *look=(vorbis_look_floor0 *)i; + if(look){ + + if(look->linearmap){ + + if(look->linearmap[0])_ogg_free(look->linearmap[0]); + if(look->linearmap[1])_ogg_free(look->linearmap[1]); + + _ogg_free(look->linearmap); + } + if(look->lsp_look)_ogg_free(look->lsp_look); + lpc_clear(&look->lpclook); + memset(look,0,sizeof(*look)); + _ogg_free(look); + } +} + +static vorbis_info_floor *floor0_unpack (vorbis_info *vi,oggpack_buffer *opb){ + codec_setup_info *ci=vi->codec_setup; + int j; + + vorbis_info_floor0 *info=_ogg_malloc(sizeof(*info)); + info->order=oggpack_read(opb,8); + info->rate=oggpack_read(opb,16); + info->barkmap=oggpack_read(opb,16); + info->ampbits=oggpack_read(opb,6); + info->ampdB=oggpack_read(opb,8); + info->numbooks=oggpack_read(opb,4)+1; + + if(info->order<1)goto err_out; + if(info->rate<1)goto err_out; + if(info->barkmap<1)goto err_out; + if(info->numbooks<1)goto err_out; + + for(j=0;jnumbooks;j++){ + info->books[j]=oggpack_read(opb,8); + if(info->books[j]<0 || info->books[j]>=ci->books)goto err_out; + } + return(info); + + err_out: + floor0_free_info(info); + return(NULL); +} + +/* initialize Bark scale and normalization lookups. We could do this + with static tables, but Vorbis allows a number of possible + combinations, so it's best to do it computationally. + + The below is authoritative in terms of defining scale mapping. + Note that the scale depends on the sampling rate as well as the + linear block and mapping sizes */ + +static void floor0_map_lazy_init(vorbis_block *vb, + vorbis_info_floor *infoX, + vorbis_look_floor0 *look){ + if(!look->linearmap[vb->W]){ + vorbis_dsp_state *vd=vb->vd; + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_floor0 *info=(vorbis_info_floor0 *)infoX; + int W=vb->W; + int n=ci->blocksizes[W]/2,j; + + /* we choose a scaling constant so that: + floor(bark(rate/2-1)*C)=mapped-1 + floor(bark(rate/2)*C)=mapped */ + float scale=look->ln/toBARK(info->rate/2.f); + + /* the mapping from a linear scale to a smaller bark scale is + straightforward. We do *not* make sure that the linear mapping + does not skip bark-scale bins; the decoder simply skips them and + the encoder may do what it wishes in filling them. They're + necessary in some mapping combinations to keep the scale spacing + accurate */ + look->linearmap[W]=_ogg_malloc((n+1)*sizeof(**look->linearmap)); + for(j=0;jrate/2.f)/n*j) + *scale); /* bark numbers represent band edges */ + if(val>=look->ln)val=look->ln-1; /* guard against the approximation */ + look->linearmap[W][j]=val; + } + look->linearmap[W][j]=-1; + look->n[W]=n; + } +} + +static vorbis_look_floor *floor0_look(vorbis_dsp_state *vd, + vorbis_info_floor *i){ + int j; + vorbis_info_floor0 *info=(vorbis_info_floor0 *)i; + vorbis_look_floor0 *look=_ogg_calloc(1,sizeof(*look)); + look->m=info->order; + look->ln=info->barkmap; + look->vi=info; + + if(vd->analysisp) + lpc_init(&look->lpclook,look->ln,look->m); + + look->linearmap=_ogg_calloc(2,sizeof(*look->linearmap)); + + look->lsp_look=_ogg_malloc(look->ln*sizeof(*look->lsp_look)); + for(j=0;jln;j++) + look->lsp_look[j]=2*cos(M_PI/look->ln*j); + + return look; +} + +static void *floor0_inverse1(vorbis_block *vb,vorbis_look_floor *i){ + vorbis_look_floor0 *look=(vorbis_look_floor0 *)i; + vorbis_info_floor0 *info=look->vi; + int j,k; + + int ampraw=oggpack_read(&vb->opb,info->ampbits); + if(ampraw>0){ /* also handles the -1 out of data case */ + long maxval=(1<ampbits)-1; + float amp=(float)ampraw/maxval*info->ampdB; + int booknum=oggpack_read(&vb->opb,_ilog(info->numbooks)); + + if(booknum!=-1 && booknumnumbooks){ /* be paranoid */ + codec_setup_info *ci=vb->vd->vi->codec_setup; + codebook *b=ci->fullbooks+info->books[booknum]; + float last=0.f; + + /* the additional b->dim is a guard against any possible stack + smash; b->dim is provably more than we can overflow the + vector */ + float *lsp=_vorbis_block_alloc(vb,sizeof(*lsp)*(look->m+b->dim+1)); + + for(j=0;jm;j+=b->dim) + if(vorbis_book_decodev_set(b,lsp+j,&vb->opb,b->dim)==-1)goto eop; + for(j=0;jm;){ + for(k=0;kdim;k++,j++)lsp[j]+=last; + last=lsp[j-1]; + } + + lsp[look->m]=amp; + return(lsp); + } + } + eop: + return(NULL); +} + +static int floor0_inverse2(vorbis_block *vb,vorbis_look_floor *i, + void *memo,float *out){ + vorbis_look_floor0 *look=(vorbis_look_floor0 *)i; + vorbis_info_floor0 *info=look->vi; + + floor0_map_lazy_init(vb,info,look); + + if(memo){ + float *lsp=(float *)memo; + float amp=lsp[look->m]; + + /* take the coefficients back to a spectral envelope curve */ + vorbis_lsp_to_curve(out, + look->linearmap[vb->W], + look->n[vb->W], + look->ln, + lsp,look->m,amp,(float)info->ampdB); + return(1); + } + memset(out,0,sizeof(*out)*look->n[vb->W]); + return(0); +} + +/* export hooks */ +vorbis_func_floor floor0_exportbundle={ + NULL,&floor0_unpack,&floor0_look,&floor0_free_info, + &floor0_free_look,&floor0_inverse1,&floor0_inverse2 +}; + + + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/floor1.c b/src/add-ons/media/plugins/vorbis/libvorbis/floor1.c new file mode 100644 index 0000000000..e2a9fd5112 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/floor1.c @@ -0,0 +1,1087 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: floor backend 1 implementation + last mod: $Id: floor1.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "registry.h" +#include "codebook.h" +#include "misc.h" +#include "scales.h" + +#include + +#define floor1_rangedB 140 /* floor 1 fixed at -140dB to 0dB range */ + +typedef struct { + int sorted_index[VIF_POSIT+2]; + int forward_index[VIF_POSIT+2]; + int reverse_index[VIF_POSIT+2]; + + int hineighbor[VIF_POSIT]; + int loneighbor[VIF_POSIT]; + int posts; + + int n; + int quant_q; + vorbis_info_floor1 *vi; + + long phrasebits; + long postbits; + long frames; +} vorbis_look_floor1; + +typedef struct lsfit_acc{ + long x0; + long x1; + + long xa; + long ya; + long x2a; + long y2a; + long xya; + long n; + long an; +} lsfit_acc; + +/***********************************************/ + +static void floor1_free_info(vorbis_info_floor *i){ + vorbis_info_floor1 *info=(vorbis_info_floor1 *)i; + if(info){ + memset(info,0,sizeof(*info)); + _ogg_free(info); + } +} + +static void floor1_free_look(vorbis_look_floor *i){ + vorbis_look_floor1 *look=(vorbis_look_floor1 *)i; + if(look){ + /*fprintf(stderr,"floor 1 bit usage %f:%f (%f total)\n", + (float)look->phrasebits/look->frames, + (float)look->postbits/look->frames, + (float)(look->postbits+look->phrasebits)/look->frames);*/ + + memset(look,0,sizeof(*look)); + _ogg_free(look); + } +} + +static int ilog(unsigned int v){ + int ret=0; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +static int ilog2(unsigned int v){ + int ret=0; + if(v)--v; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +static void floor1_pack (vorbis_info_floor *i,oggpack_buffer *opb){ + vorbis_info_floor1 *info=(vorbis_info_floor1 *)i; + int j,k; + int count=0; + int rangebits; + int maxposit=info->postlist[1]; + int maxclass=-1; + + /* save out partitions */ + oggpack_write(opb,info->partitions,5); /* only 0 to 31 legal */ + for(j=0;jpartitions;j++){ + oggpack_write(opb,info->partitionclass[j],4); /* only 0 to 15 legal */ + if(maxclasspartitionclass[j])maxclass=info->partitionclass[j]; + } + + /* save out partition classes */ + for(j=0;jclass_dim[j]-1,3); /* 1 to 8 */ + oggpack_write(opb,info->class_subs[j],2); /* 0 to 3 */ + if(info->class_subs[j])oggpack_write(opb,info->class_book[j],8); + for(k=0;k<(1<class_subs[j]);k++) + oggpack_write(opb,info->class_subbook[j][k]+1,8); + } + + /* save out the post list */ + oggpack_write(opb,info->mult-1,2); /* only 1,2,3,4 legal now */ + oggpack_write(opb,ilog2(maxposit),4); + rangebits=ilog2(maxposit); + + for(j=0,k=0;jpartitions;j++){ + count+=info->class_dim[info->partitionclass[j]]; + for(;kpostlist[k+2],rangebits); + } +} + + +static vorbis_info_floor *floor1_unpack (vorbis_info *vi,oggpack_buffer *opb){ + codec_setup_info *ci=vi->codec_setup; + int j,k,count=0,maxclass=-1,rangebits; + + vorbis_info_floor1 *info=_ogg_calloc(1,sizeof(*info)); + /* read partitions */ + info->partitions=oggpack_read(opb,5); /* only 0 to 31 legal */ + for(j=0;jpartitions;j++){ + info->partitionclass[j]=oggpack_read(opb,4); /* only 0 to 15 legal */ + if(maxclasspartitionclass[j])maxclass=info->partitionclass[j]; + } + + /* read partition classes */ + for(j=0;jclass_dim[j]=oggpack_read(opb,3)+1; /* 1 to 8 */ + info->class_subs[j]=oggpack_read(opb,2); /* 0,1,2,3 bits */ + if(info->class_subs[j]<0) + goto err_out; + if(info->class_subs[j])info->class_book[j]=oggpack_read(opb,8); + if(info->class_book[j]<0 || info->class_book[j]>=ci->books) + goto err_out; + for(k=0;k<(1<class_subs[j]);k++){ + info->class_subbook[j][k]=oggpack_read(opb,8)-1; + if(info->class_subbook[j][k]<-1 || info->class_subbook[j][k]>=ci->books) + goto err_out; + } + } + + /* read the post list */ + info->mult=oggpack_read(opb,2)+1; /* only 1,2,3,4 legal now */ + rangebits=oggpack_read(opb,4); + + for(j=0,k=0;jpartitions;j++){ + count+=info->class_dim[info->partitionclass[j]]; + for(;kpostlist[k+2]=oggpack_read(opb,rangebits); + if(t<0 || t>=(1<postlist[0]=0; + info->postlist[1]=1<vi=info; + look->n=info->postlist[1]; + + /* we drop each position value in-between already decoded values, + and use linear interpolation to predict each new value past the + edges. The positions are read in the order of the position + list... we precompute the bounding positions in the lookup. Of + course, the neighbors can change (if a position is declined), but + this is an initial mapping */ + + for(i=0;ipartitions;i++)n+=info->class_dim[info->partitionclass[i]]; + n+=2; + look->posts=n; + + /* also store a sorted position index */ + for(i=0;ipostlist+i; + qsort(sortpointer,n,sizeof(*sortpointer),icomp); + + /* points from sort order back to range number */ + for(i=0;iforward_index[i]=sortpointer[i]-info->postlist; + /* points from range order to sorted position */ + for(i=0;ireverse_index[look->forward_index[i]]=i; + /* we actually need the post values too */ + for(i=0;isorted_index[i]=info->postlist[look->forward_index[i]]; + + /* quantize values to multiplier spec */ + switch(info->mult){ + case 1: /* 1024 -> 256 */ + look->quant_q=256; + break; + case 2: /* 1024 -> 128 */ + look->quant_q=128; + break; + case 3: /* 1024 -> 86 */ + look->quant_q=86; + break; + case 4: /* 1024 -> 64 */ + look->quant_q=64; + break; + } + + /* discover our neighbors for decode where we don't use fit flags + (that would push the neighbors outward) */ + for(i=0;in; + int currentx=info->postlist[i+2]; + for(j=0;jpostlist[j]; + if(x>lx && xcurrentx){ + hi=j; + hx=x; + } + } + look->loneighbor[i]=lo; + look->hineighbor[i]=hi; + } + + return(look); +} + +static int render_point(int x0,int x1,int y0,int y1,int x){ + y0&=0x7fff; /* mask off flag */ + y1&=0x7fff; + + { + int dy=y1-y0; + int adx=x1-x0; + int ady=abs(dy); + int err=ady*(x-x0); + + int off=err/adx; + if(dy<0)return(y0-off); + return(y0+off); + } +} + +static int vorbis_dBquant(const float *x){ + int i= *x*7.3142857f+1023.5f; + if(i>1023)return(1023); + if(i<0)return(0); + return i; +} + +static float FLOOR1_fromdB_LOOKUP[256]={ + 1.0649863e-07F, 1.1341951e-07F, 1.2079015e-07F, 1.2863978e-07F, + 1.3699951e-07F, 1.4590251e-07F, 1.5538408e-07F, 1.6548181e-07F, + 1.7623575e-07F, 1.8768855e-07F, 1.9988561e-07F, 2.128753e-07F, + 2.2670913e-07F, 2.4144197e-07F, 2.5713223e-07F, 2.7384213e-07F, + 2.9163793e-07F, 3.1059021e-07F, 3.3077411e-07F, 3.5226968e-07F, + 3.7516214e-07F, 3.9954229e-07F, 4.2550680e-07F, 4.5315863e-07F, + 4.8260743e-07F, 5.1396998e-07F, 5.4737065e-07F, 5.8294187e-07F, + 6.2082472e-07F, 6.6116941e-07F, 7.0413592e-07F, 7.4989464e-07F, + 7.9862701e-07F, 8.5052630e-07F, 9.0579828e-07F, 9.6466216e-07F, + 1.0273513e-06F, 1.0941144e-06F, 1.1652161e-06F, 1.2409384e-06F, + 1.3215816e-06F, 1.4074654e-06F, 1.4989305e-06F, 1.5963394e-06F, + 1.7000785e-06F, 1.8105592e-06F, 1.9282195e-06F, 2.0535261e-06F, + 2.1869758e-06F, 2.3290978e-06F, 2.4804557e-06F, 2.6416497e-06F, + 2.8133190e-06F, 2.9961443e-06F, 3.1908506e-06F, 3.3982101e-06F, + 3.6190449e-06F, 3.8542308e-06F, 4.1047004e-06F, 4.3714470e-06F, + 4.6555282e-06F, 4.9580707e-06F, 5.2802740e-06F, 5.6234160e-06F, + 5.9888572e-06F, 6.3780469e-06F, 6.7925283e-06F, 7.2339451e-06F, + 7.7040476e-06F, 8.2047000e-06F, 8.7378876e-06F, 9.3057248e-06F, + 9.9104632e-06F, 1.0554501e-05F, 1.1240392e-05F, 1.1970856e-05F, + 1.2748789e-05F, 1.3577278e-05F, 1.4459606e-05F, 1.5399272e-05F, + 1.6400004e-05F, 1.7465768e-05F, 1.8600792e-05F, 1.9809576e-05F, + 2.1096914e-05F, 2.2467911e-05F, 2.3928002e-05F, 2.5482978e-05F, + 2.7139006e-05F, 2.8902651e-05F, 3.0780908e-05F, 3.2781225e-05F, + 3.4911534e-05F, 3.7180282e-05F, 3.9596466e-05F, 4.2169667e-05F, + 4.4910090e-05F, 4.7828601e-05F, 5.0936773e-05F, 5.4246931e-05F, + 5.7772202e-05F, 6.1526565e-05F, 6.5524908e-05F, 6.9783085e-05F, + 7.4317983e-05F, 7.9147585e-05F, 8.4291040e-05F, 8.9768747e-05F, + 9.5602426e-05F, 0.00010181521F, 0.00010843174F, 0.00011547824F, + 0.00012298267F, 0.00013097477F, 0.00013948625F, 0.00014855085F, + 0.00015820453F, 0.00016848555F, 0.00017943469F, 0.00019109536F, + 0.00020351382F, 0.00021673929F, 0.00023082423F, 0.00024582449F, + 0.00026179955F, 0.00027881276F, 0.00029693158F, 0.00031622787F, + 0.00033677814F, 0.00035866388F, 0.00038197188F, 0.00040679456F, + 0.00043323036F, 0.00046138411F, 0.00049136745F, 0.00052329927F, + 0.00055730621F, 0.00059352311F, 0.00063209358F, 0.00067317058F, + 0.00071691700F, 0.00076350630F, 0.00081312324F, 0.00086596457F, + 0.00092223983F, 0.00098217216F, 0.0010459992F, 0.0011139742F, + 0.0011863665F, 0.0012634633F, 0.0013455702F, 0.0014330129F, + 0.0015261382F, 0.0016253153F, 0.0017309374F, 0.0018434235F, + 0.0019632195F, 0.0020908006F, 0.0022266726F, 0.0023713743F, + 0.0025254795F, 0.0026895994F, 0.0028643847F, 0.0030505286F, + 0.0032487691F, 0.0034598925F, 0.0036847358F, 0.0039241906F, + 0.0041792066F, 0.0044507950F, 0.0047400328F, 0.0050480668F, + 0.0053761186F, 0.0057254891F, 0.0060975636F, 0.0064938176F, + 0.0069158225F, 0.0073652516F, 0.0078438871F, 0.0083536271F, + 0.0088964928F, 0.009474637F, 0.010090352F, 0.010746080F, + 0.011444421F, 0.012188144F, 0.012980198F, 0.013823725F, + 0.014722068F, 0.015678791F, 0.016697687F, 0.017782797F, + 0.018938423F, 0.020169149F, 0.021479854F, 0.022875735F, + 0.024362330F, 0.025945531F, 0.027631618F, 0.029427276F, + 0.031339626F, 0.033376252F, 0.035545228F, 0.037855157F, + 0.040315199F, 0.042935108F, 0.045725273F, 0.048696758F, + 0.051861348F, 0.055231591F, 0.058820850F, 0.062643361F, + 0.066714279F, 0.071049749F, 0.075666962F, 0.080584227F, + 0.085821044F, 0.091398179F, 0.097337747F, 0.10366330F, + 0.11039993F, 0.11757434F, 0.12521498F, 0.13335215F, + 0.14201813F, 0.15124727F, 0.16107617F, 0.17154380F, + 0.18269168F, 0.19456402F, 0.20720788F, 0.22067342F, + 0.23501402F, 0.25028656F, 0.26655159F, 0.28387361F, + 0.30232132F, 0.32196786F, 0.34289114F, 0.36517414F, + 0.38890521F, 0.41417847F, 0.44109412F, 0.46975890F, + 0.50028648F, 0.53279791F, 0.56742212F, 0.60429640F, + 0.64356699F, 0.68538959F, 0.72993007F, 0.77736504F, + 0.82788260F, 0.88168307F, 0.9389798F, 1.F, +}; + +static void render_line(int x0,int x1,int y0,int y1,float *d){ + int dy=y1-y0; + int adx=x1-x0; + int ady=abs(dy); + int base=dy/adx; + int sy=(dy<0?base-1:base+1); + int x=x0; + int y=y0; + int err=0; + + ady-=abs(base*adx); + + d[x]*=FLOOR1_fromdB_LOOKUP[y]; + while(++x=adx){ + err-=adx; + y+=sy; + }else{ + y+=base; + } + d[x]*=FLOOR1_fromdB_LOOKUP[y]; + } +} + +static void render_line0(int x0,int x1,int y0,int y1,int *d){ + int dy=y1-y0; + int adx=x1-x0; + int ady=abs(dy); + int base=dy/adx; + int sy=(dy<0?base-1:base+1); + int x=x0; + int y=y0; + int err=0; + + ady-=abs(base*adx); + + d[x]=y; + while(++x=adx){ + err-=adx; + y+=sy; + }else{ + y+=base; + } + d[x]=y; + } +} + +/* the floor has already been filtered to only include relevant sections */ +static int accumulate_fit(const float *flr,const float *mdct, + int x0, int x1,lsfit_acc *a, + int n,vorbis_info_floor1 *info){ + long i; + int quantized=vorbis_dBquant(flr+x0); + + long xa=0,ya=0,x2a=0,y2a=0,xya=0,na=0, xb=0,yb=0,x2b=0,y2b=0,xyb=0,nb=0; + + memset(a,0,sizeof(*a)); + a->x0=x0; + a->x1=x1; + if(x1>=n)x1=n-1; + + for(i=x0;i<=x1;i++){ + int quantized=vorbis_dBquant(flr+i); + if(quantized){ + if(mdct[i]+info->twofitatten>=flr[i]){ + xa += i; + ya += quantized; + x2a += i*i; + y2a += quantized*quantized; + xya += i*quantized; + na++; + }else{ + xb += i; + yb += quantized; + x2b += i*i; + y2b += quantized*quantized; + xyb += i*quantized; + nb++; + } + } + } + + xb+=xa; + yb+=ya; + x2b+=x2a; + y2b+=y2a; + xyb+=xya; + nb+=na; + + /* weight toward the actually used frequencies if we meet the threshhold */ + { + int weight=nb*info->twofitweight/(na+1); + + a->xa=xa*weight+xb; + a->ya=ya*weight+yb; + a->x2a=x2a*weight+x2b; + a->y2a=y2a*weight+y2b; + a->xya=xya*weight+xyb; + a->an=na*weight+nb; + a->n=nb; + } + + return(na); +} + +static void fit_line(lsfit_acc *a,int fits,int *y0,int *y1){ + long x=0,y=0,x2=0,y2=0,xy=0,n=0,an=0,i; + long x0=a[0].x0; + long x1=a[fits-1].x1; + + for(i=0;i=0){ + x+= x0; + y+= *y0; + x2+= x0 * x0; + y2+= *y0 * *y0; + xy+= *y0 * x0; + n++; + an++; + } + + if(*y1>=0){ + x+= x1; + y+= *y1; + x2+= x1 * x1; + y2+= *y1 * *y1; + xy+= *y1 * x1; + n++; + an++; + } + + { + /* need 64 bit multiplies, which C doesn't give portably as int */ + double fx=x; + double fy=y; + double fx2=x2; + double fxy=xy; + double denom=1./(an*fx2-fx*fx); + double a=(fy*fx2-fxy*fx)*denom; + double b=(an*fxy-fx*fy)*denom; + *y0=rint(a+b*x0); + *y1=rint(a+b*x1); + + /* limit to our range! */ + if(*y0>1023)*y0=1023; + if(*y1>1023)*y1=1023; + if(*y0<0)*y0=0; + if(*y1<0)*y1=0; + + } +} + +/*static void fit_line_point(lsfit_acc *a,int fits,int *y0,int *y1){ + long y=0; + int i; + + for(i=0;itwofitatten>=mask[x]){ + if(y+info->maxovermaxunder>val)return(1); + } + + while(++x=adx){ + err-=adx; + y+=sy; + }else{ + y+=base; + } + + val=vorbis_dBquant(mask+x); + mse+=((y-val)*(y-val)); + n++; + if(mdct[x]+info->twofitatten>=mask[x]){ + if(val){ + if(y+info->maxovermaxunder>val)return(1); + } + } + } + + if(info->maxover*info->maxover/n>info->maxerr)return(0); + if(info->maxunder*info->maxunder/n>info->maxerr)return(0); + if(mse/n>info->maxerr)return(1); + return(0); +} + +static int post_Y(int *A,int *B,int pos){ + if(A[pos]<0) + return B[pos]; + if(B[pos]<0) + return A[pos]; + + return (A[pos]+B[pos])>>1; +} + +static int seq=0; + +int *floor1_fit(vorbis_block *vb,vorbis_look_floor1 *look, + const float *logmdct, /* in */ + const float *logmask){ + long i,j; + vorbis_info_floor1 *info=look->vi; + long n=look->n; + long posts=look->posts; + long nonzero=0; + lsfit_acc fits[VIF_POSIT+1]; + int fit_valueA[VIF_POSIT+2]; /* index by range list position */ + int fit_valueB[VIF_POSIT+2]; /* index by range list position */ + + int loneighbor[VIF_POSIT+2]; /* sorted index of range list position (+2) */ + int hineighbor[VIF_POSIT+2]; + int *output=NULL; + int memo[VIF_POSIT+2]; + + for(i=0;isorted_index[i], + look->sorted_index[i+1],fits+i, + n,info); + } + + if(nonzero){ + /* start by fitting the implicit base case.... */ + int y0=-200; + int y1=-200; + fit_line(fits,posts-1,&y0,&y1); + + fit_valueA[0]=y0; + fit_valueB[0]=y0; + fit_valueB[1]=y1; + fit_valueA[1]=y1; + + /* Non degenerate case */ + /* start progressive splitting. This is a greedy, non-optimal + algorithm, but simple and close enough to the best + answer. */ + for(i=2;ireverse_index[i]; + int ln=loneighbor[sortpos]; + int hn=hineighbor[sortpos]; + + /* eliminate repeat searches of a particular range with a memo */ + if(memo[ln]!=hn){ + /* haven't performed this error search yet */ + int lsortpos=look->reverse_index[ln]; + int hsortpos=look->reverse_index[hn]; + memo[ln]=hn; + + { + /* A note: we want to bound/minimize *local*, not global, error */ + int lx=info->postlist[ln]; + int hx=info->postlist[hn]; + int ly=post_Y(fit_valueA,fit_valueB,ln); + int hy=post_Y(fit_valueA,fit_valueB,hn); + + if(inspect_error(lx,hx,ly,hy,logmask,logmdct,info)){ + /* outside error bounds/begin search area. Split it. */ + int ly0=-200; + int ly1=-200; + int hy0=-200; + int hy1=-200; + fit_line(fits+lsortpos,sortpos-lsortpos,&ly0,&ly1); + fit_line(fits+sortpos,hsortpos-sortpos,&hy0,&hy1); + + /* store new edge values */ + fit_valueB[ln]=ly0; + if(ln==0)fit_valueA[ln]=ly0; + fit_valueA[i]=ly1; + fit_valueB[i]=hy0; + fit_valueA[hn]=hy1; + if(hn==1)fit_valueB[hn]=hy1; + + if(ly1>=0 || hy0>=0){ + /* store new neighbor values */ + for(j=sortpos-1;j>=0;j--) + if(hineighbor[j]==hn) + hineighbor[j]=i; + else + break; + for(j=sortpos+1;jloneighbor[i-2]; + int hn=look->hineighbor[i-2]; + int x0=info->postlist[ln]; + int x1=info->postlist[hn]; + int y0=output[ln]; + int y1=output[hn]; + + int predicted=render_point(x0,x1,y0,y1,info->postlist[i]); + int vx=post_Y(fit_valueA,fit_valueB,i); + + if(vx>=0 && predicted!=vx){ + output[i]=vx; + }else{ + output[i]= predicted|0x8000; + } + } + } + + return(output); + +} + +int *floor1_interpolate_fit(vorbis_block *vb,vorbis_look_floor1 *look, + int *A,int *B, + int del){ + + long i; + long posts=look->posts; + int *output=NULL; + + if(A && B){ + output=_vorbis_block_alloc(vb,sizeof(*output)*posts); + + for(i=0;i>16; + if(A[i]&0x8000 && B[i]&0x8000)output[i]|=0x8000; + } + } + + return(output); +} + + +int floor1_encode(vorbis_block *vb,vorbis_look_floor1 *look, + int *post,int *ilogmask){ + + long i,j; + vorbis_info_floor1 *info=look->vi; + long n=look->n; + long posts=look->posts; + codec_setup_info *ci=vb->vd->vi->codec_setup; + int out[VIF_POSIT+2]; + static_codebook **sbooks=ci->book_param; + codebook *books=ci->fullbooks; + static long seq=0; + + /* quantize values to multiplier spec */ + if(post){ + for(i=0;imult){ + case 1: /* 1024 -> 256 */ + val>>=2; + break; + case 2: /* 1024 -> 128 */ + val>>=3; + break; + case 3: /* 1024 -> 86 */ + val/=12; + break; + case 4: /* 1024 -> 64 */ + val>>=4; + break; + } + post[i]=val | (post[i]&0x8000); + } + + out[0]=post[0]; + out[1]=post[1]; + + /* find prediction values for each post and subtract them */ + for(i=2;iloneighbor[i-2]; + int hn=look->hineighbor[i-2]; + int x0=info->postlist[ln]; + int x1=info->postlist[hn]; + int y0=post[ln]; + int y1=post[hn]; + + int predicted=render_point(x0,x1,y0,y1,info->postlist[i]); + + if((post[i]&0x8000) || (predicted==post[i])){ + post[i]=predicted|0x8000; /* in case there was roundoff jitter + in interpolation */ + out[i]=0; + }else{ + int headroom=(look->quant_q-predictedquant_q-predicted:predicted); + + int val=post[i]-predicted; + + /* at this point the 'deviation' value is in the range +/- max + range, but the real, unique range can always be mapped to + only [0-maxrange). So we want to wrap the deviation into + this limited range, but do it in the way that least screws + an essentially gaussian probability distribution. */ + + if(val<0) + if(val<-headroom) + val=headroom-val-1; + else + val=-1-(val<<1); + else + if(val>=headroom) + val= val+headroom; + else + val<<=1; + + out[i]=val; + post[ln]&=0x7fff; + post[hn]&=0x7fff; + } + } + + /* we have everything we need. pack it out */ + /* mark nontrivial floor */ + oggpack_write(&vb->opb,1,1); + + /* beginning/end post */ + look->frames++; + look->postbits+=ilog(look->quant_q-1)*2; + oggpack_write(&vb->opb,out[0],ilog(look->quant_q-1)); + oggpack_write(&vb->opb,out[1],ilog(look->quant_q-1)); + + + /* partition by partition */ + for(i=0,j=2;ipartitions;i++){ + int class=info->partitionclass[i]; + int cdim=info->class_dim[class]; + int csubbits=info->class_subs[class]; + int csub=1<class_subbook[class][k]; + if(booknum<0){ + maxval[k]=1; + }else{ + maxval[k]=sbooks[info->class_subbook[class][k]]->entries; + } + } + for(k=0;kphrasebits+= + vorbis_book_encode(books+info->class_book[class],cval,&vb->opb); + +#ifdef TRAIN_FLOOR1 + { + FILE *of; + char buffer[80]; + sprintf(buffer,"line_%dx%ld_class%d.vqd", + vb->pcmend/2,posts-2,class); + of=fopen(buffer,"a"); + fprintf(of,"%d\n",cval); + fclose(of); + } +#endif + } + + /* write post values */ + for(k=0;kclass_subbook[class][bookas[k]]; + if(book>=0){ + /* hack to allow training with 'bad' books */ + if(out[j+k]<(books+book)->entries) + look->postbits+=vorbis_book_encode(books+book, + out[j+k],&vb->opb); + /*else + fprintf(stderr,"+!");*/ + +#ifdef TRAIN_FLOOR1 + { + FILE *of; + char buffer[80]; + sprintf(buffer,"line_%dx%ld_%dsub%d.vqd", + vb->pcmend/2,posts-2,class,bookas[k]); + of=fopen(buffer,"a"); + fprintf(of,"%d\n",out[j+k]); + fclose(of); + } +#endif + } + } + j+=cdim; + } + + { + /* generate quantized floor equivalent to what we'd unpack in decode */ + /* render the lines */ + int hx=0; + int lx=0; + int ly=post[0]*info->mult; + for(j=1;jposts;j++){ + int current=look->forward_index[j]; + int hy=post[current]&0x7fff; + if(hy==post[current]){ + + hy*=info->mult; + hx=info->postlist[current]; + + render_line0(lx,hx,ly,hy,ilogmask); + + lx=hx; + ly=hy; + } + } + for(j=hx;jpcmend/2;j++)ilogmask[j]=ly; /* be certain */ + seq++; + return(1); + } + }else{ + oggpack_write(&vb->opb,0,1); + memset(ilogmask,0,vb->pcmend/2*sizeof(*ilogmask)); + seq++; + return(0); + } +} + +static void *floor1_inverse1(vorbis_block *vb,vorbis_look_floor *in){ + vorbis_look_floor1 *look=(vorbis_look_floor1 *)in; + vorbis_info_floor1 *info=look->vi; + codec_setup_info *ci=vb->vd->vi->codec_setup; + + int i,j,k; + codebook *books=ci->fullbooks; + + /* unpack wrapped/predicted values from stream */ + if(oggpack_read(&vb->opb,1)==1){ + int *fit_value=_vorbis_block_alloc(vb,(look->posts)*sizeof(*fit_value)); + + fit_value[0]=oggpack_read(&vb->opb,ilog(look->quant_q-1)); + fit_value[1]=oggpack_read(&vb->opb,ilog(look->quant_q-1)); + + /* partition by partition */ + for(i=0,j=2;ipartitions;i++){ + int class=info->partitionclass[i]; + int cdim=info->class_dim[class]; + int csubbits=info->class_subs[class]; + int csub=1<class_book[class],&vb->opb); + + if(cval==-1)goto eop; + } + + for(k=0;kclass_subbook[class][cval&(csub-1)]; + cval>>=csubbits; + if(book>=0){ + if((fit_value[j+k]=vorbis_book_decode(books+book,&vb->opb))==-1) + goto eop; + }else{ + fit_value[j+k]=0; + } + } + j+=cdim; + } + + /* unwrap positive values and reconsitute via linear interpolation */ + for(i=2;iposts;i++){ + int predicted=render_point(info->postlist[look->loneighbor[i-2]], + info->postlist[look->hineighbor[i-2]], + fit_value[look->loneighbor[i-2]], + fit_value[look->hineighbor[i-2]], + info->postlist[i]); + int hiroom=look->quant_q-predicted; + int loroom=predicted; + int room=(hiroom=room){ + if(hiroom>loroom){ + val = val-loroom; + }else{ + val = -1-(val-hiroom); + } + }else{ + if(val&1){ + val= -((val+1)>>1); + }else{ + val>>=1; + } + } + + fit_value[i]=val+predicted; + fit_value[look->loneighbor[i-2]]&=0x7fff; + fit_value[look->hineighbor[i-2]]&=0x7fff; + + }else{ + fit_value[i]=predicted|0x8000; + } + + } + + return(fit_value); + } + eop: + return(NULL); +} + +static int floor1_inverse2(vorbis_block *vb,vorbis_look_floor *in,void *memo, + float *out){ + vorbis_look_floor1 *look=(vorbis_look_floor1 *)in; + vorbis_info_floor1 *info=look->vi; + + codec_setup_info *ci=vb->vd->vi->codec_setup; + int n=ci->blocksizes[vb->W]/2; + int j; + + if(memo){ + /* render the lines */ + int *fit_value=(int *)memo; + int hx=0; + int lx=0; + int ly=fit_value[0]*info->mult; + for(j=1;jposts;j++){ + int current=look->forward_index[j]; + int hy=fit_value[current]&0x7fff; + if(hy==fit_value[current]){ + + hy*=info->mult; + hx=info->postlist[current]; + + render_line(lx,hx,ly,hy,out); + + lx=hx; + ly=hy; + } + } + for(j=hx;j header packets + last mod: $Id: info.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +/* general handling of the header and the vorbis_info structure (and + substructures) */ + +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "codebook.h" +#include "registry.h" +#include "window.h" +#include "psy.h" +#include "misc.h" +#include "os.h" + +/* helpers */ +static int ilog2(unsigned int v){ + int ret=0; + if(v)--v; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +static void _v_writestring(oggpack_buffer *o,char *s, int bytes){ + + while(bytes--){ + oggpack_write(o,*s++,8); + } +} + +static void _v_readstring(oggpack_buffer *o,char *buf,int bytes){ + while(bytes--){ + *buf++=oggpack_read(o,8); + } +} + +void vorbis_comment_init(vorbis_comment *vc){ + memset(vc,0,sizeof(*vc)); +} + +void vorbis_comment_add(vorbis_comment *vc,char *comment){ + vc->user_comments=_ogg_realloc(vc->user_comments, + (vc->comments+2)*sizeof(*vc->user_comments)); + vc->comment_lengths=_ogg_realloc(vc->comment_lengths, + (vc->comments+2)*sizeof(*vc->comment_lengths)); + vc->comment_lengths[vc->comments]=strlen(comment); + vc->user_comments[vc->comments]=_ogg_malloc(vc->comment_lengths[vc->comments]+1); + strcpy(vc->user_comments[vc->comments], comment); + vc->comments++; + vc->user_comments[vc->comments]=NULL; +} + +void vorbis_comment_add_tag(vorbis_comment *vc, char *tag, char *contents){ + char *comment=alloca(strlen(tag)+strlen(contents)+2); /* +2 for = and \0 */ + strcpy(comment, tag); + strcat(comment, "="); + strcat(comment, contents); + vorbis_comment_add(vc, comment); +} + +/* This is more or less the same as strncasecmp - but that doesn't exist + * everywhere, and this is a fairly trivial function, so we include it */ +static int tagcompare(const char *s1, const char *s2, int n){ + int c=0; + while(c < n){ + if(toupper(s1[c]) != toupper(s2[c])) + return !0; + c++; + } + return 0; +} + +char *vorbis_comment_query(vorbis_comment *vc, char *tag, int count){ + long i; + int found = 0; + int taglen = strlen(tag)+1; /* +1 for the = we append */ + char *fulltag = alloca(taglen+ 1); + + strcpy(fulltag, tag); + strcat(fulltag, "="); + + for(i=0;icomments;i++){ + if(!tagcompare(vc->user_comments[i], fulltag, taglen)){ + if(count == found) + /* We return a pointer to the data, not a copy */ + return vc->user_comments[i] + taglen; + else + found++; + } + } + return NULL; /* didn't find anything */ +} + +int vorbis_comment_query_count(vorbis_comment *vc, char *tag){ + int i,count=0; + int taglen = strlen(tag)+1; /* +1 for the = we append */ + char *fulltag = alloca(taglen+1); + strcpy(fulltag,tag); + strcat(fulltag, "="); + + for(i=0;icomments;i++){ + if(!tagcompare(vc->user_comments[i], fulltag, taglen)) + count++; + } + + return count; +} + +void vorbis_comment_clear(vorbis_comment *vc){ + if(vc){ + long i; + for(i=0;icomments;i++) + if(vc->user_comments[i])_ogg_free(vc->user_comments[i]); + if(vc->user_comments)_ogg_free(vc->user_comments); + if(vc->comment_lengths)_ogg_free(vc->comment_lengths); + if(vc->vendor)_ogg_free(vc->vendor); + } + memset(vc,0,sizeof(*vc)); +} + +/* blocksize 0 is guaranteed to be short, 1 is guarantted to be long. + They may be equal, but short will never ge greater than long */ +int vorbis_info_blocksize(vorbis_info *vi,int zo){ + codec_setup_info *ci = vi->codec_setup; + return ci ? ci->blocksizes[zo] : -1; +} + +/* used by synthesis, which has a full, alloced vi */ +void vorbis_info_init(vorbis_info *vi){ + memset(vi,0,sizeof(*vi)); + vi->codec_setup=_ogg_calloc(1,sizeof(codec_setup_info)); +} + +void vorbis_info_clear(vorbis_info *vi){ + codec_setup_info *ci=vi->codec_setup; + int i; + + if(ci){ + + for(i=0;imodes;i++) + if(ci->mode_param[i])_ogg_free(ci->mode_param[i]); + + for(i=0;imaps;i++) /* unpack does the range checking */ + _mapping_P[ci->map_type[i]]->free_info(ci->map_param[i]); + + for(i=0;ifloors;i++) /* unpack does the range checking */ + _floor_P[ci->floor_type[i]]->free_info(ci->floor_param[i]); + + for(i=0;iresidues;i++) /* unpack does the range checking */ + _residue_P[ci->residue_type[i]]->free_info(ci->residue_param[i]); + + for(i=0;ibooks;i++){ + if(ci->book_param[i]){ + /* knows if the book was not alloced */ + vorbis_staticbook_destroy(ci->book_param[i]); + } + if(ci->fullbooks) + vorbis_book_clear(ci->fullbooks+i); + } + if(ci->fullbooks) + _ogg_free(ci->fullbooks); + + for(i=0;ipsys;i++) + _vi_psy_free(ci->psy_param[i]); + + _ogg_free(ci); + } + + memset(vi,0,sizeof(*vi)); +} + +/* Header packing/unpacking ********************************************/ + +static int _vorbis_unpack_info(vorbis_info *vi,oggpack_buffer *opb){ + codec_setup_info *ci=vi->codec_setup; + if(!ci)return(OV_EFAULT); + + vi->version=oggpack_read(opb,32); + if(vi->version!=0)return(OV_EVERSION); + + vi->channels=oggpack_read(opb,8); + vi->rate=oggpack_read(opb,32); + + vi->bitrate_upper=oggpack_read(opb,32); + vi->bitrate_nominal=oggpack_read(opb,32); + vi->bitrate_lower=oggpack_read(opb,32); + + ci->blocksizes[0]=1<blocksizes[1]=1<rate<1)goto err_out; + if(vi->channels<1)goto err_out; + if(ci->blocksizes[0]<8)goto err_out; + if(ci->blocksizes[1]blocksizes[0])goto err_out; + + if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */ + + return(0); + err_out: + vorbis_info_clear(vi); + return(OV_EBADHEADER); +} + +static int _vorbis_unpack_comment(vorbis_comment *vc,oggpack_buffer *opb){ + int i; + int vendorlen=oggpack_read(opb,32); + if(vendorlen<0)goto err_out; + vc->vendor=_ogg_calloc(vendorlen+1,1); + _v_readstring(opb,vc->vendor,vendorlen); + vc->comments=oggpack_read(opb,32); + if(vc->comments<0)goto err_out; + vc->user_comments=_ogg_calloc(vc->comments+1,sizeof(*vc->user_comments)); + vc->comment_lengths=_ogg_calloc(vc->comments+1, sizeof(*vc->comment_lengths)); + + for(i=0;icomments;i++){ + int len=oggpack_read(opb,32); + if(len<0)goto err_out; + vc->comment_lengths[i]=len; + vc->user_comments[i]=_ogg_calloc(len+1,1); + _v_readstring(opb,vc->user_comments[i],len); + } + if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */ + + return(0); + err_out: + vorbis_comment_clear(vc); + return(OV_EBADHEADER); +} + +/* all of the real encoding details are here. The modes, books, + everything */ +static int _vorbis_unpack_books(vorbis_info *vi,oggpack_buffer *opb){ + codec_setup_info *ci=vi->codec_setup; + int i; + if(!ci)return(OV_EFAULT); + + /* codebooks */ + ci->books=oggpack_read(opb,8)+1; + /*ci->book_param=_ogg_calloc(ci->books,sizeof(*ci->book_param));*/ + for(i=0;ibooks;i++){ + ci->book_param[i]=_ogg_calloc(1,sizeof(*ci->book_param[i])); + if(vorbis_staticbook_unpack(opb,ci->book_param[i]))goto err_out; + } + + /* time backend settings; hooks are unused */ + { + int times=oggpack_read(opb,6)+1; + for(i=0;i=VI_TIMEB)goto err_out; + } + } + + /* floor backend settings */ + ci->floors=oggpack_read(opb,6)+1; + /*ci->floor_type=_ogg_malloc(ci->floors*sizeof(*ci->floor_type));*/ + /*ci->floor_param=_ogg_calloc(ci->floors,sizeof(void *));*/ + for(i=0;ifloors;i++){ + ci->floor_type[i]=oggpack_read(opb,16); + if(ci->floor_type[i]<0 || ci->floor_type[i]>=VI_FLOORB)goto err_out; + ci->floor_param[i]=_floor_P[ci->floor_type[i]]->unpack(vi,opb); + if(!ci->floor_param[i])goto err_out; + } + + /* residue backend settings */ + ci->residues=oggpack_read(opb,6)+1; + /*ci->residue_type=_ogg_malloc(ci->residues*sizeof(*ci->residue_type));*/ + /*ci->residue_param=_ogg_calloc(ci->residues,sizeof(void *));*/ + for(i=0;iresidues;i++){ + ci->residue_type[i]=oggpack_read(opb,16); + if(ci->residue_type[i]<0 || ci->residue_type[i]>=VI_RESB)goto err_out; + ci->residue_param[i]=_residue_P[ci->residue_type[i]]->unpack(vi,opb); + if(!ci->residue_param[i])goto err_out; + } + + /* map backend settings */ + ci->maps=oggpack_read(opb,6)+1; + /*ci->map_type=_ogg_malloc(ci->maps*sizeof(*ci->map_type));*/ + /*ci->map_param=_ogg_calloc(ci->maps,sizeof(void *));*/ + for(i=0;imaps;i++){ + ci->map_type[i]=oggpack_read(opb,16); + if(ci->map_type[i]<0 || ci->map_type[i]>=VI_MAPB)goto err_out; + ci->map_param[i]=_mapping_P[ci->map_type[i]]->unpack(vi,opb); + if(!ci->map_param[i])goto err_out; + } + + /* mode settings */ + ci->modes=oggpack_read(opb,6)+1; + /*vi->mode_param=_ogg_calloc(vi->modes,sizeof(void *));*/ + for(i=0;imodes;i++){ + ci->mode_param[i]=_ogg_calloc(1,sizeof(*ci->mode_param[i])); + ci->mode_param[i]->blockflag=oggpack_read(opb,1); + ci->mode_param[i]->windowtype=oggpack_read(opb,16); + ci->mode_param[i]->transformtype=oggpack_read(opb,16); + ci->mode_param[i]->mapping=oggpack_read(opb,8); + + if(ci->mode_param[i]->windowtype>=VI_WINDOWB)goto err_out; + if(ci->mode_param[i]->transformtype>=VI_WINDOWB)goto err_out; + if(ci->mode_param[i]->mapping>=ci->maps)goto err_out; + } + + if(oggpack_read(opb,1)!=1)goto err_out; /* top level EOP check */ + + return(0); + err_out: + vorbis_info_clear(vi); + return(OV_EBADHEADER); +} + +/* The Vorbis header is in three packets; the initial small packet in + the first page that identifies basic parameters, a second packet + with bitstream comments and a third packet that holds the + codebook. */ + +int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,ogg_packet *op){ + oggpack_buffer opb; + + if(op){ + oggpack_readinit(&opb,op->packet,op->bytes); + + /* Which of the three types of header is this? */ + /* Also verify header-ness, vorbis */ + { + char buffer[6]; + int packtype=oggpack_read(&opb,8); + memset(buffer,0,6); + _v_readstring(&opb,buffer,6); + if(memcmp(buffer,"vorbis",6)){ + /* not a vorbis header */ + return(OV_ENOTVORBIS); + } + switch(packtype){ + case 0x01: /* least significant *bit* is read first */ + if(!op->b_o_s){ + /* Not the initial packet */ + return(OV_EBADHEADER); + } + if(vi->rate!=0){ + /* previously initialized info header */ + return(OV_EBADHEADER); + } + + return(_vorbis_unpack_info(vi,&opb)); + + case 0x03: /* least significant *bit* is read first */ + if(vi->rate==0){ + /* um... we didn't get the initial header */ + return(OV_EBADHEADER); + } + + return(_vorbis_unpack_comment(vc,&opb)); + + case 0x05: /* least significant *bit* is read first */ + if(vi->rate==0 || vc->vendor==NULL){ + /* um... we didn;t get the initial header or comments yet */ + return(OV_EBADHEADER); + } + + return(_vorbis_unpack_books(vi,&opb)); + + default: + /* Not a valid vorbis header type */ + return(OV_EBADHEADER); + break; + } + } + } + return(OV_EBADHEADER); +} + +/* pack side **********************************************************/ + +static int _vorbis_pack_info(oggpack_buffer *opb,vorbis_info *vi){ + codec_setup_info *ci=vi->codec_setup; + if(!ci)return(OV_EFAULT); + + /* preamble */ + oggpack_write(opb,0x01,8); + _v_writestring(opb,"vorbis", 6); + + /* basic information about the stream */ + oggpack_write(opb,0x00,32); + oggpack_write(opb,vi->channels,8); + oggpack_write(opb,vi->rate,32); + + oggpack_write(opb,vi->bitrate_upper,32); + oggpack_write(opb,vi->bitrate_nominal,32); + oggpack_write(opb,vi->bitrate_lower,32); + + oggpack_write(opb,ilog2(ci->blocksizes[0]),4); + oggpack_write(opb,ilog2(ci->blocksizes[1]),4); + oggpack_write(opb,1,1); + + return(0); +} + +static int _vorbis_pack_comment(oggpack_buffer *opb,vorbis_comment *vc){ + char temp[]="Xiph.Org libVorbis I 20020717"; + int bytes = strlen(temp); + + /* preamble */ + oggpack_write(opb,0x03,8); + _v_writestring(opb,"vorbis", 6); + + /* vendor */ + oggpack_write(opb,bytes,32); + _v_writestring(opb,temp, bytes); + + /* comments */ + + oggpack_write(opb,vc->comments,32); + if(vc->comments){ + int i; + for(i=0;icomments;i++){ + if(vc->user_comments[i]){ + oggpack_write(opb,vc->comment_lengths[i],32); + _v_writestring(opb,vc->user_comments[i], vc->comment_lengths[i]); + }else{ + oggpack_write(opb,0,32); + } + } + } + oggpack_write(opb,1,1); + + return(0); +} + +static int _vorbis_pack_books(oggpack_buffer *opb,vorbis_info *vi){ + codec_setup_info *ci=vi->codec_setup; + int i; + if(!ci)return(OV_EFAULT); + + oggpack_write(opb,0x05,8); + _v_writestring(opb,"vorbis", 6); + + /* books */ + oggpack_write(opb,ci->books-1,8); + for(i=0;ibooks;i++) + if(vorbis_staticbook_pack(ci->book_param[i],opb))goto err_out; + + /* times; hook placeholders */ + oggpack_write(opb,0,6); + oggpack_write(opb,0,16); + + /* floors */ + oggpack_write(opb,ci->floors-1,6); + for(i=0;ifloors;i++){ + oggpack_write(opb,ci->floor_type[i],16); + if(_floor_P[ci->floor_type[i]]->pack) + _floor_P[ci->floor_type[i]]->pack(ci->floor_param[i],opb); + else + goto err_out; + } + + /* residues */ + oggpack_write(opb,ci->residues-1,6); + for(i=0;iresidues;i++){ + oggpack_write(opb,ci->residue_type[i],16); + _residue_P[ci->residue_type[i]]->pack(ci->residue_param[i],opb); + } + + /* maps */ + oggpack_write(opb,ci->maps-1,6); + for(i=0;imaps;i++){ + oggpack_write(opb,ci->map_type[i],16); + _mapping_P[ci->map_type[i]]->pack(vi,ci->map_param[i],opb); + } + + /* modes */ + oggpack_write(opb,ci->modes-1,6); + for(i=0;imodes;i++){ + oggpack_write(opb,ci->mode_param[i]->blockflag,1); + oggpack_write(opb,ci->mode_param[i]->windowtype,16); + oggpack_write(opb,ci->mode_param[i]->transformtype,16); + oggpack_write(opb,ci->mode_param[i]->mapping,8); + } + oggpack_write(opb,1,1); + + return(0); +err_out: + return(-1); +} + +int vorbis_commentheader_out(vorbis_comment *vc, + ogg_packet *op){ + + oggpack_buffer opb; + + oggpack_writeinit(&opb); + if(_vorbis_pack_comment(&opb,vc)) return OV_EIMPL; + + op->packet = _ogg_malloc(oggpack_bytes(&opb)); + memcpy(op->packet, opb.buffer, oggpack_bytes(&opb)); + + op->bytes=oggpack_bytes(&opb); + op->b_o_s=0; + op->e_o_s=0; + op->granulepos=0; + + return 0; +} + +int vorbis_analysis_headerout(vorbis_dsp_state *v, + vorbis_comment *vc, + ogg_packet *op, + ogg_packet *op_comm, + ogg_packet *op_code){ + int ret=OV_EIMPL; + vorbis_info *vi=v->vi; + oggpack_buffer opb; + backend_lookup_state *b=v->backend_state; + + if(!b){ + ret=OV_EFAULT; + goto err_out; + } + + /* first header packet **********************************************/ + + oggpack_writeinit(&opb); + if(_vorbis_pack_info(&opb,vi))goto err_out; + + /* build the packet */ + if(b->header)_ogg_free(b->header); + b->header=_ogg_malloc(oggpack_bytes(&opb)); + memcpy(b->header,opb.buffer,oggpack_bytes(&opb)); + op->packet=b->header; + op->bytes=oggpack_bytes(&opb); + op->b_o_s=1; + op->e_o_s=0; + op->granulepos=0; + + /* second header packet (comments) **********************************/ + + oggpack_reset(&opb); + if(_vorbis_pack_comment(&opb,vc))goto err_out; + + if(b->header1)_ogg_free(b->header1); + b->header1=_ogg_malloc(oggpack_bytes(&opb)); + memcpy(b->header1,opb.buffer,oggpack_bytes(&opb)); + op_comm->packet=b->header1; + op_comm->bytes=oggpack_bytes(&opb); + op_comm->b_o_s=0; + op_comm->e_o_s=0; + op_comm->granulepos=0; + + /* third header packet (modes/codebooks) ****************************/ + + oggpack_reset(&opb); + if(_vorbis_pack_books(&opb,vi))goto err_out; + + if(b->header2)_ogg_free(b->header2); + b->header2=_ogg_malloc(oggpack_bytes(&opb)); + memcpy(b->header2,opb.buffer,oggpack_bytes(&opb)); + op_code->packet=b->header2; + op_code->bytes=oggpack_bytes(&opb); + op_code->b_o_s=0; + op_code->e_o_s=0; + op_code->granulepos=0; + + oggpack_writeclear(&opb); + return(0); + err_out: + oggpack_writeclear(&opb); + memset(op,0,sizeof(*op)); + memset(op_comm,0,sizeof(*op_comm)); + memset(op_code,0,sizeof(*op_code)); + + if(b->header)_ogg_free(b->header); + if(b->header1)_ogg_free(b->header1); + if(b->header2)_ogg_free(b->header2); + b->header=NULL; + b->header1=NULL; + b->header2=NULL; + return(ret); +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/lookup.c b/src/add-ons/media/plugins/vorbis/libvorbis/lookup.c new file mode 100644 index 0000000000..f67671eac3 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/lookup.c @@ -0,0 +1,94 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: lookup based functions + last mod: $Id: lookup.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include "lookup.h" +#include "lookup_data.h" +#include "os.h" +#include "misc.h" + +#ifdef FLOAT_LOOKUP + +/* interpolated lookup based cos function, domain 0 to PI only */ +float vorbis_coslook(float a){ + double d=a*(.31830989*(float)COS_LOOKUP_SZ); + int i=vorbis_ftoi(d-.5); + + return COS_LOOKUP[i]+ (d-i)*(COS_LOOKUP[i+1]-COS_LOOKUP[i]); +} + +/* interpolated 1./sqrt(p) where .5 <= p < 1. */ +float vorbis_invsqlook(float a){ + double d=a*(2.f*(float)INVSQ_LOOKUP_SZ)-(float)INVSQ_LOOKUP_SZ; + int i=vorbis_ftoi(d-.5f); + return INVSQ_LOOKUP[i]+ (d-i)*(INVSQ_LOOKUP[i+1]-INVSQ_LOOKUP[i]); +} + +/* interpolated 1./sqrt(p) where .5 <= p < 1. */ +float vorbis_invsq2explook(int a){ + return INVSQ2EXP_LOOKUP[a-INVSQ2EXP_LOOKUP_MIN]; +} + +#include +/* interpolated lookup based fromdB function, domain -140dB to 0dB only */ +float vorbis_fromdBlook(float a){ + int i=vorbis_ftoi(a*((float)(-(1<=(FROMdB_LOOKUP_SZ<>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]); +} + +#endif + +#ifdef INT_LOOKUP +/* interpolated 1./sqrt(p) where .5 <= a < 1. (.100000... to .111111...) in + 16.16 format + + returns in m.8 format */ +long vorbis_invsqlook_i(long a,long e){ + long i=(a&0x7fff)>>(INVSQ_LOOKUP_I_SHIFT-1); + long d=(a&INVSQ_LOOKUP_I_MASK)<<(16-INVSQ_LOOKUP_I_SHIFT); /* 0.16 */ + long val=INVSQ_LOOKUP_I[i]- /* 1.16 */ + (((INVSQ_LOOKUP_I[i]-INVSQ_LOOKUP_I[i+1])* /* 0.16 */ + d)>>16); /* result 1.16 */ + + e+=32; + if(e&1)val=(val*5792)>>13; /* multiply val by 1/sqrt(2) */ + e=(e>>1)-8; + + return(val>>e); +} + +/* interpolated lookup based fromdB function, domain -140dB to 0dB only */ +/* a is in n.12 format */ +float vorbis_fromdBlook_i(long a){ + int i=(-a)>>(12-FROMdB2_SHIFT); + return (i<0)?1.f: + ((i>=(FROMdB_LOOKUP_SZ<>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]); +} + +/* interpolated lookup based cos function, domain 0 to PI only */ +/* a is in 0.16 format, where 0==0, 2^^16-1==PI, return 0.14 */ +long vorbis_coslook_i(long a){ + int i=a>>COS_LOOKUP_I_SHIFT; + int d=a&COS_LOOKUP_I_MASK; + return COS_LOOKUP_I[i]- ((d*(COS_LOOKUP_I[i]-COS_LOOKUP_I[i+1]))>> + COS_LOOKUP_I_SHIFT); +} + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/lookup.h b/src/add-ons/media/plugins/vorbis/libvorbis/lookup.h new file mode 100644 index 0000000000..a9b5567faf --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/lookup.h @@ -0,0 +1,32 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: lookup based functions + last mod: $Id: lookup.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_LOOKUP_H_ + +#ifdef FLOAT_LOOKUP +extern float vorbis_coslook(float a); +extern float vorbis_invsqlook(float a); +extern float vorbis_invsq2explook(int a); +extern float vorbis_fromdBlook(float a); +#endif +#ifdef INT_LOOKUP +extern long vorbis_invsqlook_i(long a,long e); +extern long vorbis_coslook_i(long a); +extern float vorbis_fromdBlook_i(long a); +#endif + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/lookup_data.h b/src/add-ons/media/plugins/vorbis/libvorbis/lookup_data.h new file mode 100644 index 0000000000..d3b5bec188 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/lookup_data.h @@ -0,0 +1,189 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: lookup data; generated by lookups.pl; edit there + last mod: $Id: lookup_data.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_LOOKUP_DATA_H_ + +#ifdef FLOAT_LOOKUP +#define COS_LOOKUP_SZ 128 +static float COS_LOOKUP[COS_LOOKUP_SZ+1]={ + +1.0000000000000f,+0.9996988186962f,+0.9987954562052f,+0.9972904566787f, + +0.9951847266722f,+0.9924795345987f,+0.9891765099648f,+0.9852776423889f, + +0.9807852804032f,+0.9757021300385f,+0.9700312531945f,+0.9637760657954f, + +0.9569403357322f,+0.9495281805930f,+0.9415440651830f,+0.9329927988347f, + +0.9238795325113f,+0.9142097557035f,+0.9039892931234f,+0.8932243011955f, + +0.8819212643484f,+0.8700869911087f,+0.8577286100003f,+0.8448535652497f, + +0.8314696123025f,+0.8175848131516f,+0.8032075314806f,+0.7883464276266f, + +0.7730104533627f,+0.7572088465065f,+0.7409511253550f,+0.7242470829515f, + +0.7071067811865f,+0.6895405447371f,+0.6715589548470f,+0.6531728429538f, + +0.6343932841636f,+0.6152315905806f,+0.5956993044924f,+0.5758081914178f, + +0.5555702330196f,+0.5349976198871f,+0.5141027441932f,+0.4928981922298f, + +0.4713967368260f,+0.4496113296546f,+0.4275550934303f,+0.4052413140050f, + +0.3826834323651f,+0.3598950365350f,+0.3368898533922f,+0.3136817403989f, + +0.2902846772545f,+0.2667127574749f,+0.2429801799033f,+0.2191012401569f, + +0.1950903220161f,+0.1709618887603f,+0.1467304744554f,+0.1224106751992f, + +0.0980171403296f,+0.0735645635997f,+0.0490676743274f,+0.0245412285229f, + +0.0000000000000f,-0.0245412285229f,-0.0490676743274f,-0.0735645635997f, + -0.0980171403296f,-0.1224106751992f,-0.1467304744554f,-0.1709618887603f, + -0.1950903220161f,-0.2191012401569f,-0.2429801799033f,-0.2667127574749f, + -0.2902846772545f,-0.3136817403989f,-0.3368898533922f,-0.3598950365350f, + -0.3826834323651f,-0.4052413140050f,-0.4275550934303f,-0.4496113296546f, + -0.4713967368260f,-0.4928981922298f,-0.5141027441932f,-0.5349976198871f, + -0.5555702330196f,-0.5758081914178f,-0.5956993044924f,-0.6152315905806f, + -0.6343932841636f,-0.6531728429538f,-0.6715589548470f,-0.6895405447371f, + -0.7071067811865f,-0.7242470829515f,-0.7409511253550f,-0.7572088465065f, + -0.7730104533627f,-0.7883464276266f,-0.8032075314806f,-0.8175848131516f, + -0.8314696123025f,-0.8448535652497f,-0.8577286100003f,-0.8700869911087f, + -0.8819212643484f,-0.8932243011955f,-0.9039892931234f,-0.9142097557035f, + -0.9238795325113f,-0.9329927988347f,-0.9415440651830f,-0.9495281805930f, + -0.9569403357322f,-0.9637760657954f,-0.9700312531945f,-0.9757021300385f, + -0.9807852804032f,-0.9852776423889f,-0.9891765099648f,-0.9924795345987f, + -0.9951847266722f,-0.9972904566787f,-0.9987954562052f,-0.9996988186962f, + -1.0000000000000f, +}; + +#define INVSQ_LOOKUP_SZ 32 +static float INVSQ_LOOKUP[INVSQ_LOOKUP_SZ+1]={ + 1.414213562373f,1.392621247646f,1.371988681140f,1.352246807566f, + 1.333333333333f,1.315191898443f,1.297771369046f,1.281025230441f, + 1.264911064067f,1.249390095109f,1.234426799697f,1.219988562661f, + 1.206045378311f,1.192569588000f,1.179535649239f,1.166919931983f, + 1.154700538379f,1.142857142857f,1.131370849898f,1.120224067222f, + 1.109400392450f,1.098884511590f,1.088662107904f,1.078719779941f, + 1.069044967650f,1.059625885652f,1.050451462878f,1.041511287847f, + 1.032795558989f,1.024295039463f,1.016001016002f,1.007905261358f, + 1.000000000000f, +}; + +#define INVSQ2EXP_LOOKUP_MIN -32 +#define INVSQ2EXP_LOOKUP_MAX 32 +static float INVSQ2EXP_LOOKUP[INVSQ2EXP_LOOKUP_MAX-\ + INVSQ2EXP_LOOKUP_MIN+1]={ + 65536.f, 46340.95001f, 32768.f, 23170.47501f, + 16384.f, 11585.2375f, 8192.f, 5792.618751f, + 4096.f, 2896.309376f, 2048.f, 1448.154688f, + 1024.f, 724.0773439f, 512.f, 362.038672f, + 256.f, 181.019336f, 128.f, 90.50966799f, + 64.f, 45.254834f, 32.f, 22.627417f, + 16.f, 11.3137085f, 8.f, 5.656854249f, + 4.f, 2.828427125f, 2.f, 1.414213562f, + 1.f, 0.7071067812f, 0.5f, 0.3535533906f, + 0.25f, 0.1767766953f, 0.125f, 0.08838834765f, + 0.0625f, 0.04419417382f, 0.03125f, 0.02209708691f, + 0.015625f, 0.01104854346f, 0.0078125f, 0.005524271728f, + 0.00390625f, 0.002762135864f, 0.001953125f, 0.001381067932f, + 0.0009765625f, 0.000690533966f, 0.00048828125f, 0.000345266983f, + 0.000244140625f,0.0001726334915f,0.0001220703125f,8.631674575e-05f, + 6.103515625e-05f,4.315837288e-05f,3.051757812e-05f,2.157918644e-05f, + 1.525878906e-05f, +}; + +#endif + +#define FROMdB_LOOKUP_SZ 35 +#define FROMdB2_LOOKUP_SZ 32 +#define FROMdB_SHIFT 5 +#define FROMdB2_SHIFT 3 +#define FROMdB2_MASK 31 +static float FROMdB_LOOKUP[FROMdB_LOOKUP_SZ]={ + 1.f, 0.6309573445f, 0.3981071706f, 0.2511886432f, + 0.1584893192f, 0.1f, 0.06309573445f, 0.03981071706f, + 0.02511886432f, 0.01584893192f, 0.01f, 0.006309573445f, + 0.003981071706f, 0.002511886432f, 0.001584893192f, 0.001f, + 0.0006309573445f,0.0003981071706f,0.0002511886432f,0.0001584893192f, + 0.0001f,6.309573445e-05f,3.981071706e-05f,2.511886432e-05f, + 1.584893192e-05f, 1e-05f,6.309573445e-06f,3.981071706e-06f, + 2.511886432e-06f,1.584893192e-06f, 1e-06f,6.309573445e-07f, + 3.981071706e-07f,2.511886432e-07f,1.584893192e-07f, +}; + +static float FROMdB2_LOOKUP[FROMdB2_LOOKUP_SZ]={ + 0.9928302478f, 0.9786445908f, 0.9646616199f, 0.9508784391f, + 0.9372921937f, 0.92390007f, 0.9106992942f, 0.8976871324f, + 0.8848608897f, 0.8722179097f, 0.8597555737f, 0.8474713009f, + 0.835362547f, 0.8234268041f, 0.8116616003f, 0.8000644989f, + 0.7886330981f, 0.7773650302f, 0.7662579617f, 0.755309592f, + 0.7445176537f, 0.7338799116f, 0.7233941627f, 0.7130582353f, + 0.7028699885f, 0.6928273125f, 0.6829281272f, 0.6731703824f, + 0.6635520573f, 0.6540711597f, 0.6447257262f, 0.6355138211f, +}; + +#ifdef INT_LOOKUP + +#define INVSQ_LOOKUP_I_SHIFT 10 +#define INVSQ_LOOKUP_I_MASK 1023 +static long INVSQ_LOOKUP_I[64+1]={ + 92682l, 91966l, 91267l, 90583l, + 89915l, 89261l, 88621l, 87995l, + 87381l, 86781l, 86192l, 85616l, + 85051l, 84497l, 83953l, 83420l, + 82897l, 82384l, 81880l, 81385l, + 80899l, 80422l, 79953l, 79492l, + 79039l, 78594l, 78156l, 77726l, + 77302l, 76885l, 76475l, 76072l, + 75674l, 75283l, 74898l, 74519l, + 74146l, 73778l, 73415l, 73058l, + 72706l, 72359l, 72016l, 71679l, + 71347l, 71019l, 70695l, 70376l, + 70061l, 69750l, 69444l, 69141l, + 68842l, 68548l, 68256l, 67969l, + 67685l, 67405l, 67128l, 66855l, + 66585l, 66318l, 66054l, 65794l, + 65536l, +}; + +#define COS_LOOKUP_I_SHIFT 9 +#define COS_LOOKUP_I_MASK 511 +#define COS_LOOKUP_I_SZ 128 +static long COS_LOOKUP_I[COS_LOOKUP_I_SZ+1]={ + 16384l, 16379l, 16364l, 16340l, + 16305l, 16261l, 16207l, 16143l, + 16069l, 15986l, 15893l, 15791l, + 15679l, 15557l, 15426l, 15286l, + 15137l, 14978l, 14811l, 14635l, + 14449l, 14256l, 14053l, 13842l, + 13623l, 13395l, 13160l, 12916l, + 12665l, 12406l, 12140l, 11866l, + 11585l, 11297l, 11003l, 10702l, + 10394l, 10080l, 9760l, 9434l, + 9102l, 8765l, 8423l, 8076l, + 7723l, 7366l, 7005l, 6639l, + 6270l, 5897l, 5520l, 5139l, + 4756l, 4370l, 3981l, 3590l, + 3196l, 2801l, 2404l, 2006l, + 1606l, 1205l, 804l, 402l, + 0l, -401l, -803l, -1204l, + -1605l, -2005l, -2403l, -2800l, + -3195l, -3589l, -3980l, -4369l, + -4755l, -5138l, -5519l, -5896l, + -6269l, -6638l, -7004l, -7365l, + -7722l, -8075l, -8422l, -8764l, + -9101l, -9433l, -9759l, -10079l, + -10393l, -10701l, -11002l, -11296l, + -11584l, -11865l, -12139l, -12405l, + -12664l, -12915l, -13159l, -13394l, + -13622l, -13841l, -14052l, -14255l, + -14448l, -14634l, -14810l, -14977l, + -15136l, -15285l, -15425l, -15556l, + -15678l, -15790l, -15892l, -15985l, + -16068l, -16142l, -16206l, -16260l, + -16304l, -16339l, -16363l, -16378l, + -16383l, +}; + +#endif + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/lpc.c b/src/add-ons/media/plugins/vorbis/libvorbis/lpc.c new file mode 100644 index 0000000000..750efa587c --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/lpc.c @@ -0,0 +1,200 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: LPC low level routines + last mod: $Id: lpc.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +/* Some of these routines (autocorrelator, LPC coefficient estimator) + are derived from code written by Jutta Degener and Carsten Bormann; + thus we include their copyright below. The entirety of this file + is freely redistributable on the condition that both of these + copyright notices are preserved without modification. */ + +/* Preserved Copyright: *********************************************/ + +/* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann, +Technische Universita"t Berlin + +Any use of this software is permitted provided that this notice is not +removed and that neither the authors nor the Technische Universita"t +Berlin are deemed to have made any representations as to the +suitability of this software for any purpose nor are held responsible +for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR +THIS SOFTWARE. + +As a matter of courtesy, the authors request to be informed about uses +this software has found, about bugs in this software, and about any +improvements that may be of general interest. + +Berlin, 28.11.1994 +Jutta Degener +Carsten Bormann + +*********************************************************************/ + +#include +#include +#include +#include "os.h" +#include "smallft.h" +#include "lpc.h" +#include "scales.h" +#include "misc.h" + +/* Autocorrelation LPC coeff generation algorithm invented by + N. Levinson in 1947, modified by J. Durbin in 1959. */ + +/* Input : n elements of time doamin data + Output: m lpc coefficients, excitation energy */ + +float vorbis_lpc_from_data(float *data,float *lpc,int n,int m){ + float *aut=alloca(sizeof(*aut)*(m+1)); + float error; + int i,j; + + /* autocorrelation, p+1 lag coefficients */ + + j=m+1; + while(j--){ + double d=0; /* double needed for accumulator depth */ + for(i=j;iln; + int m=l->m; + float *work=alloca(sizeof(*work)*(n+n)); + float fscale=.5f/n; + int i,j; + + /* input is a real curve. make it complex-real */ + /* This mixes phase, but the LPC generation doesn't care. */ + for(i=0;ifft,work); + + /* The autocorrelation will not be circular. Shift, else we lose + most of the power in the edges. */ + + for(i=0,j=n/2;iln=mapped; + l->m=m; + + /* we cheat decoding the LPC spectrum via FFTs */ + drft_init(&l->fft,mapped*2); + +} + +void lpc_clear(lpc_lookup *l){ + if(l){ + drft_clear(&l->fft); + } +} + +void vorbis_lpc_predict(float *coeff,float *prime,int m, + float *data,long n){ + + /* in: coeff[0...m-1] LPC coefficients + prime[0...m-1] initial values (allocated size of n+m-1) + out: data[0...n-1] data samples */ + + long i,j,o,p; + float y; + float *work=alloca(sizeof(*work)*(m+n)); + + if(!prime) + for(i=0;i, available at: + + http://www2.xtdl.com/~rothwlr/lsfpaper/lsfpage.html + + ********************************************************************/ + +/* Note that the lpc-lsp conversion finds the roots of polynomial with + an iterative root polisher (CACM algorithm 283). It *is* possible + to confuse this algorithm into not converging; that should only + happen with absurdly closely spaced roots (very sharp peaks in the + LPC f response) which in turn should be impossible in our use of + the code. If this *does* happen anyway, it's a bug in the floor + finder; find the cause of the confusion (probably a single bin + spike or accidental near-float-limit resolution problems) and + correct it. */ + +#include +#include +#include +#include "lsp.h" +#include "os.h" +#include "misc.h" +#include "lookup.h" +#include "scales.h" + +/* three possible LSP to f curve functions; the exact computation + (float), a lookup based float implementation, and an integer + implementation. The float lookup is likely the optimal choice on + any machine with an FPU. The integer implementation is *not* fixed + point (due to the need for a large dynamic range and thus a + seperately tracked exponent) and thus much more complex than the + relatively simple float implementations. It's mostly for future + work on a fully fixed point implementation for processors like the + ARM family. */ + +/* undefine both for the 'old' but more precise implementation */ +#define FLOAT_LOOKUP +#undef INT_LOOKUP + +#ifdef FLOAT_LOOKUP +#include "lookup.c" /* catch this in the build system; we #include for + compilers (like gcc) that can't inline across + modules */ + +/* side effect: changes *lsp to cosines of lsp */ +void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m, + float amp,float ampoffset){ + int i; + float wdel=M_PI/ln; + vorbis_fpu_control fpu; + + vorbis_fpu_setround(&fpu); + for(i=0;i>1; + + do{ + q*=ftmp[0]-w; + p*=ftmp[1]-w; + ftmp+=2; + }while(--c); + + if(m&1){ + /* odd order filter; slightly assymetric */ + /* the last coefficient */ + q*=ftmp[0]-w; + q*=q; + p*=p*(1.f-w*w); + }else{ + /* even order filter; still symmetric */ + q*=q*(1.f+w); + p*=p*(1.f-w); + } + + q=frexp(p+q,&qexp); + q=vorbis_fromdBlook(amp* + vorbis_invsqlook(q)* + vorbis_invsq2explook(qexp+m)- + ampoffset); + + do{ + curve[i++]*=q; + }while(map[i]==k); + } + vorbis_fpu_restore(fpu); +} + +#else + +#ifdef INT_LOOKUP +#include "lookup.c" /* catch this in the build system; we #include for + compilers (like gcc) that can't inline across + modules */ + +static int MLOOP_1[64]={ + 0,10,11,11, 12,12,12,12, 13,13,13,13, 13,13,13,13, + 14,14,14,14, 14,14,14,14, 14,14,14,14, 14,14,14,14, + 15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15, + 15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15, +}; + +static int MLOOP_2[64]={ + 0,4,5,5, 6,6,6,6, 7,7,7,7, 7,7,7,7, + 8,8,8,8, 8,8,8,8, 8,8,8,8, 8,8,8,8, + 9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9, + 9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9, +}; + +static int MLOOP_3[8]={0,1,2,2,3,3,3,3}; + + +/* side effect: changes *lsp to cosines of lsp */ +void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m, + float amp,float ampoffset){ + + /* 0 <= m < 256 */ + + /* set up for using all int later */ + int i; + int ampoffseti=rint(ampoffset*4096.f); + int ampi=rint(amp*16.f); + long *ilsp=alloca(m*sizeof(*ilsp)); + for(i=0;i>25])) + if(!(shift=MLOOP_2[(pi|qi)>>19])) + shift=MLOOP_3[(pi|qi)>>16]; + qi=(qi>>shift)*labs(ilsp[j-1]-wi); + pi=(pi>>shift)*labs(ilsp[j]-wi); + qexp+=shift; + } + if(!(shift=MLOOP_1[(pi|qi)>>25])) + if(!(shift=MLOOP_2[(pi|qi)>>19])) + shift=MLOOP_3[(pi|qi)>>16]; + + /* pi,qi normalized collectively, both tracked using qexp */ + + if(m&1){ + /* odd order filter; slightly assymetric */ + /* the last coefficient */ + qi=(qi>>shift)*labs(ilsp[j-1]-wi); + pi=(pi>>shift)<<14; + qexp+=shift; + + if(!(shift=MLOOP_1[(pi|qi)>>25])) + if(!(shift=MLOOP_2[(pi|qi)>>19])) + shift=MLOOP_3[(pi|qi)>>16]; + + pi>>=shift; + qi>>=shift; + qexp+=shift-14*((m+1)>>1); + + pi=((pi*pi)>>16); + qi=((qi*qi)>>16); + qexp=qexp*2+m; + + pi*=(1<<14)-((wi*wi)>>14); + qi+=pi>>14; + + }else{ + /* even order filter; still symmetric */ + + /* p*=p(1-w), q*=q(1+w), let normalization drift because it isn't + worth tracking step by step */ + + pi>>=shift; + qi>>=shift; + qexp+=shift-7*m; + + pi=((pi*pi)>>16); + qi=((qi*qi)>>16); + qexp=qexp*2+m; + + pi*=(1<<14)-wi; + qi*=(1<<14)+wi; + qi=(qi+pi)>>14; + + } + + + /* we've let the normalization drift because it wasn't important; + however, for the lookup, things must be normalized again. We + need at most one right shift or a number of left shifts */ + + if(qi&0xffff0000){ /* checks for 1.xxxxxxxxxxxxxxxx */ + qi>>=1; qexp++; + }else + while(qi && !(qi&0x8000)){ /* checks for 0.0xxxxxxxxxxxxxxx or less*/ + qi<<=1; qexp--; + } + + amp=vorbis_fromdBlook_i(ampi* /* n.4 */ + vorbis_invsqlook_i(qi,qexp)- + /* m.8, m+n<=8 */ + ampoffseti); /* 8.12[0] */ + + curve[i]*=amp; + while(map[++i]==k)curve[i]*=amp; + } +} + +#else + +/* old, nonoptimized but simple version for any poor sap who needs to + figure out what the hell this code does, or wants the other + fraction of a dB precision */ + +/* side effect: changes *lsp to cosines of lsp */ +void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m, + float amp,float ampoffset){ + int i; + float wdel=M_PI/ln; + for(i=0;i= i; j--) { + g[j-2] -= g[j]; + g[j] += g[j]; + } + } +} + +static int comp(const void *a,const void *b){ + if(*(float *)a<*(float *)b) + return(1); + else + return(-1); +} + +/* Newton-Raphson-Maehly actually functioned as a decent root finder, + but there are root sets for which it gets into limit cycles + (exacerbated by zero suppression) and fails. We can't afford to + fail, even if the failure is 1 in 100,000,000, so we now use + Laguerre and later polish with Newton-Raphson (which can then + afford to fail) */ + +#define EPSILON 10e-7 +static int Laguerre_With_Deflation(float *a,int ord,float *r){ + int i,m; + double lastdelta=0.f; + double *defl=alloca(sizeof(*defl)*(ord+1)); + for(i=0;i<=ord;i++)defl[i]=a[i]; + + for(m=ord;m>0;m--){ + double new=0.f,delta; + + /* iterate a root */ + while(1){ + double p=defl[m],pp=0.f,ppp=0.f,denom; + + /* eval the polynomial and its first two derivatives */ + for(i=m;i>0;i--){ + ppp = new*ppp + pp; + pp = new*pp + p; + p = new*p + defl[i-1]; + } + + /* Laguerre's method */ + denom=(m-1) * ((m-1)*pp*pp - m*p*ppp); + if(denom<0) + return(-1); /* complex root! The LPC generator handed us a bad filter */ + + if(pp>0){ + denom = pp + sqrt(denom); + if(denom-(EPSILON))denom=-(EPSILON); + } + + delta = m*p/denom; + new -= delta; + + if(delta<0.f)delta*=-1; + + if(fabs(delta/new)<10e-12)break; + lastdelta=delta; + } + + r[m-1]=new; + + /* forward deflation */ + + for(i=m;i>0;i--) + defl[i-1]+=new*defl[i]; + defl++; + + } + return(0); +} + + +/* for spit-and-polish only */ +static int Newton_Raphson(float *a,int ord,float *r){ + int i, k, count=0; + double error=1.f; + double *root=alloca(ord*sizeof(*root)); + + for(i=0; i1e-20){ + error=0; + + for(i=0; i= 0; k--) { + + pp= pp* rooti + p; + p = p * rooti + a[k]; + } + + delta = p/pp; + root[i] -= delta; + error+= delta*delta; + } + + if(count>40)return(-1); + + count++; + } + + /* Replaced the original bubble sort with a real sort. With your + help, we can eliminate the bubble sort in our lifetime. --Monty */ + + for(i=0; i>1; + int g1_order,g2_order; + float *g1=alloca(sizeof(*g1)*(order2+1)); + float *g2=alloca(sizeof(*g2)*(order2+1)); + float *g1r=alloca(sizeof(*g1r)*(order2+1)); + float *g2r=alloca(sizeof(*g2r)*(order2+1)); + int i; + + /* even and odd are slightly different base cases */ + g1_order=(m+1)>>1; + g2_order=(m) >>1; + + /* Compute the lengths of the x polynomials. */ + /* Compute the first half of K & R F1 & F2 polynomials. */ + /* Compute half of the symmetric and antisymmetric polynomials. */ + /* Remove the roots at +1 and -1. */ + + g1[g1_order] = 1.f; + for(i=1;i<=g1_order;i++) g1[g1_order-i] = lpc[i-1]+lpc[m-i]; + g2[g2_order] = 1.f; + for(i=1;i<=g2_order;i++) g2[g2_order-i] = lpc[i-1]-lpc[m-i]; + + if(g1_order>g2_order){ + for(i=2; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+2]; + }else{ + for(i=1; i<=g1_order;i++) g1[g1_order-i] -= g1[g1_order-i+1]; + for(i=1; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+1]; + } + + /* Convert into polynomials in cos(alpha) */ + cheby(g1,g1_order); + cheby(g2,g2_order); + + /* Find the roots of the 2 even polynomials.*/ + if(Laguerre_With_Deflation(g1,g1_order,g1r) || + Laguerre_With_Deflation(g2,g2_order,g2r)) + return(-1); + + Newton_Raphson(g1,g1_order,g1r); /* if it fails, it leaves g1r alone */ + Newton_Raphson(g2,g2_order,g2r); /* if it fails, it leaves g2r alone */ + + qsort(g1r,g1_order,sizeof(*g1r),comp); + qsort(g2r,g2_order,sizeof(*g2r),comp); + + for(i=0;i +#include +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "codebook.h" +#include "window.h" +#include "registry.h" +#include "psy.h" +#include "misc.h" + +/* simplistic, wasteful way of doing this (unique lookup for each + mode/submapping); there should be a central repository for + identical lookups. That will require minor work, so I'm putting it + off as low priority. + + Why a lookup for each backend in a given mode? Because the + blocksize is set by the mode, and low backend lookups may require + parameters from other areas of the mode/mapping */ + +static void mapping0_free_info(vorbis_info_mapping *i){ + vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)i; + if(info){ + memset(info,0,sizeof(*info)); + _ogg_free(info); + } +} + +static int ilog(unsigned int v){ + int ret=0; + if(v)--v; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +static void mapping0_pack(vorbis_info *vi,vorbis_info_mapping *vm, + oggpack_buffer *opb){ + int i; + vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)vm; + + /* another 'we meant to do it this way' hack... up to beta 4, we + packed 4 binary zeros here to signify one submapping in use. We + now redefine that to mean four bitflags that indicate use of + deeper features; bit0:submappings, bit1:coupling, + bit2,3:reserved. This is backward compatable with all actual uses + of the beta code. */ + + if(info->submaps>1){ + oggpack_write(opb,1,1); + oggpack_write(opb,info->submaps-1,4); + }else + oggpack_write(opb,0,1); + + if(info->coupling_steps>0){ + oggpack_write(opb,1,1); + oggpack_write(opb,info->coupling_steps-1,8); + + for(i=0;icoupling_steps;i++){ + oggpack_write(opb,info->coupling_mag[i],ilog(vi->channels)); + oggpack_write(opb,info->coupling_ang[i],ilog(vi->channels)); + } + }else + oggpack_write(opb,0,1); + + oggpack_write(opb,0,2); /* 2,3:reserved */ + + /* we don't write the channel submappings if we only have one... */ + if(info->submaps>1){ + for(i=0;ichannels;i++) + oggpack_write(opb,info->chmuxlist[i],4); + } + for(i=0;isubmaps;i++){ + oggpack_write(opb,0,8); /* time submap unused */ + oggpack_write(opb,info->floorsubmap[i],8); + oggpack_write(opb,info->residuesubmap[i],8); + } +} + +/* also responsible for range checking */ +static vorbis_info_mapping *mapping0_unpack(vorbis_info *vi,oggpack_buffer *opb){ + int i; + vorbis_info_mapping0 *info=_ogg_calloc(1,sizeof(*info)); + codec_setup_info *ci=vi->codec_setup; + memset(info,0,sizeof(*info)); + + if(oggpack_read(opb,1)) + info->submaps=oggpack_read(opb,4)+1; + else + info->submaps=1; + + if(oggpack_read(opb,1)){ + info->coupling_steps=oggpack_read(opb,8)+1; + + for(i=0;icoupling_steps;i++){ + int testM=info->coupling_mag[i]=oggpack_read(opb,ilog(vi->channels)); + int testA=info->coupling_ang[i]=oggpack_read(opb,ilog(vi->channels)); + + if(testM<0 || + testA<0 || + testM==testA || + testM>=vi->channels || + testA>=vi->channels) goto err_out; + } + + } + + if(oggpack_read(opb,2)>0)goto err_out; /* 2,3:reserved */ + + if(info->submaps>1){ + for(i=0;ichannels;i++){ + info->chmuxlist[i]=oggpack_read(opb,4); + if(info->chmuxlist[i]>=info->submaps)goto err_out; + } + } + for(i=0;isubmaps;i++){ + oggpack_read(opb,8); /* time submap unused */ + info->floorsubmap[i]=oggpack_read(opb,8); + if(info->floorsubmap[i]>=ci->floors)goto err_out; + info->residuesubmap[i]=oggpack_read(opb,8); + if(info->residuesubmap[i]>=ci->residues)goto err_out; + } + + return info; + + err_out: + mapping0_free_info(info); + return(NULL); +} + +#include "os.h" +#include "lpc.h" +#include "lsp.h" +#include "envelope.h" +#include "mdct.h" +#include "psy.h" +#include "scales.h" + +#if 0 +static long seq=0; +static ogg_int64_t total=0; +static float FLOOR1_fromdB_LOOKUP[256]={ + 1.0649863e-07F, 1.1341951e-07F, 1.2079015e-07F, 1.2863978e-07F, + 1.3699951e-07F, 1.4590251e-07F, 1.5538408e-07F, 1.6548181e-07F, + 1.7623575e-07F, 1.8768855e-07F, 1.9988561e-07F, 2.128753e-07F, + 2.2670913e-07F, 2.4144197e-07F, 2.5713223e-07F, 2.7384213e-07F, + 2.9163793e-07F, 3.1059021e-07F, 3.3077411e-07F, 3.5226968e-07F, + 3.7516214e-07F, 3.9954229e-07F, 4.2550680e-07F, 4.5315863e-07F, + 4.8260743e-07F, 5.1396998e-07F, 5.4737065e-07F, 5.8294187e-07F, + 6.2082472e-07F, 6.6116941e-07F, 7.0413592e-07F, 7.4989464e-07F, + 7.9862701e-07F, 8.5052630e-07F, 9.0579828e-07F, 9.6466216e-07F, + 1.0273513e-06F, 1.0941144e-06F, 1.1652161e-06F, 1.2409384e-06F, + 1.3215816e-06F, 1.4074654e-06F, 1.4989305e-06F, 1.5963394e-06F, + 1.7000785e-06F, 1.8105592e-06F, 1.9282195e-06F, 2.0535261e-06F, + 2.1869758e-06F, 2.3290978e-06F, 2.4804557e-06F, 2.6416497e-06F, + 2.8133190e-06F, 2.9961443e-06F, 3.1908506e-06F, 3.3982101e-06F, + 3.6190449e-06F, 3.8542308e-06F, 4.1047004e-06F, 4.3714470e-06F, + 4.6555282e-06F, 4.9580707e-06F, 5.2802740e-06F, 5.6234160e-06F, + 5.9888572e-06F, 6.3780469e-06F, 6.7925283e-06F, 7.2339451e-06F, + 7.7040476e-06F, 8.2047000e-06F, 8.7378876e-06F, 9.3057248e-06F, + 9.9104632e-06F, 1.0554501e-05F, 1.1240392e-05F, 1.1970856e-05F, + 1.2748789e-05F, 1.3577278e-05F, 1.4459606e-05F, 1.5399272e-05F, + 1.6400004e-05F, 1.7465768e-05F, 1.8600792e-05F, 1.9809576e-05F, + 2.1096914e-05F, 2.2467911e-05F, 2.3928002e-05F, 2.5482978e-05F, + 2.7139006e-05F, 2.8902651e-05F, 3.0780908e-05F, 3.2781225e-05F, + 3.4911534e-05F, 3.7180282e-05F, 3.9596466e-05F, 4.2169667e-05F, + 4.4910090e-05F, 4.7828601e-05F, 5.0936773e-05F, 5.4246931e-05F, + 5.7772202e-05F, 6.1526565e-05F, 6.5524908e-05F, 6.9783085e-05F, + 7.4317983e-05F, 7.9147585e-05F, 8.4291040e-05F, 8.9768747e-05F, + 9.5602426e-05F, 0.00010181521F, 0.00010843174F, 0.00011547824F, + 0.00012298267F, 0.00013097477F, 0.00013948625F, 0.00014855085F, + 0.00015820453F, 0.00016848555F, 0.00017943469F, 0.00019109536F, + 0.00020351382F, 0.00021673929F, 0.00023082423F, 0.00024582449F, + 0.00026179955F, 0.00027881276F, 0.00029693158F, 0.00031622787F, + 0.00033677814F, 0.00035866388F, 0.00038197188F, 0.00040679456F, + 0.00043323036F, 0.00046138411F, 0.00049136745F, 0.00052329927F, + 0.00055730621F, 0.00059352311F, 0.00063209358F, 0.00067317058F, + 0.00071691700F, 0.00076350630F, 0.00081312324F, 0.00086596457F, + 0.00092223983F, 0.00098217216F, 0.0010459992F, 0.0011139742F, + 0.0011863665F, 0.0012634633F, 0.0013455702F, 0.0014330129F, + 0.0015261382F, 0.0016253153F, 0.0017309374F, 0.0018434235F, + 0.0019632195F, 0.0020908006F, 0.0022266726F, 0.0023713743F, + 0.0025254795F, 0.0026895994F, 0.0028643847F, 0.0030505286F, + 0.0032487691F, 0.0034598925F, 0.0036847358F, 0.0039241906F, + 0.0041792066F, 0.0044507950F, 0.0047400328F, 0.0050480668F, + 0.0053761186F, 0.0057254891F, 0.0060975636F, 0.0064938176F, + 0.0069158225F, 0.0073652516F, 0.0078438871F, 0.0083536271F, + 0.0088964928F, 0.009474637F, 0.010090352F, 0.010746080F, + 0.011444421F, 0.012188144F, 0.012980198F, 0.013823725F, + 0.014722068F, 0.015678791F, 0.016697687F, 0.017782797F, + 0.018938423F, 0.020169149F, 0.021479854F, 0.022875735F, + 0.024362330F, 0.025945531F, 0.027631618F, 0.029427276F, + 0.031339626F, 0.033376252F, 0.035545228F, 0.037855157F, + 0.040315199F, 0.042935108F, 0.045725273F, 0.048696758F, + 0.051861348F, 0.055231591F, 0.058820850F, 0.062643361F, + 0.066714279F, 0.071049749F, 0.075666962F, 0.080584227F, + 0.085821044F, 0.091398179F, 0.097337747F, 0.10366330F, + 0.11039993F, 0.11757434F, 0.12521498F, 0.13335215F, + 0.14201813F, 0.15124727F, 0.16107617F, 0.17154380F, + 0.18269168F, 0.19456402F, 0.20720788F, 0.22067342F, + 0.23501402F, 0.25028656F, 0.26655159F, 0.28387361F, + 0.30232132F, 0.32196786F, 0.34289114F, 0.36517414F, + 0.38890521F, 0.41417847F, 0.44109412F, 0.46975890F, + 0.50028648F, 0.53279791F, 0.56742212F, 0.60429640F, + 0.64356699F, 0.68538959F, 0.72993007F, 0.77736504F, + 0.82788260F, 0.88168307F, 0.9389798F, 1.F, +}; + +#endif + +extern int *floor1_fit(vorbis_block *vb,vorbis_look_floor *look, + const float *logmdct, /* in */ + const float *logmask); +extern int *floor1_interpolate_fit(vorbis_block *vb,vorbis_look_floor *look, + int *A,int *B, + int del); +extern int floor1_encode(vorbis_block *vb,vorbis_look_floor *look, + int *post,int *ilogmask); + + +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; + backend_lookup_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 **ilogmaskch= _vorbis_block_alloc(vb,vi->channels*sizeof(*ilogmaskch)); + 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;ichannels;i++){ + float scale=4.f/n; + float scale_dB; + + float *pcm =vb->pcm[i]; + float *logfft =pcm; + + gmdct[i]=_vorbis_block_alloc(vb,n/2*sizeof(**gmdct)); + + scale_dB=todB(&scale); + +#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); +#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); +#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); + local_ampmax[i]=logfft[0]; + for(j=1;j>1]=scale_dB+.5f*todB(&temp); + 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); +#endif + + } + + { + float *noise = _vorbis_block_alloc(vb,n/2*sizeof(*noise)); + float *tone = _vorbis_block_alloc(vb,n/2*sizeof(*tone)); + + for(i=0;ichannels;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;jchannels==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); + } +#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); + } +#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. */ + + _vp_offset_and_mix(psy_look, + noise, + tone, + 1, + logmask); + +#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); + } +#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); + +#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); + } +#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); + +#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); +#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;kflr[info->floorsubmap[submap]], + floor_posts[i][0], + floor_posts[i][PACKETBLOBS/2], + k*65536/(PACKETBLOBS/2)); + for(k=PACKETBLOBS/2+1;kflr[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 */ + + { + float **res_bundle=alloca(sizeof(*res_bundle)*vi->channels); + float **couple_bundle=alloca(sizeof(*couple_bundle)*vi->channels); + int *zerobundle=alloca(sizeof(*zerobundle)*vi->channels); + int **sortindex=alloca(sizeof(*sortindex)*vi->channels); + float **mag_memo; + int **mag_sort; + + if(info->coupling_steps){ + mag_memo=_vp_quantize_couple_memo(vb, + &ci->psy_g_param, + psy_look, + info, + gmdct); + + mag_sort=_vp_quantize_couple_sort(vb, + psy_look, + info, + mag_memo); + } + + memset(sortindex,0,sizeof(*sortindex)*vi->channels); + if(psy_look->vi->normal_channel_p){ + for(i=0;ichannels;i++){ + float *mdct =gmdct[i]; + sortindex[i]=alloca(sizeof(**sortindex)*n/2); + _vp_noise_normalize_sort(psy_look,mdct,sortindex[i]); + } + } + + for(k=(vorbis_bitrate_managed(vb)?0:PACKETBLOBS/2); + k<=(vorbis_bitrate_managed(vb)?PACKETBLOBS-1:PACKETBLOBS/2); + k++){ + + /* start out our new packet blob with packet type and mode */ + /* Encode the packet type */ + oggpack_write(&vb->opb,0,1); + /* Encode the modenumber */ + /* Encode frame mode, pre,post windowsize, then dispatch */ + oggpack_write(&vb->opb,modenumber,b->modebits); + if(vb->W){ + oggpack_write(&vb->opb,vb->lW,1); + oggpack_write(&vb->opb,vb->nW,1); + } + + /* encode floor, compute masking curve, sep out residue */ + for(i=0;ichannels;i++){ + int submap=info->chmuxlist[i]; + float *mdct =gmdct[i]; + float *res =vb->pcm[i]; + int *ilogmask=ilogmaskch[i]= + _vorbis_block_alloc(vb,n/2*sizeof(**gmdct)); + + nonzero[i]=floor1_encode(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;jpsy_g_param.sliding_lowpass[vb->W][k]); + + _vp_noise_normalize(psy_look,res,res+n/2,sortindex[i]); + + +#if 0 + { + char buf[80]; + float work[n/2]; + for(j=0;jcoupling_steps){ + _vp_couple(k, + &ci->psy_g_param, + psy_look, + info, + vb->pcm, + mag_memo, + mag_sort, + ilogmaskch, + nonzero, + ci->psy_g_param.sliding_lowpass[vb->W][k]); + } + + /* classify and encode by submap */ + for(i=0;isubmaps;i++){ + int ch_in_bundle=0; + long **classifications; + int resnum=info->residuesubmap[i]; + + for(j=0;jchannels;j++){ + if(info->chmuxlist[j]==i){ + zerobundle[ch_in_bundle]=0; + if(nonzero[j])zerobundle[ch_in_bundle]=1; + res_bundle[ch_in_bundle]=vb->pcm[j]; + couple_bundle[ch_in_bundle++]=vb->pcm[j]+n/2; + } + } + + classifications=_residue_P[ci->residue_type[resnum]]-> + class(vb,b->residue[resnum],couple_bundle,zerobundle,ch_in_bundle); + + _residue_P[ci->residue_type[resnum]]-> + forward(vb,b->residue[resnum], + couple_bundle,NULL,zerobundle,ch_in_bundle,classifications); + } + + /* ok, done encoding. Mark this protopacket and prepare next. */ + oggpack_writealign(&vb->opb); + vbi->packetblob_markers[k]=oggpack_bytes(&vb->opb); + + } + + } + +#if 0 + seq++; + total+=ci->blocksizes[vb->W]/4+ci->blocksizes[vb->nW]/4; +#endif + return(0); +} + +static int mapping0_inverse(vorbis_block *vb,vorbis_info_mapping *l){ + vorbis_dsp_state *vd=vb->vd; + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + backend_lookup_state *b=vd->backend_state; + vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)l; + + int i,j; + long n=vb->pcmend=ci->blocksizes[vb->W]; + + float **pcmbundle=alloca(sizeof(*pcmbundle)*vi->channels); + int *zerobundle=alloca(sizeof(*zerobundle)*vi->channels); + + int *nonzero =alloca(sizeof(*nonzero)*vi->channels); + void **floormemo=alloca(sizeof(*floormemo)*vi->channels); + + /* recover the spectral envelope; store it in the PCM vector for now */ + for(i=0;ichannels;i++){ + int submap=info->chmuxlist[i]; + floormemo[i]=_floor_P[ci->floor_type[info->floorsubmap[submap]]]-> + inverse1(vb,b->flr[info->floorsubmap[submap]]); + if(floormemo[i]) + nonzero[i]=1; + else + nonzero[i]=0; + memset(vb->pcm[i],0,sizeof(*vb->pcm[i])*n/2); + } + + /* channel coupling can 'dirty' the nonzero listing */ + for(i=0;icoupling_steps;i++){ + if(nonzero[info->coupling_mag[i]] || + nonzero[info->coupling_ang[i]]){ + nonzero[info->coupling_mag[i]]=1; + nonzero[info->coupling_ang[i]]=1; + } + } + + /* recover the residue into our working vectors */ + for(i=0;isubmaps;i++){ + int ch_in_bundle=0; + for(j=0;jchannels;j++){ + if(info->chmuxlist[j]==i){ + if(nonzero[j]) + zerobundle[ch_in_bundle]=1; + else + zerobundle[ch_in_bundle]=0; + pcmbundle[ch_in_bundle++]=vb->pcm[j]; + } + } + + _residue_P[ci->residue_type[info->residuesubmap[i]]]-> + inverse(vb,b->residue[info->residuesubmap[i]], + pcmbundle,zerobundle,ch_in_bundle); + } + + /* channel coupling */ + for(i=info->coupling_steps-1;i>=0;i--){ + float *pcmM=vb->pcm[info->coupling_mag[i]]; + float *pcmA=vb->pcm[info->coupling_ang[i]]; + + for(j=0;j0) + 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; + } + } + } + + /* compute and apply spectral envelope */ + for(i=0;ichannels;i++){ + float *pcm=vb->pcm[i]; + int submap=info->chmuxlist[i]; + _floor_P[ci->floor_type[info->floorsubmap[submap]]]-> + inverse2(vb,b->flr[info->floorsubmap[submap]], + floormemo[i],pcm); + } + + /* transform the PCM data; takes PCM vector, vb; modifies PCM vector */ + /* only MDCT right now.... */ + for(i=0;ichannels;i++){ + float *pcm=vb->pcm[i]; + mdct_backward(b->transform[vb->W][0],pcm,pcm); + } + + /* window the data */ + for(i=0;ichannels;i++){ + float *pcm=vb->pcm[i]; + if(nonzero[i]) + _vorbis_apply_window(pcm,b->window,ci->blocksizes,vb->lW,vb->W,vb->nW); + else + for(j=0;j +#include +#include +#include +#include "vorbis/codec.h" +#include "mdct.h" +#include "os.h" +#include "misc.h" + +/* build lookups for trig functions; also pre-figure scaling and + some window function algebra. */ + +void mdct_init(mdct_lookup *lookup,int n){ + int *bitrev=_ogg_malloc(sizeof(*bitrev)*(n/4)); + DATA_TYPE *T=_ogg_malloc(sizeof(*T)*(n+n/4)); + + int i; + int n2=n>>1; + int log2n=lookup->log2n=rint(log((float)n)/log(2.f)); + lookup->n=n; + lookup->trig=T; + lookup->bitrev=bitrev; + +/* trig lookups... */ + + for(i=0;i>j;j++) + if((msb>>j)&i)acc|=1<scale=FLOAT_CONV(4.f/n); +} + +/* 8 point butterfly (in place, 4 register) */ +STIN void mdct_butterfly_8(DATA_TYPE *x){ + REG_TYPE r0 = x[6] + x[2]; + REG_TYPE r1 = x[6] - x[2]; + REG_TYPE r2 = x[4] + x[0]; + REG_TYPE r3 = x[4] - x[0]; + + x[6] = r0 + r2; + x[4] = r0 - r2; + + r0 = x[5] - x[1]; + r2 = x[7] - x[3]; + x[0] = r1 + r0; + x[2] = r1 - r0; + + r0 = x[5] + x[1]; + r1 = x[7] + x[3]; + x[3] = r2 + r3; + x[1] = r2 - r3; + x[7] = r1 + r0; + x[5] = r1 - r0; + +} + +/* 16 point butterfly (in place, 4 register) */ +STIN void mdct_butterfly_16(DATA_TYPE *x){ + REG_TYPE r0 = x[1] - x[9]; + REG_TYPE r1 = x[0] - x[8]; + + x[8] += x[0]; + x[9] += x[1]; + x[0] = MULT_NORM((r0 + r1) * cPI2_8); + x[1] = MULT_NORM((r0 - r1) * cPI2_8); + + r0 = x[3] - x[11]; + r1 = x[10] - x[2]; + x[10] += x[2]; + x[11] += x[3]; + x[2] = r0; + x[3] = r1; + + r0 = x[12] - x[4]; + r1 = x[13] - x[5]; + x[12] += x[4]; + x[13] += x[5]; + x[4] = MULT_NORM((r0 - r1) * cPI2_8); + x[5] = MULT_NORM((r0 + r1) * cPI2_8); + + r0 = x[14] - x[6]; + r1 = x[15] - x[7]; + x[14] += x[6]; + x[15] += x[7]; + x[6] = r0; + x[7] = r1; + + mdct_butterfly_8(x); + mdct_butterfly_8(x+8); +} + +/* 32 point butterfly (in place, 4 register) */ +STIN void mdct_butterfly_32(DATA_TYPE *x){ + REG_TYPE r0 = x[30] - x[14]; + REG_TYPE r1 = x[31] - x[15]; + + x[30] += x[14]; + x[31] += x[15]; + x[14] = r0; + x[15] = r1; + + r0 = x[28] - x[12]; + r1 = x[29] - x[13]; + x[28] += x[12]; + x[29] += x[13]; + x[12] = MULT_NORM( r0 * cPI1_8 - r1 * cPI3_8 ); + x[13] = MULT_NORM( r0 * cPI3_8 + r1 * cPI1_8 ); + + r0 = x[26] - x[10]; + r1 = x[27] - x[11]; + x[26] += x[10]; + x[27] += x[11]; + x[10] = MULT_NORM(( r0 - r1 ) * cPI2_8); + x[11] = MULT_NORM(( r0 + r1 ) * cPI2_8); + + r0 = x[24] - x[8]; + r1 = x[25] - x[9]; + x[24] += x[8]; + x[25] += x[9]; + x[8] = MULT_NORM( r0 * cPI3_8 - r1 * cPI1_8 ); + x[9] = MULT_NORM( r1 * cPI3_8 + r0 * cPI1_8 ); + + r0 = x[22] - x[6]; + r1 = x[7] - x[23]; + x[22] += x[6]; + x[23] += x[7]; + x[6] = r1; + x[7] = r0; + + r0 = x[4] - x[20]; + r1 = x[5] - x[21]; + x[20] += x[4]; + x[21] += x[5]; + x[4] = MULT_NORM( r1 * cPI1_8 + r0 * cPI3_8 ); + x[5] = MULT_NORM( r1 * cPI3_8 - r0 * cPI1_8 ); + + r0 = x[2] - x[18]; + r1 = x[3] - x[19]; + x[18] += x[2]; + x[19] += x[3]; + x[2] = MULT_NORM(( r1 + r0 ) * cPI2_8); + x[3] = MULT_NORM(( r1 - r0 ) * cPI2_8); + + r0 = x[0] - x[16]; + r1 = x[1] - x[17]; + x[16] += x[0]; + x[17] += x[1]; + x[0] = MULT_NORM( r1 * cPI3_8 + r0 * cPI1_8 ); + x[1] = MULT_NORM( r1 * cPI1_8 - r0 * cPI3_8 ); + + mdct_butterfly_16(x); + mdct_butterfly_16(x+16); + +} + +/* N point first stage butterfly (in place, 2 register) */ +STIN void mdct_butterfly_first(DATA_TYPE *T, + DATA_TYPE *x, + int points){ + + DATA_TYPE *x1 = x + points - 8; + DATA_TYPE *x2 = x + (points>>1) - 8; + REG_TYPE r0; + REG_TYPE r1; + + do{ + + r0 = x1[6] - x2[6]; + r1 = x1[7] - x2[7]; + x1[6] += x2[6]; + x1[7] += x2[7]; + x2[6] = MULT_NORM(r1 * T[1] + r0 * T[0]); + x2[7] = MULT_NORM(r1 * T[0] - r0 * T[1]); + + r0 = x1[4] - x2[4]; + r1 = x1[5] - x2[5]; + x1[4] += x2[4]; + x1[5] += x2[5]; + x2[4] = MULT_NORM(r1 * T[5] + r0 * T[4]); + x2[5] = MULT_NORM(r1 * T[4] - r0 * T[5]); + + r0 = x1[2] - x2[2]; + r1 = x1[3] - x2[3]; + x1[2] += x2[2]; + x1[3] += x2[3]; + x2[2] = MULT_NORM(r1 * T[9] + r0 * T[8]); + x2[3] = MULT_NORM(r1 * T[8] - r0 * T[9]); + + r0 = x1[0] - x2[0]; + r1 = x1[1] - x2[1]; + x1[0] += x2[0]; + x1[1] += x2[1]; + x2[0] = MULT_NORM(r1 * T[13] + r0 * T[12]); + x2[1] = MULT_NORM(r1 * T[12] - r0 * T[13]); + + x1-=8; + x2-=8; + T+=16; + + }while(x2>=x); +} + +/* N/stage point generic N stage butterfly (in place, 2 register) */ +STIN void mdct_butterfly_generic(DATA_TYPE *T, + DATA_TYPE *x, + int points, + int trigint){ + + DATA_TYPE *x1 = x + points - 8; + DATA_TYPE *x2 = x + (points>>1) - 8; + REG_TYPE r0; + REG_TYPE r1; + + do{ + + r0 = x1[6] - x2[6]; + r1 = x1[7] - x2[7]; + x1[6] += x2[6]; + x1[7] += x2[7]; + x2[6] = MULT_NORM(r1 * T[1] + r0 * T[0]); + x2[7] = MULT_NORM(r1 * T[0] - r0 * T[1]); + + T+=trigint; + + r0 = x1[4] - x2[4]; + r1 = x1[5] - x2[5]; + x1[4] += x2[4]; + x1[5] += x2[5]; + x2[4] = MULT_NORM(r1 * T[1] + r0 * T[0]); + x2[5] = MULT_NORM(r1 * T[0] - r0 * T[1]); + + T+=trigint; + + r0 = x1[2] - x2[2]; + r1 = x1[3] - x2[3]; + x1[2] += x2[2]; + x1[3] += x2[3]; + x2[2] = MULT_NORM(r1 * T[1] + r0 * T[0]); + x2[3] = MULT_NORM(r1 * T[0] - r0 * T[1]); + + T+=trigint; + + r0 = x1[0] - x2[0]; + r1 = x1[1] - x2[1]; + x1[0] += x2[0]; + x1[1] += x2[1]; + x2[0] = MULT_NORM(r1 * T[1] + r0 * T[0]); + x2[1] = MULT_NORM(r1 * T[0] - r0 * T[1]); + + T+=trigint; + x1-=8; + x2-=8; + + }while(x2>=x); +} + +STIN void mdct_butterflies(mdct_lookup *init, + DATA_TYPE *x, + int points){ + + DATA_TYPE *T=init->trig; + int stages=init->log2n-5; + int i,j; + + if(--stages>0){ + mdct_butterfly_first(T,x,points); + } + + for(i=1;--stages>0;i++){ + for(j=0;j<(1<>i)*j,points>>i,4<trig)_ogg_free(l->trig); + if(l->bitrev)_ogg_free(l->bitrev); + memset(l,0,sizeof(*l)); + } +} + +STIN void mdct_bitreverse(mdct_lookup *init, + DATA_TYPE *x){ + int n = init->n; + int *bit = init->bitrev; + DATA_TYPE *w0 = x; + DATA_TYPE *w1 = x = w0+(n>>1); + DATA_TYPE *T = init->trig+n; + + do{ + DATA_TYPE *x0 = x+bit[0]; + DATA_TYPE *x1 = x+bit[1]; + + REG_TYPE r0 = x0[1] - x1[1]; + REG_TYPE r1 = x0[0] + x1[0]; + REG_TYPE r2 = MULT_NORM(r1 * T[0] + r0 * T[1]); + REG_TYPE r3 = MULT_NORM(r1 * T[1] - r0 * T[0]); + + w1 -= 4; + + r0 = HALVE(x0[1] + x1[1]); + r1 = HALVE(x0[0] - x1[0]); + + w0[0] = r0 + r2; + w1[2] = r0 - r2; + w0[1] = r1 + r3; + w1[3] = r3 - r1; + + x0 = x+bit[2]; + x1 = x+bit[3]; + + r0 = x0[1] - x1[1]; + r1 = x0[0] + x1[0]; + r2 = MULT_NORM(r1 * T[2] + r0 * T[3]); + r3 = MULT_NORM(r1 * T[3] - r0 * T[2]); + + r0 = HALVE(x0[1] + x1[1]); + r1 = HALVE(x0[0] - x1[0]); + + w0[2] = r0 + r2; + w1[0] = r0 - r2; + w0[3] = r1 + r3; + w1[1] = r3 - r1; + + T += 4; + bit += 4; + w0 += 4; + + }while(w0n; + int n2=n>>1; + int n4=n>>2; + + /* rotate */ + + DATA_TYPE *iX = in+n2-7; + DATA_TYPE *oX = out+n2+n4; + DATA_TYPE *T = init->trig+n4; + + do{ + oX -= 4; + oX[0] = MULT_NORM(-iX[2] * T[3] - iX[0] * T[2]); + oX[1] = MULT_NORM (iX[0] * T[3] - iX[2] * T[2]); + oX[2] = MULT_NORM(-iX[6] * T[1] - iX[4] * T[0]); + oX[3] = MULT_NORM (iX[4] * T[1] - iX[6] * T[0]); + iX -= 8; + T += 4; + }while(iX>=in); + + iX = in+n2-8; + oX = out+n2+n4; + T = init->trig+n4; + + do{ + T -= 4; + oX[0] = MULT_NORM (iX[4] * T[3] + iX[6] * T[2]); + oX[1] = MULT_NORM (iX[4] * T[2] - iX[6] * T[3]); + oX[2] = MULT_NORM (iX[0] * T[1] + iX[2] * T[0]); + oX[3] = MULT_NORM (iX[0] * T[0] - iX[2] * T[1]); + iX -= 8; + oX += 4; + }while(iX>=in); + + mdct_butterflies(init,out+n2,n2); + mdct_bitreverse(init,out); + + /* roatate + window */ + + { + DATA_TYPE *oX1=out+n2+n4; + DATA_TYPE *oX2=out+n2+n4; + DATA_TYPE *iX =out; + T =init->trig+n2; + + do{ + oX1-=4; + + oX1[3] = MULT_NORM (iX[0] * T[1] - iX[1] * T[0]); + oX2[0] = -MULT_NORM (iX[0] * T[0] + iX[1] * T[1]); + + oX1[2] = MULT_NORM (iX[2] * T[3] - iX[3] * T[2]); + oX2[1] = -MULT_NORM (iX[2] * T[2] + iX[3] * T[3]); + + oX1[1] = MULT_NORM (iX[4] * T[5] - iX[5] * T[4]); + oX2[2] = -MULT_NORM (iX[4] * T[4] + iX[5] * T[5]); + + oX1[0] = MULT_NORM (iX[6] * T[7] - iX[7] * T[6]); + oX2[3] = -MULT_NORM (iX[6] * T[6] + iX[7] * T[7]); + + oX2+=4; + iX += 8; + T += 8; + }while(iXoX2); + } +} + +void mdct_forward(mdct_lookup *init, DATA_TYPE *in, DATA_TYPE *out){ + int n=init->n; + int n2=n>>1; + int n4=n>>2; + int n8=n>>3; + DATA_TYPE *w=alloca(n*sizeof(*w)); /* forward needs working space */ + DATA_TYPE *w2=w+n2; + + /* rotate */ + + /* window + rotate + step 1 */ + + REG_TYPE r0; + REG_TYPE r1; + DATA_TYPE *x0=in+n2+n4; + DATA_TYPE *x1=x0+1; + DATA_TYPE *T=init->trig+n2; + + int i=0; + + for(i=0;itrig+n2; + x0=out+n2; + + for(i=0;iscale); + x0[0] =MULT_NORM((w[0]*T[1]-w[1]*T[0])*init->scale); + w+=2; + T+=2; + } +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/mdct.h b/src/add-ons/media/plugins/vorbis/libvorbis/mdct.h new file mode 100644 index 0000000000..18be976327 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/mdct.h @@ -0,0 +1,83 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: modified discrete cosine transform prototypes + last mod: $Id: mdct.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _OGG_mdct_H_ +#define _OGG_mdct_H_ + +#include "vorbis/codec.h" + + + + + +/*#define MDCT_INTEGERIZED <- be warned there could be some hurt left here*/ +#ifdef MDCT_INTEGERIZED + +#define DATA_TYPE int +#define REG_TYPE register int +#define TRIGBITS 14 +#define cPI3_8 6270 +#define cPI2_8 11585 +#define cPI1_8 15137 + +#define FLOAT_CONV(x) ((int)((x)*(1<>TRIGBITS) +#define HALVE(x) ((x)>>1) + +#else + +#define DATA_TYPE float +#define REG_TYPE float +#define cPI3_8 .38268343236508977175F +#define cPI2_8 .70710678118654752441F +#define cPI1_8 .92387953251128675613F + +#define FLOAT_CONV(x) (x) +#define MULT_NORM(x) (x) +#define HALVE(x) ((x)*.5f) + +#endif + + +typedef struct { + int n; + int log2n; + + DATA_TYPE *trig; + int *bitrev; + + DATA_TYPE scale; +} mdct_lookup; + +extern void mdct_init(mdct_lookup *lookup,int n); +extern void mdct_clear(mdct_lookup *l); +extern void mdct_forward(mdct_lookup *init, DATA_TYPE *in, DATA_TYPE *out); +extern void mdct_backward(mdct_lookup *init, DATA_TYPE *in, DATA_TYPE *out); + +#endif + + + + + + + + + + + + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/misc.h b/src/add-ons/media/plugins/vorbis/libvorbis/misc.h new file mode 100644 index 0000000000..f6fff29314 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/misc.h @@ -0,0 +1,52 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: miscellaneous prototypes + last mod: $Id: misc.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_RANDOM_H_ +#define _V_RANDOM_H_ +#include "vorbis/codec.h" + +extern int analysis_noisy; + +extern void *_vorbis_block_alloc(vorbis_block *vb,long bytes); +extern void _vorbis_block_ripcord(vorbis_block *vb); +extern void _analysis_output(char *base,int i,float *v,int n,int bark,int dB, + ogg_int64_t off); + +#ifdef DEBUG_MALLOC + +#define _VDBG_GRAPHFILE "malloc.m" +extern void *_VDBG_malloc(void *ptr,long bytes,char *file,long line); +extern void _VDBG_free(void *ptr,char *file,long line); + +#ifndef MISC_C +#undef _ogg_malloc +#undef _ogg_calloc +#undef _ogg_realloc +#undef _ogg_free + +#define _ogg_malloc(x) _VDBG_malloc(NULL,(x),__FILE__,__LINE__) +#define _ogg_calloc(x,y) _VDBG_malloc(NULL,(x)*(y),__FILE__,__LINE__) +#define _ogg_realloc(x,y) _VDBG_malloc((x),(y),__FILE__,__LINE__) +#define _ogg_free(x) _VDBG_free((x),__FILE__,__LINE__) +#endif +#endif + +#endif + + + + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/os.h b/src/add-ons/media/plugins/vorbis/libvorbis/os.h new file mode 100644 index 0000000000..e5f46081aa --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/os.h @@ -0,0 +1,143 @@ +#ifndef _OS_H +#define _OS_H +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: #ifdef jail to whip a few platforms into the UNIX ideal. + last mod: $Id: os.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include + +#include "misc.h" + +#ifndef _V_IFDEFJAIL_H_ +# define _V_IFDEFJAIL_H_ + +# ifdef __GNUC__ +# define STIN static __inline__ +# elif _WIN32 +# define STIN static __inline +#else +# define STIN static +#endif + +#ifndef M_PI +# define M_PI (3.1415926536f) +#endif + +#ifdef _WIN32 +# include +# define rint(x) (floor((x)+0.5f)) +# define NO_FLOAT_MATH_LIB +# define FAST_HYPOT(a, b) sqrt((a)*(a) + (b)*(b)) +#endif + +#ifndef FAST_HYPOT +# define FAST_HYPOT hypot +#endif + +#endif + +#ifdef HAVE_ALLOCA_H +# include +#endif + +#ifdef USE_MEMORY_H +# include +#endif + +#ifndef min +# define min(x,y) ((x)>(y)?(y):(x)) +#endif + +#ifndef max +# define max(x,y) ((x)<(y)?(y):(x)) +#endif + +#if defined(__i386__) && defined(__GNUC__) && !defined(__BEOS__) +# define VORBIS_FPU_CONTROL +/* both GCC and MSVC are kinda stupid about rounding/casting to int. + Because of encapsulation constraints (GCC can't see inside the asm + block and so we end up doing stupid things like a store/load that + is collectively a noop), we do it this way */ + +/* we must set up the fpu before this works!! */ + +typedef ogg_int16_t vorbis_fpu_control; + +static inline void vorbis_fpu_setround(vorbis_fpu_control *fpu){ + ogg_int16_t ret; + ogg_int16_t temp; + __asm__ __volatile__("fnstcw %0\n\t" + "movw %0,%%dx\n\t" + "orw $62463,%%dx\n\t" + "movw %%dx,%1\n\t" + "fldcw %1\n\t":"=m"(ret):"m"(temp): "dx"); + *fpu=ret; +} + +static inline void vorbis_fpu_restore(vorbis_fpu_control fpu){ + __asm__ __volatile__("fldcw %0":: "m"(fpu)); +} + +/* assumes the FPU is in round mode! */ +static inline int vorbis_ftoi(double f){ /* yes, double! Otherwise, + we get extra fst/fld to + truncate precision */ + int i; + __asm__("fistl %0": "=m"(i) : "t"(f)); + return(i); +} +#endif + + +#if defined(_WIN32) && !defined(__GNUC__) && !defined(__BORLANDC__) +# define VORBIS_FPU_CONTROL + +typedef ogg_int16_t vorbis_fpu_control; + +static __inline int vorbis_ftoi(double f){ + int i; + __asm{ + fld f + fistp i + } + return i; +} + +static __inline void vorbis_fpu_setround(vorbis_fpu_control *fpu){ +} + +static __inline void vorbis_fpu_restore(vorbis_fpu_control fpu){ +} + +#endif + + +#ifndef VORBIS_FPU_CONTROL + +typedef int vorbis_fpu_control; + +static int vorbis_ftoi(double f){ + return (int)(f+.5); +} + +/* We don't have special code for this compiler/arch, so do it the slow way */ +# define vorbis_fpu_setround(vorbis_fpu_control) {} +# define vorbis_fpu_restore(vorbis_fpu_control) {} + +#endif + +#endif /* _OS_H */ diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/psy.c b/src/add-ons/media/plugins/vorbis/libvorbis/psy.c new file mode 100644 index 0000000000..67dbd11e27 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/psy.c @@ -0,0 +1,1131 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: psychoacoustics not including preecho + last mod: $Id: psy.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" + +#include "masking.h" +#include "psy.h" +#include "os.h" +#include "lpc.h" +#include "smallft.h" +#include "scales.h" +#include "misc.h" + +#define NEGINF -9999.f +static double stereo_threshholds[]={0.0, .5, 1.0, 1.5, 2.5, 4.5, 8.5, 16.5, 9e10}; + +vorbis_look_psy_global *_vp_global_look(vorbis_info *vi){ + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *gi=&ci->psy_g_param; + vorbis_look_psy_global *look=_ogg_calloc(1,sizeof(*look)); + + look->channels=vi->channels; + + look->ampmax=-9999.; + look->gi=gi; + return(look); +} + +void _vp_global_free(vorbis_look_psy_global *look){ + if(look){ + memset(look,0,sizeof(*look)); + _ogg_free(look); + } +} + +void _vi_gpsy_free(vorbis_info_psy_global *i){ + if(i){ + memset(i,0,sizeof(*i)); + _ogg_free(i); + } +} + +void _vi_psy_free(vorbis_info_psy *i){ + if(i){ + memset(i,0,sizeof(*i)); + _ogg_free(i); + } +} + +static void min_curve(float *c, + float *c2){ + int i; + for(i=0;ic[i])c[i]=c2[i]; +} + +static void attenuate_curve(float *c,float att){ + int i; + for(i=0;iATH[j+k+ath_offset])min=ATH[j+k+ath_offset]; + }else{ + if(min>ATH[MAX_ATH-1])min=ATH[MAX_ATH-1]; + } + ath[j]=min; + } + + /* copy curves into working space, replicate the 50dB curve to 30 + and 40, replicate the 100dB curve to 110 */ + for(j=0;j<6;j++) + memcpy(workc[i][j+2],tonemasks[i][j],EHMER_MAX*sizeof(*tonemasks[i][j])); + memcpy(workc[i][0],tonemasks[i][0],EHMER_MAX*sizeof(*tonemasks[i][0])); + memcpy(workc[i][1],tonemasks[i][0],EHMER_MAX*sizeof(*tonemasks[i][0])); + + /* apply centered curve boost/decay */ + for(j=0;j0)adj=0.; + if(adj>0. && center_boost<0)adj=0.; + workc[i][j][k]+=adj; + } + } + + /* normalize curves so the driving amplitude is 0dB */ + /* make temp curves with the ATH overlayed */ + for(j=0;j an eighth of an octave and that the eighth + octave values may also be composited. */ + + /* which octave curves will we be compositing? */ + bin=floor(fromOC(i*.5)/binHz); + lo_curve= ceil(toOC(bin*binHz+1)*2); + hi_curve= floor(toOC((bin+1)*binHz)*2); + if(lo_curve>i)lo_curve=i; + if(lo_curve<0)lo_curve=0; + if(hi_curve>=P_BANDS)hi_curve=P_BANDS-1; + + for(m=0;mn)lo_bin=n; + if(lo_binn)hi_bin=n; + + for(;lworkc[k][m][j]) + brute_buffer[l]=workc[k][m][j]; + } + + for(;lworkc[k][m][EHMER_MAX-1]) + brute_buffer[l]=workc[k][m][EHMER_MAX-1]; + + } + + /* be equally paranoid about being valid up to next half ocatve */ + if(i+1n)lo_bin=n; + if(lo_binn)hi_bin=n; + + for(;lworkc[k][m][j]) + brute_buffer[l]=workc[k][m][j]; + } + + for(;lworkc[k][m][EHMER_MAX-1]) + brute_buffer[l]=workc[k][m][EHMER_MAX-1]; + + } + + + for(j=0;j=n){ + ret[i][m][j+2]=-999.; + }else{ + ret[i][m][j+2]=brute_buffer[bin]; + } + } + } + + /* add fenceposts */ + for(j=0;j-200.f)break; + ret[i][m][0]=j; + + for(j=EHMER_MAX-1;j>EHMER_OFFSET+1;j--) + if(ret[i][m][j+2]>-200.f) + break; + ret[i][m][1]=j; + + } + } + + return(ret); +} + +void _vp_psy_init(vorbis_look_psy *p,vorbis_info_psy *vi, + vorbis_info_psy_global *gi,int n,long rate){ + long i,j,lo=-99,hi=0; + long maxoc; + memset(p,0,sizeof(*p)); + + p->eighth_octave_lines=gi->eighth_octave_lines; + p->shiftoc=rint(log(gi->eighth_octave_lines*8.f)/log(2.f))-1; + + p->firstoc=toOC(.25f*rate*.5/n)*(1<<(p->shiftoc+1))-gi->eighth_octave_lines; + maxoc=toOC((n+.25f)*rate*.5/n)*(1<<(p->shiftoc+1))+.5f; + p->total_octave_lines=maxoc-p->firstoc+1; + p->ath=_ogg_malloc(n*sizeof(*p->ath)); + + p->octave=_ogg_malloc(n*sizeof(*p->octave)); + p->bark=_ogg_malloc(n*sizeof(*p->bark)); + p->vi=vi; + p->n=n; + p->rate=rate; + + /* set up the lookups for a given blocksize and sample rate */ + + for(i=0,j=0;iath[j]=base+100.; + base+=delta; + } + } + } + + for(i=0;inoisewindowlominnoisewindowlo);lo++); + + for(;hinoisewindowhimin || + toBARK(rate/(2*n)*hi)<(bark+vi->noisewindowhi));hi++); + + p->bark[i]=(lo<<16)+hi; + + } + + for(i=0;ioctave[i]=toOC((i+.25f)*.5*rate/n)*(1<<(p->shiftoc+1))+.5f; + + p->tonecurves=setup_tone_curves(vi->toneatt,rate*.5/n,n, + vi->tone_centerboost,vi->tone_decay); + + /* set up rolling noise median */ + p->noiseoffset=_ogg_malloc(P_NOISECURVES*sizeof(*p->noiseoffset)); + for(i=0;inoiseoffset[i]=_ogg_malloc(n*sizeof(**p->noiseoffset)); + + for(i=0;i=P_BANDS-1)halfoc=P_BANDS-1; + inthalfoc=(int)halfoc; + del=halfoc-inthalfoc; + + for(j=0;jnoiseoffset[j][i]= + p->vi->noiseoff[j][inthalfoc]*(1.-del) + + p->vi->noiseoff[j][inthalfoc+1]*del; + + } +#if 0 + { + static int ls=0; + _analysis_output_always("noiseoff0",ls,p->noiseoffset[0],n,1,0,0); + _analysis_output_always("noiseoff1",ls,p->noiseoffset[1],n,1,0,0); + _analysis_output_always("noiseoff2",ls++,p->noiseoffset[2],n,1,0,0); + } +#endif +} + +void _vp_psy_clear(vorbis_look_psy *p){ + int i,j; + if(p){ + if(p->ath)_ogg_free(p->ath); + if(p->octave)_ogg_free(p->octave); + if(p->bark)_ogg_free(p->bark); + if(p->tonecurves){ + for(i=0;itonecurves[i][j]); + } + _ogg_free(p->tonecurves[i]); + } + _ogg_free(p->tonecurves); + } + if(p->noiseoffset){ + for(i=0;inoiseoffset[i]); + } + _ogg_free(p->noiseoffset); + } + memset(p,0,sizeof(*p)); + } +} + +/* octave/(8*eighth_octave_lines) x scale and dB y scale */ +static void seed_curve(float *seed, + const float **curves, + float amp, + int oc, int n, + int linesper,float dBoffset){ + int i,post1; + int seedptr; + const float *posts,*curve; + + int choice=(int)((amp+dBoffset-P_LEVEL_0)*.1f); + choice=max(choice,0); + choice=min(choice,P_LEVELS-1); + posts=curves[choice]; + curve=posts+2; + post1=(int)posts[1]; + seedptr=oc+(posts[0]-EHMER_OFFSET)*linesper-(linesper>>1); + + for(i=posts[0];i0){ + float lin=amp+curve[i]; + if(seed[seedptr]=n)break; + } +} + +static void seed_loop(vorbis_look_psy *p, + const float ***curves, + const float *f, + const float *flr, + float *seed, + float specmax){ + vorbis_info_psy *vi=p->vi; + long n=p->n,i; + float dBoffset=vi->max_curve_dB-specmax; + + /* prime the working vector with peak values */ + + for(i=0;ioctave[i]; + while(i+1octave[i+1]==oc){ + i++; + if(f[i]>max)max=f[i]; + } + + if(max+6.f>flr[i]){ + oc=oc>>p->shiftoc; + + if(oc>=P_BANDS)oc=P_BANDS-1; + if(oc<0)oc=0; + + seed_curve(seed, + curves[oc], + max, + p->octave[i]-p->firstoc, + p->total_octave_lines, + p->eighth_octave_lines, + dBoffset); + } + } +} + +static void seed_chase(float *seeds, int linesper, long n){ + long *posstack=alloca(n*sizeof(*posstack)); + float *ampstack=alloca(n*sizeof(*ampstack)); + long stack=0; + long pos=0; + long i; + + for(i=0;i1 && ampstack[stack-1]<=ampstack[stack-2] && + iampstack[i]){ + endpos=posstack[i+1]; + }else{ + endpos=posstack[i]+linesper+1; /* +1 is important, else bin 0 is + discarded in short frames */ + } + if(endpos>n)endpos=n; + for(;pos +static void max_seeds(vorbis_look_psy *p, + float *seed, + float *flr){ + long n=p->total_octave_lines; + int linesper=p->eighth_octave_lines; + long linpos=0; + long pos; + + seed_chase(seed,linesper,n); /* for masking */ + + pos=p->octave[0]-p->firstoc-(linesper>>1); + + while(linpos+1n){ + float minV=seed[pos]; + long end=((p->octave[linpos]+p->octave[linpos+1])>>1)-p->firstoc; + if(minV>p->vi->tone_abs_limit)minV=p->vi->tone_abs_limit; + while(pos+1<=end){ + pos++; + if((seed[pos]>NEGINF && seed[pos]firstoc; + for(;linposn && p->octave[linpos]<=end;linpos++) + if(flr[linpos]total_octave_lines-1]; + for(;linposn;linpos++) + if(flr[linpos]> 16; + if( lo>=0 ) break; + hi = b[i] & 0xffff; + + tN = N[hi] + N[-lo]; + tX = X[hi] - X[-lo]; + tXX = XX[hi] + XX[-lo]; + tY = Y[hi] + Y[-lo]; + tXY = XY[hi] - XY[-lo]; + + A = tY * tXX - tX * tXY; + B = tN * tXY - tX * tY; + D = tN * tXX - tX * tX; + R = (A + fi * B) / D; + if (R < 0.f) + R = 0.f; + + noise[i] = R - offset; + } + + for ( ; hi < n; i++, fi += 1.f) { + + lo = b[i] >> 16; + hi = b[i] & 0xffff; + + tN = N[hi] - N[lo]; + tX = X[hi] - X[lo]; + tXX = XX[hi] - XX[lo]; + tY = Y[hi] - Y[lo]; + tXY = XY[hi] - XY[lo]; + + A = tY * tXX - tX * tXY; + B = tN * tXY - tX * tY; + D = tN * tXX - tX * tX; + R = (A + fi * B) / D; + if (R < 0.f) R = 0.f; + + noise[i] = R - offset; + } + for ( ; i < n; i++, fi += 1.f) { + + R = (A + fi * B) / D; + if (R < 0.f) R = 0.f; + + noise[i] = R - offset; + } + + if (fixed <= 0) return; + + for (i = 0, fi = 0.f; i < (fixed + 1) / 2; i++, fi += 1.f) { + hi = i + fixed / 2; + lo = hi - fixed; + + tN = N[hi] + N[-lo]; + tX = X[hi] - X[-lo]; + tXX = XX[hi] + XX[-lo]; + tY = Y[hi] + Y[-lo]; + tXY = XY[hi] - XY[-lo]; + + + A = tY * tXX - tX * tXY; + B = tN * tXY - tX * tY; + D = tN * tXX - tX * tX; + R = (A + fi * B) / D; + + if (R > 0.f && R - offset < noise[i]) noise[i] = R - offset; + } + for ( ; hi < n; i++, fi += 1.f) { + + hi = i + fixed / 2; + lo = hi - fixed; + + tN = N[hi] - N[lo]; + tX = X[hi] - X[lo]; + tXX = XX[hi] - XX[lo]; + tY = Y[hi] - Y[lo]; + tXY = XY[hi] - XY[lo]; + + A = tY * tXX - tX * tXY; + B = tN * tXY - tX * tY; + D = tN * tXX - tX * tX; + R = (A + fi * B) / D; + + if (R > 0.f && R - offset < noise[i]) noise[i] = R - offset; + } + for ( ; i < n; i++, fi += 1.f) { + R = (A + fi * B) / D; + if (R > 0.f && R - offset < noise[i]) noise[i] = R - offset; + } +} + +static float FLOOR1_fromdB_INV_LOOKUP[256]={ + 0.F, 8.81683e+06F, 8.27882e+06F, 7.77365e+06F, + 7.29930e+06F, 6.85389e+06F, 6.43567e+06F, 6.04296e+06F, + 5.67422e+06F, 5.32798e+06F, 5.00286e+06F, 4.69759e+06F, + 4.41094e+06F, 4.14178e+06F, 3.88905e+06F, 3.65174e+06F, + 3.42891e+06F, 3.21968e+06F, 3.02321e+06F, 2.83873e+06F, + 2.66551e+06F, 2.50286e+06F, 2.35014e+06F, 2.20673e+06F, + 2.07208e+06F, 1.94564e+06F, 1.82692e+06F, 1.71544e+06F, + 1.61076e+06F, 1.51247e+06F, 1.42018e+06F, 1.33352e+06F, + 1.25215e+06F, 1.17574e+06F, 1.10400e+06F, 1.03663e+06F, + 973377.F, 913981.F, 858210.F, 805842.F, + 756669.F, 710497.F, 667142.F, 626433.F, + 588208.F, 552316.F, 518613.F, 486967.F, + 457252.F, 429351.F, 403152.F, 378551.F, + 355452.F, 333762.F, 313396.F, 294273.F, + 276316.F, 259455.F, 243623.F, 228757.F, + 214798.F, 201691.F, 189384.F, 177828.F, + 166977.F, 156788.F, 147221.F, 138237.F, + 129802.F, 121881.F, 114444.F, 107461.F, + 100903.F, 94746.3F, 88964.9F, 83536.2F, + 78438.8F, 73652.5F, 69158.2F, 64938.1F, + 60975.6F, 57254.9F, 53761.2F, 50480.6F, + 47400.3F, 44507.9F, 41792.0F, 39241.9F, + 36847.3F, 34598.9F, 32487.7F, 30505.3F, + 28643.8F, 26896.0F, 25254.8F, 23713.7F, + 22266.7F, 20908.0F, 19632.2F, 18434.2F, + 17309.4F, 16253.1F, 15261.4F, 14330.1F, + 13455.7F, 12634.6F, 11863.7F, 11139.7F, + 10460.0F, 9821.72F, 9222.39F, 8659.64F, + 8131.23F, 7635.06F, 7169.17F, 6731.70F, + 6320.93F, 5935.23F, 5573.06F, 5232.99F, + 4913.67F, 4613.84F, 4332.30F, 4067.94F, + 3819.72F, 3586.64F, 3367.78F, 3162.28F, + 2969.31F, 2788.13F, 2617.99F, 2458.24F, + 2308.24F, 2167.39F, 2035.14F, 1910.95F, + 1794.35F, 1684.85F, 1582.04F, 1485.51F, + 1394.86F, 1309.75F, 1229.83F, 1154.78F, + 1084.32F, 1018.15F, 956.024F, 897.687F, + 842.910F, 791.475F, 743.179F, 697.830F, + 655.249F, 615.265F, 577.722F, 542.469F, + 509.367F, 478.286F, 449.101F, 421.696F, + 395.964F, 371.803F, 349.115F, 327.812F, + 307.809F, 289.026F, 271.390F, 254.830F, + 239.280F, 224.679F, 210.969F, 198.096F, + 186.008F, 174.658F, 164.000F, 153.993F, + 144.596F, 135.773F, 127.488F, 119.708F, + 112.404F, 105.545F, 99.1046F, 93.0572F, + 87.3788F, 82.0469F, 77.0404F, 72.3394F, + 67.9252F, 63.7804F, 59.8885F, 56.2341F, + 52.8027F, 49.5807F, 46.5553F, 43.7144F, + 41.0470F, 38.5423F, 36.1904F, 33.9821F, + 31.9085F, 29.9614F, 28.1332F, 26.4165F, + 24.8045F, 23.2910F, 21.8697F, 20.5352F, + 19.2822F, 18.1056F, 17.0008F, 15.9634F, + 14.9893F, 14.0746F, 13.2158F, 12.4094F, + 11.6522F, 10.9411F, 10.2735F, 9.64662F, + 9.05798F, 8.50526F, 7.98626F, 7.49894F, + 7.04135F, 6.61169F, 6.20824F, 5.82941F, + 5.47370F, 5.13970F, 4.82607F, 4.53158F, + 4.25507F, 3.99542F, 3.75162F, 3.52269F, + 3.30774F, 3.10590F, 2.91638F, 2.73842F, + 2.57132F, 2.41442F, 2.26709F, 2.12875F, + 1.99885F, 1.87688F, 1.76236F, 1.65482F, + 1.55384F, 1.45902F, 1.36999F, 1.28640F, + 1.20790F, 1.13419F, 1.06499F, 1.F +}; + +void _vp_remove_floor(vorbis_look_psy *p, + float *mdct, + int *codedflr, + float *residue, + int sliding_lowpass){ + + int i,n=p->n; + + if(sliding_lowpass>n)sliding_lowpass=n; + + for(i=0;in; + float *work=alloca(n*sizeof(*work)); + + bark_noise_hybridmp(n,p->bark,logmdct,logmask, + 140.,-1); + + for(i=0;ibark,work,logmask,0., + p->vi->noisewindowfixed); + + for(i=0;i=NOISE_COMPAND_LEVELS)dB=NOISE_COMPAND_LEVELS-1; + logmask[i]= work[i]+p->vi->noisecompand[dB]; + } + +} + +void _vp_tonemask(vorbis_look_psy *p, + float *logfft, + float *logmask, + float global_specmax, + float local_specmax){ + + int i,n=p->n; + + float *seed=alloca(sizeof(*seed)*p->total_octave_lines); + float att=local_specmax+p->vi->ath_adjatt; + for(i=0;itotal_octave_lines;i++)seed[i]=NEGINF; + + /* set the ATH (floating below localmax, not global max by a + specified att) */ + if(attvi->ath_maxatt)att=p->vi->ath_maxatt; + + for(i=0;iath[i]+att; + + /* tone masking */ + seed_loop(p,(const float ***)p->tonecurves,logfft,logmask,seed,global_specmax); + max_seeds(p,seed,logmask); + +} + +void _vp_offset_and_mix(vorbis_look_psy *p, + float *noise, + float *tone, + int offset_select, + float *logmask){ + int i,n=p->n; + float toneatt=p->vi->tone_masteratt[offset_select]; + + for(i=0;inoiseoffset[offset_select][i]; + if(val>p->vi->noisemaxsupp)val=p->vi->noisemaxsupp; + logmask[i]=max(val,tone[i]+toneatt); + } +} + +float _vp_ampmax_decay(float amp,vorbis_dsp_state *vd){ + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *gi=&ci->psy_g_param; + + int n=ci->blocksizes[vd->W]/2; + float secs=(float)n/vi->rate; + + amp+=secs*gi->ampmax_att_per_sec; + if(amp<-9999)amp=-9999; + return(amp); +} + +static void couple_lossless(float A, float B, + float *qA, float *qB){ + int test1=fabs(*qA)>fabs(*qB); + test1-= fabs(*qA)fabs(B))<<1)-1; + if(test1==1){ + *qB=(*qA>0.f?*qA-*qB:*qB-*qA); + }else{ + float temp=*qB; + *qB=(*qB>0.f?*qA-*qB:*qB-*qA); + *qA=temp; + } + + if(*qB>fabs(*qA)*1.9999f){ + *qB= -fabs(*qA)*2.f; + *qA= -*qA; + } +} + +static float hypot_lookup[32]={ + -0.009935, -0.011245, -0.012726, -0.014397, + -0.016282, -0.018407, -0.020800, -0.023494, + -0.026522, -0.029923, -0.033737, -0.038010, + -0.042787, -0.048121, -0.054064, -0.060671, + -0.068000, -0.076109, -0.085054, -0.094892, + -0.105675, -0.117451, -0.130260, -0.144134, + -0.159093, -0.175146, -0.192286, -0.210490, + -0.229718, -0.249913, -0.271001, -0.292893}; + +static void precomputed_couple_point(float premag, + int floorA,int floorB, + float *mag, float *ang){ + + int test=(floorA>floorB)-1; + int offset=31-abs(floorA-floorB); + float floormag=hypot_lookup[((offset<0)-1)&offset]+1.f; + + floormag*=FLOOR1_fromdB_INV_LOOKUP[(floorB&test)|(floorA&(~test))]; + + *mag=premag*floormag; + *ang=0.f; +} + +/* just like below, this is currently set up to only do + single-step-depth coupling. Otherwise, we'd have to do more + copying (which will be inevitable later) */ + +/* doing the real circular magnitude calculation is audibly superior + to (A+B)/sqrt(2) */ +static float dipole_hypot(float a, float b){ + if(a>0.){ + if(b>0.)return sqrt(a*a+b*b); + if(a>-b)return sqrt(a*a-b*b); + return -sqrt(b*b-a*a); + } + if(b<0.)return -sqrt(a*a+b*b); + if(-a>b)return -sqrt(a*a-b*b); + return sqrt(b*b-a*a); +} +static float round_hypot(float a, float b){ + if(a>0.){ + if(b>0.)return sqrt(a*a+b*b); + if(a>-b)return sqrt(a*a+b*b); + return -sqrt(b*b+a*a); + } + if(b<0.)return -sqrt(a*a+b*b); + if(-a>b)return -sqrt(a*a+b*b); + return sqrt(b*b+a*a); +} + +/* revert to round hypot for now */ +float **_vp_quantize_couple_memo(vorbis_block *vb, + vorbis_info_psy_global *g, + vorbis_look_psy *p, + vorbis_info_mapping0 *vi, + float **mdct){ + + int i,j,n=p->n; + float **ret=_vorbis_block_alloc(vb,vi->coupling_steps*sizeof(*ret)); + int limit=g->coupling_pointlimit[p->vi->blockflag][PACKETBLOBS/2]; + + for(i=0;icoupling_steps;i++){ + float *mdctM=mdct[vi->coupling_mag[i]]; + float *mdctA=mdct[vi->coupling_ang[i]]; + ret[i]=_vorbis_block_alloc(vb,n*sizeof(**ret)); + for(j=0;jfabs(**(float **)b))return -1; + return 1; +} + +int **_vp_quantize_couple_sort(vorbis_block *vb, + vorbis_look_psy *p, + vorbis_info_mapping0 *vi, + float **mags){ + + + if(p->vi->normal_point_p){ + int i,j,k,n=p->n; + int **ret=_vorbis_block_alloc(vb,vi->coupling_steps*sizeof(*ret)); + int partition=p->vi->normal_partition; + float **work=alloca(sizeof(*work)*partition); + + for(i=0;icoupling_steps;i++){ + ret[i]=_vorbis_block_alloc(vb,n*sizeof(**ret)); + + for(j=0;jn; + vorbis_info_psy *vi=p->vi; + int partition=vi->normal_partition; + float **work=alloca(sizeof(*work)*partition); + int start=vi->normal_start; + + for(j=start;jn)partition=n-j; + for(i=0;in; + vorbis_info_psy *vi=p->vi; + int partition=vi->normal_partition; + int start=vi->normal_start; + + if(start>n)start=n; + + if(vi->normal_channel_p){ + for(;j=.25f){ + out[k]=rint(in[k]); + acc-=in[k]*in[k]; + flag=1; + }else{ + if(accnormal_thresh)break; + out[k]=unitnorm(in[k]); + acc-=1.; + } + } + + for(;in; + + /* perform any requested channel coupling */ + /* point stereo can only be used in a first stage (in this encoder) + because of the dependency on floor lookups */ + for(i=0;icoupling_steps;i++){ + + /* once we're doing multistage coupling in which a channel goes + through more than one coupling step, the floor vector + magnitudes will also have to be recalculated an propogated + along with PCM. Right now, we're not (that will wait until 5.1 + most likely), so the code isn't here yet. The memory management + here is all assuming single depth couplings anyway. */ + + /* make sure coupling a zero and a nonzero channel results in two + nonzero channels. */ + if(nonzero[vi->coupling_mag[i]] || + nonzero[vi->coupling_ang[i]]){ + + + float *rM=res[vi->coupling_mag[i]]; + float *rA=res[vi->coupling_ang[i]]; + float *qM=rM+n; + float *qA=rA+n; + int *floorM=ifloor[vi->coupling_mag[i]]; + int *floorA=ifloor[vi->coupling_ang[i]]; + float prepoint=stereo_threshholds[g->coupling_prepointamp[blobno]]; + float postpoint=stereo_threshholds[g->coupling_postpointamp[blobno]]; + int partition=(p->vi->normal_point_p?p->vi->normal_partition:p->n); + int limit=g->coupling_pointlimit[p->vi->blockflag][blobno]; + int pointlimit=limit; + + nonzero[vi->coupling_mag[i]]=1; + nonzero[vi->coupling_ang[i]]=1; + + for(j=0;jn;j+=partition){ + float acc=0.f; + + for(k=0;k=limit && fabs(rM[l])vi->normal_point_p){ + for(k=0;k=p->vi->normal_thresh;k++){ + int l=mag_sort[i][j+k]; + if(l=pointlimit && rint(qM[l])==0.f){ + qM[l]=unitnorm(qM[l]); + acc-=1.f; + } + } + } + } + } + } +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/psy.h b/src/add-ons/media/plugins/vorbis/libvorbis/psy.h new file mode 100644 index 0000000000..4fd953f7a9 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/psy.h @@ -0,0 +1,175 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: random psychoacoustics (not including preecho) + last mod: $Id: psy.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_PSY_H_ +#define _V_PSY_H_ +#include "smallft.h" + +#include "backends.h" +#include "envelope.h" + +#ifndef EHMER_MAX +#define EHMER_MAX 56 +#endif + +/* psychoacoustic setup ********************************************/ +#define P_BANDS 17 /* 62Hz to 16kHz */ +#define P_LEVELS 8 /* 30dB to 100dB */ +#define P_LEVEL_0 30. /* 30 dB */ +#define P_NOISECURVES 3 + +#define NOISE_COMPAND_LEVELS 40 +typedef struct vorbis_info_psy{ + int blockflag; + + float ath_adjatt; + float ath_maxatt; + + float tone_masteratt[P_NOISECURVES]; + float tone_centerboost; + float tone_decay; + float tone_abs_limit; + float toneatt[P_BANDS]; + + int noisemaskp; + float noisemaxsupp; + float noisewindowlo; + float noisewindowhi; + int noisewindowlomin; + int noisewindowhimin; + int noisewindowfixed; + float noiseoff[P_NOISECURVES][P_BANDS]; + float noisecompand[NOISE_COMPAND_LEVELS]; + + float max_curve_dB; + + int normal_channel_p; + int normal_point_p; + int normal_start; + int normal_partition; + double normal_thresh; +} vorbis_info_psy; + +typedef struct{ + int eighth_octave_lines; + + /* for block long/short tuning; encode only */ + float preecho_thresh[VE_BANDS]; + float postecho_thresh[VE_BANDS]; + float stretch_penalty; + float preecho_minenergy; + + float ampmax_att_per_sec; + + /* channel coupling config */ + int coupling_pkHz[PACKETBLOBS]; + int coupling_pointlimit[2][PACKETBLOBS]; + int coupling_prepointamp[PACKETBLOBS]; + int coupling_postpointamp[PACKETBLOBS]; + int sliding_lowpass[2][PACKETBLOBS]; + +} vorbis_info_psy_global; + +typedef struct { + float ampmax; + int channels; + + vorbis_info_psy_global *gi; + int coupling_pointlimit[2][P_NOISECURVES]; +} vorbis_look_psy_global; + + +typedef struct { + int n; + struct vorbis_info_psy *vi; + + float ***tonecurves; + float **noiseoffset; + + float *ath; + long *octave; /* in n.ocshift format */ + long *bark; + + long firstoc; + long shiftoc; + int eighth_octave_lines; /* power of two, please */ + int total_octave_lines; + long rate; /* cache it */ +} vorbis_look_psy; + +extern void _vp_psy_init(vorbis_look_psy *p,vorbis_info_psy *vi, + vorbis_info_psy_global *gi,int n,long rate); +extern void _vp_psy_clear(vorbis_look_psy *p); +extern void *_vi_psy_dup(void *source); + +extern void _vi_psy_free(vorbis_info_psy *i); +extern vorbis_info_psy *_vi_psy_copy(vorbis_info_psy *i); + +extern void _vp_remove_floor(vorbis_look_psy *p, + float *mdct, + int *icodedflr, + float *residue, + int sliding_lowpass); + +extern void _vp_noisemask(vorbis_look_psy *p, + float *logmdct, + float *logmask); + +extern void _vp_tonemask(vorbis_look_psy *p, + float *logfft, + float *logmask, + float global_specmax, + float local_specmax); + +extern void _vp_offset_and_mix(vorbis_look_psy *p, + float *noise, + float *tone, + int offset_select, + float *logmask); + +extern float _vp_ampmax_decay(float amp,vorbis_dsp_state *vd); + +extern float **_vp_quantize_couple_memo(vorbis_block *vb, + vorbis_info_psy_global *g, + vorbis_look_psy *p, + vorbis_info_mapping0 *vi, + float **mdct); + +extern void _vp_couple(int blobno, + vorbis_info_psy_global *g, + vorbis_look_psy *p, + vorbis_info_mapping0 *vi, + float **res, + float **mag_memo, + int **mag_sort, + int **ifloor, + int *nonzero, + int sliding_lowpass); + +extern void _vp_noise_normalize(vorbis_look_psy *p, + float *in,float *out,int *sortedindex); + +extern void _vp_noise_normalize_sort(vorbis_look_psy *p, + float *magnitudes,int *sortedindex); + +extern int **_vp_quantize_couple_sort(vorbis_block *vb, + vorbis_look_psy *p, + vorbis_info_mapping0 *vi, + float **mags); + +#endif + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/psytune.c b/src/add-ons/media/plugins/vorbis/libvorbis/psytune.c new file mode 100644 index 0000000000..ffcbf656c9 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/psytune.c @@ -0,0 +1,521 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: simple utility that runs audio through the psychoacoustics + without encoding + last mod: $Id: psytune.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include + +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "os.h" +#include "misc.h" +#include "psy.h" +#include "mdct.h" +#include "smallft.h" +#include "window.h" +#include "scales.h" +#include "lpc.h" +#include "lsp.h" +#include "masking.h" +#include "registry.h" + +static vorbis_info_psy_global _psy_set0G={ + 0, /* decaydBpms */ + 8, /* lines per eighth octave */ + + /* thresh sample period, preecho clamp trigger threshhold, range, minenergy */ + 256, {26.f,26.f,26.f,30.f}, {-90.f,-90.f,-90.f,-90.f}, -90.f, + -6.f, + + 0, + + 0., + 0., +}; + +static vp_part _vp_part0[]={ + { 1,9e10f, 9e10f, 1.f,9999.f}, + { 9999, .75f, 9e10f, .5f,9999.f}, +/*{ 9999, 1.5f, 9e10f, .5f,9999.f},*/ + { 18,9e10f, 9e10f, .5f, 30.f}, + { 9999,9e10f, 9e10f, .5f, 30.f} +}; + +static vp_couple _vp_couple0[]={ + { 1, {9e10f,9e10f,0}, { 0.f, 0.f,0}, { 0.f, 0.f,0}, {0.f,0.f,0}}, + { 18, {9e10f,9e10f,0}, { 0.f, 0.f,0}, { 0.f, 0.f,0}, {0.f,0.f,0}}, + { 9999, {9e10f,9e10f,0}, { 0.f, 9e10f,0}, { 0.f,22.f,1}, {0.f,0.f,0}} +}; + +static vorbis_info_psy _psy_set0={ + ATH_Bark_dB_lineaggressive, + + -100.f, + -140.f, + 6.f, /* floor master att */ + + /* 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 */ + /* x: 63 88 125 175 250 350 500 700 1k 1.4k 2k 2.8k 4k 5.6k 8k 11.5k 16k Hz */ + /* y: 0 10 20 30 40 50 60 70 80 90 100 dB */ + 1, /* tonemaskp */ + 0.f, /* tone master att */ + /* 0 10 20 30 40 50 60 70 80 90 100 */ + { + {-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f}, /*63*/ + {-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f}, /*88*/ + {-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f,-999.f}, /*125*/ + + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*175*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*250*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*350*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*500*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*700*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*1000*/ + {-30.f,-30.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*1400*/ + {-40.f,-40.f,-40.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*2000*/ + {-40.f,-40.f,-40.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*2800*/ + {-40.f,-40.f,-40.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*4000*/ + + {-30.f,-35.f,-35.f,-40.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*5600*/ + + {-30.f,-30.f,-33.f,-35.f,-40.f,-50.f,-60.f,-70.f,-80.f,-90.f,-100.f}, /*8000*/ + {-30.f,-30.f,-33.f,-35.f,-40.f,-45.f,-50.f,-60.f,-70.f,-85.f,-100.f}, /*11500*/ + {-24.f,-24.f,-26.f,-32.f,-32.f,-42.f,-50.f,-60.f,-70.f,-85.f,-100.f}, /*16000*/ + + }, + + 1,/* peakattp */ + {{-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*63*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*88*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*125*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*175*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*250*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*350*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*500*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*700*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*1000*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*1400*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*2000*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*2800*/ + {-14.f,-20.f,-20.f,-20.f,-26.f,-32.f,-40.f,-40.f,-40.f,-40.f,-40.f},/*4000*/ + {-10.f,-12.f,-14.f,-16.f,-16.f,-20.f,-24.f,-30.f,-32.f,-40.f,-40.f},/*5600*/ + {-10.f,-12.f,-14.f,-16.f,-16.f,-20.f,-24.f,-30.f,-32.f,-40.f,-40.f},/*8000*/ + {-10.f,-10.f,-10.f,-12.f,-14.f,-18.f,-22.f,-28.f,-32.f,-40.f,-40.f},/*11500*/ + {-10.f,-10.f,-10.f,-12.f,-14.f,-18.f,-22.f,-28.f,-32.f,-40.f,-40.f},/*16000*/ + }, + + 1,/*noisemaskp */ + -10.f, /* suppress any noise curve over maxspec+n */ + .5f, /* low window */ + .5f, /* high window */ + 10, + 10, + 25, + {.000f, 0.f, /*63*/ + .000f, 0.f, /*88*/ + .000f, 0.f, /*125*/ + .000f, 0.f, /*175*/ + .000f, 0.f, /*250*/ + .000f, 0.f, /*350*/ + .000f, 0.f, /*500*/ + .000f, 0.f, /*700*/ + .000f, 0.f, /*1000*/ + .300f, 0.f, /*1400*/ + .300f, 0.f, /*2000*/ + .300f, 0.f, /*2800*/ + .500f, 0.f, /*4000*/ + .700f, 0.f, /*5600*/ + .850f, 0.f, /*8000*/ + .900f, 0.f, /*11500*/ + .900f, 1.f, /*16000*/ + }, + + 95.f, /* even decade + 5 is important; saves an rint() later in a + tight loop) */ + -44., + + 32, + _vp_part0,_vp_couple0 +}; + +static vorbis_info_floor1 _floor_set0={1, + {0}, + + {32}, + {0}, + {0}, + {{-1}}, + + 2, + {0,1024, + + 88,31,243, + + 14,54,143,460, + + 6,3,10, 22,18,26, 41,36,47, + 69,61,78, 112,99,126, 185,162,211, + 329,282,387, 672,553,825 + }, + + 60,30,400, + 20,8,1,18., + 20,600, + 960}; + + +static vorbis_info_mapping0 mapping_info={1,{0,1},{0},{0},{0},0, 1, {0},{1}}; +static codec_setup_info codec_setup0={ {0,0}, + 1,1,1,1,1,0,1, + {NULL}, + {0},{&mapping_info}, + {0},{NULL}, + {1},{&_floor_set0}, + {2},{NULL}, + {NULL}, + {&_psy_set0}, + &_psy_set0G}; + +static int noisy=0; +void analysis(char *base,int i,float *v,int n,int bark,int dB){ + if(noisy){ + int j; + FILE *of; + char buffer[80]; + sprintf(buffer,"%s_%d.m",base,i); + of=fopen(buffer,"w"); + + for(j=0;jlook(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>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;jforward(&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;j1500) + 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;j0) + 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;j32767){ + 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 +#include +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "registry.h" +#include "codebook.h" +#include "misc.h" +#include "os.h" + +#ifdef TRAIN_RES +#include +#endif + +typedef struct { + vorbis_info_residue0 *info; + + int parts; + int stages; + codebook *fullbooks; + codebook *phrasebook; + codebook ***partbooks; + + int partvals; + int **decodemap; + + long postbits; + long phrasebits; + long frames; + +#ifdef TRAIN_RES + int train_seq; + long *training_data[8][64]; + float training_max[8][64]; + float training_min[8][64]; + float tmin; + float tmax; +#endif + +} vorbis_look_residue0; + +void res0_free_info(vorbis_info_residue *i){ + vorbis_info_residue0 *info=(vorbis_info_residue0 *)i; + if(info){ + memset(info,0,sizeof(*info)); + _ogg_free(info); + } +} + +void res0_free_look(vorbis_look_residue *i){ + int j; + if(i){ + + vorbis_look_residue0 *look=(vorbis_look_residue0 *)i; + +#ifdef TRAIN_RES + { + int j,k,l; + for(j=0;jparts;j++){ + /*fprintf(stderr,"partition %d: ",j);*/ + for(k=0;k<8;k++) + if(look->training_data[k][j]){ + char buffer[80]; + FILE *of; + codebook *statebook=look->partbooks[j][k]; + + /* long and short into the same bucket by current convention */ + sprintf(buffer,"res_part%d_pass%d.vqd",j,k); + of=fopen(buffer,"a"); + + for(l=0;lentries;l++) + fprintf(of,"%d:%ld\n",l,look->training_data[k][j][l]); + + fclose(of); + + /*fprintf(stderr,"%d(%.2f|%.2f) ",k, + look->training_min[k][j],look->training_max[k][j]);*/ + + _ogg_free(look->training_data[k][j]); + } + /*fprintf(stderr,"\n");*/ + } + } + fprintf(stderr,"min/max residue: %g::%g\n",look->tmin,look->tmax); + + /*fprintf(stderr,"residue bit usage %f:%f (%f total)\n", + (float)look->phrasebits/look->frames, + (float)look->postbits/look->frames, + (float)(look->postbits+look->phrasebits)/look->frames);*/ +#endif + + + /*vorbis_info_residue0 *info=look->info; + + fprintf(stderr, + "%ld frames encoded in %ld phrasebits and %ld residue bits " + "(%g/frame) \n",look->frames,look->phrasebits, + look->resbitsflat, + (look->phrasebits+look->resbitsflat)/(float)look->frames); + + for(j=0;jparts;j++){ + long acc=0; + fprintf(stderr,"\t[%d] == ",j); + for(k=0;kstages;k++) + if((info->secondstages[j]>>k)&1){ + fprintf(stderr,"%ld,",look->resbits[j][k]); + acc+=look->resbits[j][k]; + } + + fprintf(stderr,":: (%ld vals) %1.2fbits/sample\n",look->resvals[j], + acc?(float)acc/(look->resvals[j]*info->grouping):0); + } + fprintf(stderr,"\n");*/ + + for(j=0;jparts;j++) + if(look->partbooks[j])_ogg_free(look->partbooks[j]); + _ogg_free(look->partbooks); + for(j=0;jpartvals;j++) + _ogg_free(look->decodemap[j]); + _ogg_free(look->decodemap); + + memset(look,0,sizeof(*look)); + _ogg_free(look); + } +} + +static int ilog(unsigned int v){ + int ret=0; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +static int icount(unsigned int v){ + int ret=0; + while(v){ + ret+=v&1; + v>>=1; + } + return(ret); +} + + +void res0_pack(vorbis_info_residue *vr,oggpack_buffer *opb){ + vorbis_info_residue0 *info=(vorbis_info_residue0 *)vr; + int j,acc=0; + oggpack_write(opb,info->begin,24); + oggpack_write(opb,info->end,24); + + oggpack_write(opb,info->grouping-1,24); /* residue vectors to group and + code with a partitioned book */ + oggpack_write(opb,info->partitions-1,6); /* possible partition choices */ + oggpack_write(opb,info->groupbook,8); /* group huffman book */ + + /* secondstages is a bitmask; as encoding progresses pass by pass, a + bitmask of one indicates this partition class has bits to write + this pass */ + for(j=0;jpartitions;j++){ + if(ilog(info->secondstages[j])>3){ + /* yes, this is a minor hack due to not thinking ahead */ + oggpack_write(opb,info->secondstages[j],3); + oggpack_write(opb,1,1); + oggpack_write(opb,info->secondstages[j]>>3,5); + }else + oggpack_write(opb,info->secondstages[j],4); /* trailing zero */ + acc+=icount(info->secondstages[j]); + } + for(j=0;jbooklist[j],8); + +} + +/* vorbis_info is for range checking */ +vorbis_info_residue *res0_unpack(vorbis_info *vi,oggpack_buffer *opb){ + int j,acc=0; + vorbis_info_residue0 *info=_ogg_calloc(1,sizeof(*info)); + codec_setup_info *ci=vi->codec_setup; + + info->begin=oggpack_read(opb,24); + info->end=oggpack_read(opb,24); + info->grouping=oggpack_read(opb,24)+1; + info->partitions=oggpack_read(opb,6)+1; + info->groupbook=oggpack_read(opb,8); + + for(j=0;jpartitions;j++){ + int cascade=oggpack_read(opb,3); + if(oggpack_read(opb,1)) + cascade|=(oggpack_read(opb,5)<<3); + info->secondstages[j]=cascade; + + acc+=icount(cascade); + } + for(j=0;jbooklist[j]=oggpack_read(opb,8); + + if(info->groupbook>=ci->books)goto errout; + for(j=0;jbooklist[j]>=ci->books)goto errout; + + return(info); + errout: + res0_free_info(info); + return(NULL); +} + +vorbis_look_residue *res0_look(vorbis_dsp_state *vd, + vorbis_info_residue *vr){ + vorbis_info_residue0 *info=(vorbis_info_residue0 *)vr; + vorbis_look_residue0 *look=_ogg_calloc(1,sizeof(*look)); + codec_setup_info *ci=vd->vi->codec_setup; + + int j,k,acc=0; + int dim; + int maxstage=0; + look->info=info; + + look->parts=info->partitions; + look->fullbooks=ci->fullbooks; + look->phrasebook=ci->fullbooks+info->groupbook; + dim=look->phrasebook->dim; + + look->partbooks=_ogg_calloc(look->parts,sizeof(*look->partbooks)); + + for(j=0;jparts;j++){ + int stages=ilog(info->secondstages[j]); + if(stages){ + if(stages>maxstage)maxstage=stages; + look->partbooks[j]=_ogg_calloc(stages,sizeof(*look->partbooks[j])); + for(k=0;ksecondstages[j]&(1<partbooks[j][k]=ci->fullbooks+info->booklist[acc++]; +#ifdef TRAIN_RES + look->training_data[k][j]=calloc(look->partbooks[j][k]->entries, + sizeof(***look->training_data)); +#endif + } + } + } + + look->partvals=rint(pow((float)look->parts,(float)dim)); + look->stages=maxstage; + look->decodemap=_ogg_malloc(look->partvals*sizeof(*look->decodemap)); + for(j=0;jpartvals;j++){ + long val=j; + long mult=look->partvals/look->parts; + look->decodemap[j]=_ogg_malloc(dim*sizeof(*look->decodemap[j])); + for(k=0;kparts; + look->decodemap[j][k]=deco; + } + } +#ifdef TRAIN_RES + { + static int train_seq=0; + look->train_seq=train_seq++; + } +#endif + return(look); +} + +/* break an abstraction and copy some code for performance purposes */ +static int local_book_besterror(codebook *book,float *a){ + int dim=book->dim,i,k,o; + int best=0; + encode_aux_threshmatch *tt=book->c->thresh_tree; + + /* find the quant val of each scalar */ + for(k=0,o=dim;kthreshvals>>1; + + if(valquantthresh[i]){ + if(valquantthresh[i-1]){ + for(--i;i>0;--i) + if(val>=tt->quantthresh[i-1]) + break; + } + }else{ + + for(++i;ithreshvals-1;++i) + if(valquantthresh[i])break; + + } + + best=(best*tt->quantvals)+tt->quantmap[i]; + } + /* regular lattices are easy :-) */ + + if(book->c->lengthlist[best]<=0){ + const static_codebook *c=book->c; + int i,j; + float bestf=0.f; + float *e=book->valuelist; + best=-1; + for(i=0;ientries;i++){ + if(c->lengthlist[i]>0){ + float this=0.f; + for(j=0;jvaluelist+best*dim; + for(i=0;idim; + int step=n/dim; + + for(i=0;iinfo; + vorbis_info *vi=vb->vd->vi; + codec_setup_info *ci=vi->codec_setup; + + /* move all this setup out later */ + int samples_per_partition=info->grouping; + int possible_partitions=info->partitions; + int n=info->end-info->begin; + + int partvals=n/samples_per_partition; + long **partword=_vorbis_block_alloc(vb,ch*sizeof(*partword)); + float scale=100./samples_per_partition; + + /* we find the partition type for each partition of each + channel. We'll go back and do the interleaved encoding in a + bit. For now, clarity */ + + for(i=0;ibegin; + for(j=0;jmax)max=fabs(in[j][offset+k]); + ent+=fabs(rint(in[j][offset+k])); + } + ent*=scale; + + for(k=0;kclassmetric1[k] && + (info->classmetric2[k]<0 || (int)entclassmetric2[k])) + break; + + partword[j][i]=k; + } + } + +#ifdef TRAIN_RESAUX + { + FILE *of; + char buffer[80]; + + for(i=0;itrain_seq); + of=fopen(buffer,"a"); + for(j=0;jframes++; + + return(partword); +} + +/* designed for stereo or other modes where the partition size is an + integer multiple of the number of channels encoded in the current + submap */ +static long **_2class(vorbis_block *vb,vorbis_look_residue *vl,float **in, + int ch){ + long i,j,k,l; + vorbis_look_residue0 *look=(vorbis_look_residue0 *)vl; + vorbis_info_residue0 *info=look->info; + + /* move all this setup out later */ + int samples_per_partition=info->grouping; + int possible_partitions=info->partitions; + int n=info->end-info->begin; + + int partvals=n/samples_per_partition; + long **partword=_vorbis_block_alloc(vb,sizeof(*partword)); + +#ifdef TRAIN_RES + FILE *of; + char buffer[80]; +#endif + + partword[0]=_vorbis_block_alloc(vb,n*ch/samples_per_partition*sizeof(*partword[0])); + memset(partword[0],0,n*ch/samples_per_partition*sizeof(*partword[0])); + + for(i=0,l=info->begin/ch;imagmax)magmax=fabs(in[0][l]); + for(k=1;kangmax)angmax=fabs(in[k][l]); + l++; + } + + for(j=0;jclassmetric1[j] && + angmax<=info->classmetric2[j]) + break; + + partword[0][i]=j; + + } + +#ifdef TRAIN_RESAUX + sprintf(buffer,"resaux_%d.vqd",look->train_seq); + of=fopen(buffer,"a"); + for(i=0;iframes++; + + return(partword); +} + +static int _01forward(vorbis_block *vb,vorbis_look_residue *vl, + float **in,int ch, + long **partword, + int (*encode)(oggpack_buffer *,float *,int, + codebook *,long *)){ + long i,j,k,s; + vorbis_look_residue0 *look=(vorbis_look_residue0 *)vl; + vorbis_info_residue0 *info=look->info; + + vorbis_dsp_state *vd=vb->vd; + + /* move all this setup out later */ + int samples_per_partition=info->grouping; + int possible_partitions=info->partitions; + int partitions_per_word=look->phrasebook->dim; + int n=info->end-info->begin; + + int partvals=n/samples_per_partition; + long resbits[128]; + long resvals[128]; + +#ifdef TRAIN_RES + for(i=0;ibegin;jend;j++){ + if(in[i][j]>look->tmax)look->tmax=in[i][j]; + if(in[i][j]tmin)look->tmin=in[i][j]; + } +#endif + + memset(resbits,0,sizeof(resbits)); + memset(resvals,0,sizeof(resvals)); + + /* we code the partition words for each channel, then the residual + words for a partition per channel until we've written all the + residual words for that partition word. Then write the next + partition channel words... */ + + for(s=0;sstages;s++){ + + for(i=0;iphrasebook->entries) + look->phrasebits+=vorbis_book_encode(look->phrasebook,val,&vb->opb); +#if 0 /*def TRAIN_RES*/ + else + fprintf(stderr,"!"); +#endif + + } + } + + /* now we encode interleaved residual values for the partitions */ + for(k=0;kbegin; + + for(j=0;jsecondstages[partword[j][i]]&(1<partbooks[partword[j][i]][s]; + if(statebook){ + int ret; + long *accumulator=NULL; + +#ifdef TRAIN_RES + accumulator=look->training_data[s][partword[j][i]]; + { + int l; + float *samples=in[j]+offset; + for(l=0;ltraining_min[s][partword[j][i]]) + look->training_min[s][partword[j][i]]=samples[l]; + if(samples[l]>look->training_max[s][partword[j][i]]) + look->training_max[s][partword[j][i]]=samples[l]; + } + } +#endif + + ret=encode(&vb->opb,in[j]+offset,samples_per_partition, + statebook,accumulator); + + look->postbits+=ret; + resbits[partword[j][i]]+=ret; + } + } + } + } + } + } + + /*{ + long total=0; + long totalbits=0; + fprintf(stderr,"%d :: ",vb->mode); + for(k=0;kinfo; + + /* move all this setup out later */ + int samples_per_partition=info->grouping; + int partitions_per_word=look->phrasebook->dim; + int n=info->end-info->begin; + + int partvals=n/samples_per_partition; + int partwords=(partvals+partitions_per_word-1)/partitions_per_word; + int ***partword=alloca(ch*sizeof(*partword)); + + for(j=0;jstages;s++){ + + /* each loop decodes on partition codeword containing + partitions_pre_word partitions */ + for(i=0,l=0;iphrasebook,&vb->opb); + if(temp==-1)goto eopbreak; + partword[j][l]=look->decodemap[temp]; + if(partword[j][l]==NULL)goto errout; + } + } + + /* now we decode residual values for the partitions */ + for(k=0;kbegin+i*samples_per_partition; + if(info->secondstages[partword[j][l][k]]&(1<partbooks[partword[j][l][k]][s]; + if(stagebook){ + if(decodepart(stagebook,in[j]+offset,&vb->opb, + samples_per_partition)==-1)goto eopbreak; + } + } + } + } + } + + errout: + eopbreak: + return(0); +} + +#if 0 +/* residue 0 and 1 are just slight variants of one another. 0 is + interleaved, 1 is not */ +long **res0_class(vorbis_block *vb,vorbis_look_residue *vl, + float **in,int *nonzero,int ch){ + /* we encode only the nonzero parts of a bundle */ + int i,used=0; + for(i=0;ipcmend/2; + for(i=0;ipcmend/2; + for(i=0;ipcmend/2,used=0; + + /* don't duplicate the code; use a working vector hack for now and + reshape ourselves into a single channel res1 */ + /* ugly; reallocs for each coupling pass :-( */ + float *work=_vorbis_block_alloc(vb,ch*n*sizeof(*work)); + for(i=0;iinfo; + + /* move all this setup out later */ + int samples_per_partition=info->grouping; + int partitions_per_word=look->phrasebook->dim; + int n=info->end-info->begin; + + int partvals=n/samples_per_partition; + int partwords=(partvals+partitions_per_word-1)/partitions_per_word; + int **partword=_vorbis_block_alloc(vb,partwords*sizeof(*partword)); + + for(i=0;istages;s++){ + for(i=0,l=0;iphrasebook,&vb->opb); + if(temp==-1)goto eopbreak; + partword[l]=look->decodemap[temp]; + if(partword[l]==NULL)goto errout; + } + + /* now we decode residual values for the partitions */ + for(k=0;ksecondstages[partword[l][k]]&(1<partbooks[partword[l][k]][s]; + + if(stagebook){ + if(vorbis_book_decodevv_add(stagebook,in, + i*samples_per_partition+info->begin,ch, + &vb->opb,samples_per_partition)==-1) + goto eopbreak; + } + } + } + } + + errout: + eopbreak: + return(0); +} + + +vorbis_func_residue residue0_exportbundle={ + NULL, + &res0_unpack, + &res0_look, + &res0_free_info, + &res0_free_look, + NULL, + NULL, + &res0_inverse +}; + +vorbis_func_residue residue1_exportbundle={ + &res0_pack, + &res0_unpack, + &res0_look, + &res0_free_info, + &res0_free_look, + &res1_class, + &res1_forward, + &res1_inverse +}; + +vorbis_func_residue residue2_exportbundle={ + &res0_pack, + &res0_unpack, + &res0_look, + &res0_free_info, + &res0_free_look, + &res2_class, + &res2_forward, + &res2_inverse +}; + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/scales.h b/src/add-ons/media/plugins/vorbis/libvorbis/scales.h new file mode 100644 index 0000000000..6c22e4e188 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/scales.h @@ -0,0 +1,88 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: linear scale -> dB, Bark and Mel scales + last mod: $Id: scales.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_SCALES_H_ +#define _V_SCALES_H_ + +#include +#include "os.h" + +/* 20log10(x) */ +#define VORBIS_IEEE_FLOAT32 1 +#ifdef VORBIS_IEEE_FLOAT32 + +static float unitnorm(float x){ + ogg_uint32_t *ix=(ogg_uint32_t *)&x; + *ix=(*ix&0x80000000UL)|(0x3f800000UL); + return(x); +} + +static float FABS(float *x){ + ogg_uint32_t *ix=(ogg_uint32_t *)x; + *ix&=0x7fffffffUL; + return(*x); +} + +static float todB(const float *x){ + float calc; + ogg_int32_t *i=(ogg_int32_t *)x; + calc = ((*i) & 0x7fffffff); + calc *= 7.1771144e-7f; + calc += -764.27118f; + return calc; +} + +#define todB_nn(x) todB(x) + +#else + +static float unitnorm(float x){ + if(x<0)return(-1.f); + return(1.f); +} + +#define FABS(x) fabs(*(x)) + +#define todB(x) (*(x)==0?-400.f:log(*(x)**(x))*4.34294480f) +#define todB_nn(x) (*(x)==0.f?-400.f:log(*(x))*8.6858896f) + +#endif + +#define fromdB(x) (exp((x)*.11512925f)) + +/* The bark scale equations are approximations, since the original + table was somewhat hand rolled. The below are chosen to have the + best possible fit to the rolled tables, thus their somewhat odd + appearance (these are more accurate and over a longer range than + the oft-quoted bark equations found in the texts I have). The + approximations are valid from 0 - 30kHz (nyquist) or so. + + all f in Hz, z in Bark */ + +#define toBARK(n) (13.1f*atan(.00074f*(n))+2.24f*atan((n)*(n)*1.85e-8f)+1e-4f*(n)) +#define fromBARK(z) (102.f*(z)-2.f*pow(z,2.f)+.4f*pow(z,3.f)+pow(1.46f,z)-1.f) +#define toMEL(n) (log(1.f+(n)*.001f)*1442.695f) +#define fromMEL(m) (1000.f*exp((m)/1442.695f)-1000.f) + +/* Frequency to octave. We arbitrarily declare 63.5 Hz to be octave + 0.0 */ + +#define toOC(n) (log(n)*1.442695f-5.965784f) +#define fromOC(o) (exp(((o)+5.965784f)*.693147f)) + +#endif + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/sharedbook.c b/src/add-ons/media/plugins/vorbis/libvorbis/sharedbook.c new file mode 100644 index 0000000000..7a026a5de6 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/sharedbook.c @@ -0,0 +1,733 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: basic shared codebook operations + last mod: $Id: sharedbook.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include "os.h" +#include "misc.h" +#include "vorbis/codec.h" +#include "codebook.h" +#include "scales.h" + +/**** pack/unpack helpers ******************************************/ +int _ilog(unsigned int v){ + int ret=0; + while(v){ + ret++; + v>>=1; + } + return(ret); +} + +/* 32 bit float (not IEEE; nonnormalized mantissa + + biased exponent) : neeeeeee eeemmmmm mmmmmmmm mmmmmmmm + Why not IEEE? It's just not that important here. */ + +#define VQ_FEXP 10 +#define VQ_FMAN 21 +#define VQ_FEXP_BIAS 768 /* bias toward values smaller than 1. */ + +/* doesn't currently guard under/overflow */ +long _float32_pack(float val){ + int sign=0; + long exp; + long mant; + if(val<0){ + sign=0x80000000; + val= -val; + } + exp= floor(log(val)/log(2.f)); + mant=rint(ldexp(val,(VQ_FMAN-1)-exp)); + exp=(exp+VQ_FEXP_BIAS)<>VQ_FMAN; + if(sign)mant= -mant; + return(ldexp(mant,exp-(VQ_FMAN-1)-VQ_FEXP_BIAS)); +} + +/* given a list of word lengths, generate a list of codewords. Works + for length ordered or unordered, always assigns the lowest valued + codewords first. Extended to handle unused entries (length 0) */ +ogg_uint32_t *_make_words(long *l,long n,long sparsecount){ + long i,j,count=0; + ogg_uint32_t marker[33]; + ogg_uint32_t *r=_ogg_malloc((sparsecount?sparsecount:n)*sizeof(*r)); + memset(marker,0,sizeof(marker)); + + for(i=0;i0){ + ogg_uint32_t entry=marker[length]; + + /* when we claim a node for an entry, we also claim the nodes + below it (pruning off the imagined tree that may have dangled + from it) as well as blocking the use of any nodes directly + above for leaves */ + + /* update ourself */ + if(length<32 && (entry>>length)){ + /* error condition; the lengths must specify an overpopulated tree */ + _ogg_free(r); + return(NULL); + } + r[count++]=entry; + + /* Look to see if the next shorter marker points to the node + above. if so, update it and repeat. */ + { + for(j=length;j>0;j--){ + + if(marker[j]&1){ + /* have to jump branches */ + if(j==1) + marker[1]++; + else + marker[j]=marker[j-1]<<1; + break; /* invariant says next upper marker would already + have been moved if it was on the same path */ + } + marker[j]++; + } + } + + /* prune the tree; the implicit invariant says all the longer + markers were dangling from our just-taken node. Dangle them + from our *new* node. */ + for(j=length+1;j<33;j++) + if((marker[j]>>1) == entry){ + entry=marker[j]; + marker[j]=marker[j-1]<<1; + }else + break; + }else + if(sparsecount==0)count++; + } + + /* bitreverse the words because our bitwise packer/unpacker is LSb + endian */ + for(i=0,count=0;i>j)&1; + } + + if(sparsecount){ + if(l[i]) + r[count++]=temp; + }else + r[count++]=temp; + } + + return(r); +} + +/* there might be a straightforward one-line way to do the below + that's portable and totally safe against roundoff, but I haven't + thought of it. Therefore, we opt on the side of caution */ +long _book_maptype1_quantvals(const static_codebook *b){ + long vals=floor(pow((float)b->entries,1.f/b->dim)); + + /* the above *should* be reliable, but we'll not assume that FP is + ever reliable when bitstream sync is at stake; verify via integer + means that vals really is the greatest value of dim for which + vals^b->bim <= b->entries */ + /* treat the above as an initial guess */ + while(1){ + long acc=1; + long acc1=1; + int i; + for(i=0;idim;i++){ + acc*=vals; + acc1*=vals+1; + } + if(acc<=b->entries && acc1>b->entries){ + return(vals); + }else{ + if(acc>b->entries){ + vals--; + }else{ + vals++; + } + } + } +} + +/* unpack the quantized list of values for encode/decode ***********/ +/* we need to deal with two map types: in map type 1, the values are + generated algorithmically (each column of the vector counts through + the values in the quant vector). in map type 2, all the values came + in in an explicit list. Both value lists must be unpacked */ +float *_book_unquantize(const static_codebook *b,int n,int *sparsemap){ + long j,k,count=0; + if(b->maptype==1 || b->maptype==2){ + int quantvals; + float mindel=_float32_unpack(b->q_min); + float delta=_float32_unpack(b->q_delta); + float *r=_ogg_calloc(n*b->dim,sizeof(*r)); + + /* maptype 1 and 2 both use a quantized value vector, but + different sizes */ + switch(b->maptype){ + case 1: + /* most of the time, entries%dimensions == 0, but we need to be + well defined. We define that the possible vales at each + scalar is values == entries/dim. If entries%dim != 0, we'll + have 'too few' values (values*dimentries;j++){ + if((sparsemap && b->lengthlist[j]) || !sparsemap){ + float last=0.f; + int indexdiv=1; + for(k=0;kdim;k++){ + int index= (j/indexdiv)%quantvals; + float val=b->quantlist[index]; + val=fabs(val)*delta+mindel+last; + if(b->q_sequencep)last=val; + if(sparsemap) + r[sparsemap[count]*b->dim+k]=val; + else + r[count*b->dim+k]=val; + indexdiv*=quantvals; + } + count++; + } + + } + break; + case 2: + for(j=0;jentries;j++){ + if((sparsemap && b->lengthlist[j]) || !sparsemap){ + float last=0.f; + + for(k=0;kdim;k++){ + float val=b->quantlist[j*b->dim+k]; + val=fabs(val)*delta+mindel+last; + if(b->q_sequencep)last=val; + if(sparsemap) + r[sparsemap[count]*b->dim+k]=val; + else + r[count*b->dim+k]=val; + } + count++; + } + } + break; + } + + return(r); + } + return(NULL); +} + +void vorbis_staticbook_clear(static_codebook *b){ + if(b->allocedp){ + if(b->quantlist)_ogg_free(b->quantlist); + if(b->lengthlist)_ogg_free(b->lengthlist); + if(b->nearest_tree){ + _ogg_free(b->nearest_tree->ptr0); + _ogg_free(b->nearest_tree->ptr1); + _ogg_free(b->nearest_tree->p); + _ogg_free(b->nearest_tree->q); + memset(b->nearest_tree,0,sizeof(*b->nearest_tree)); + _ogg_free(b->nearest_tree); + } + if(b->thresh_tree){ + _ogg_free(b->thresh_tree->quantthresh); + _ogg_free(b->thresh_tree->quantmap); + memset(b->thresh_tree,0,sizeof(*b->thresh_tree)); + _ogg_free(b->thresh_tree); + } + + memset(b,0,sizeof(*b)); + } +} + +void vorbis_staticbook_destroy(static_codebook *b){ + if(b->allocedp){ + vorbis_staticbook_clear(b); + _ogg_free(b); + } +} + +void vorbis_book_clear(codebook *b){ + /* static book is not cleared; we're likely called on the lookup and + the static codebook belongs to the info struct */ + if(b->valuelist)_ogg_free(b->valuelist); + if(b->codelist)_ogg_free(b->codelist); + + if(b->dec_index)_ogg_free(b->dec_index); + if(b->dec_codelengths)_ogg_free(b->dec_codelengths); + if(b->dec_firsttable)_ogg_free(b->dec_firsttable); + + memset(b,0,sizeof(*b)); +} + +int vorbis_book_init_encode(codebook *c,const static_codebook *s){ + + memset(c,0,sizeof(*c)); + c->c=s; + c->entries=s->entries; + c->used_entries=s->entries; + c->dim=s->dim; + c->codelist=_make_words(s->lengthlist,s->entries,0); + c->valuelist=_book_unquantize(s,s->entries,NULL); + + return(0); +} + +static ogg_uint32_t bitreverse(ogg_uint32_t x){ + x= ((x>>16)&0x0000ffffUL) | ((x<<16)&0xffff0000UL); + x= ((x>> 8)&0x00ff00ffUL) | ((x<< 8)&0xff00ff00UL); + x= ((x>> 4)&0x0f0f0f0fUL) | ((x<< 4)&0xf0f0f0f0UL); + x= ((x>> 2)&0x33333333UL) | ((x<< 2)&0xccccccccUL); + return((x>> 1)&0x55555555UL) | ((x<< 1)&0xaaaaaaaaUL); +} + +static int sort32a(const void *a,const void *b){ + return ( (**(ogg_uint32_t **)a>**(ogg_uint32_t **)b)<<1)-1; +} + +/* decode codebook arrangement is more heavily optimized than encode */ +int vorbis_book_init_decode(codebook *c,const static_codebook *s){ + int i,j,n=0,tabn; + int *sortindex; + memset(c,0,sizeof(*c)); + + /* count actually used entries */ + for(i=0;ientries;i++) + if(s->lengthlist[i]>0) + n++; + + c->entries=s->entries; + c->used_entries=n; + c->dim=s->dim; + + /* two different remappings go on here. + + First, we collapse the likely sparse codebook down only to + actually represented values/words. This collapsing needs to be + indexed as map-valueless books are used to encode original entry + positions as integers. + + Second, we reorder all vectors, including the entry index above, + by sorted bitreversed codeword to allow treeless decode. */ + + { + /* perform sort */ + ogg_uint32_t *codes=_make_words(s->lengthlist,s->entries,c->used_entries); + ogg_uint32_t **codep=alloca(sizeof(*codep)*n); + + if(codes==NULL)goto err_out; + + for(i=0;icodelist=_ogg_malloc(n*sizeof(*c->codelist)); + /* the index is a reverse index */ + for(i=0;icodelist[sortindex[i]]=codes[i]; + _ogg_free(codes); + } + + c->valuelist=_book_unquantize(s,n,sortindex); + c->dec_index=_ogg_malloc(n*sizeof(*c->dec_index)); + + for(n=0,i=0;ientries;i++) + if(s->lengthlist[i]>0) + c->dec_index[sortindex[n++]]=i; + + c->dec_codelengths=_ogg_malloc(n*sizeof(*c->dec_codelengths)); + for(n=0,i=0;ientries;i++) + if(s->lengthlist[i]>0) + c->dec_codelengths[sortindex[n++]]=s->lengthlist[i]; + + c->dec_firsttablen=_ilog(c->used_entries)-4; /* this is magic */ + if(c->dec_firsttablen<5)c->dec_firsttablen=5; + if(c->dec_firsttablen>8)c->dec_firsttablen=8; + + tabn=1<dec_firsttablen; + c->dec_firsttable=_ogg_calloc(tabn,sizeof(*c->dec_firsttable)); + c->dec_maxlength=0; + + for(i=0;idec_maxlengthdec_codelengths[i]) + c->dec_maxlength=c->dec_codelengths[i]; + if(c->dec_codelengths[i]<=c->dec_firsttablen){ + ogg_uint32_t orig=bitreverse(c->codelist[i]); + for(j=0;j<(1<<(c->dec_firsttablen-c->dec_codelengths[i]));j++) + c->dec_firsttable[orig|(j<dec_codelengths[i])]=i+1; + } + } + + /* now fill in 'unused' entries in the firsttable with hi/lo search + hints for the non-direct-hits */ + { + ogg_uint32_t mask=0xfffffffeUL<<(31-c->dec_firsttablen); + long lo=0,hi=0; + + for(i=0;idec_firsttablen); + if(c->dec_firsttable[bitreverse(word)]==0){ + while((lo+1)codelist[lo+1]<=word)lo++; + while( hi=(c->codelist[hi]&mask))hi++; + + /* we only actually have 15 bits per hint to play with here. + In order to overflow gracefully (nothing breaks, efficiency + just drops), encode as the difference from the extremes. */ + { + unsigned long loval=lo; + unsigned long hival=n-hi; + + if(loval>0x7fff)loval=0x7fff; + if(hival>0x7fff)hival=0x7fff; + c->dec_firsttable[bitreverse(word)]= + 0x80000000UL | (loval<<15) | hival; + } + } + } + } + + + return(0); + err_out: + vorbis_book_clear(c); + return(-1); +} + +static float _dist(int el,float *ref, float *b,int step){ + int i; + float acc=0.f; + for(i=0;ic->thresh_tree; + +#if 0 + encode_aux_nearestmatch *nt=book->c->nearest_tree; + encode_aux_pigeonhole *pt=book->c->pigeon_tree; +#endif + int dim=book->dim; + int k,o; + /*int savebest=-1; + float saverr;*/ + + /* do we have a threshhold encode hint? */ + if(tt){ + int index=0,i; + /* find the quant val of each scalar */ + for(k=0,o=step*(dim-1);kthreshvals>>1; + if(a[o]quantthresh[i]){ + + for(;i>0;i--) + if(a[o]>=tt->quantthresh[i-1]) + break; + + }else{ + + for(i++;ithreshvals-1;i++) + if(a[o]quantthresh[i])break; + + } + + index=(index*tt->quantvals)+tt->quantmap[i]; + } + /* regular lattices are easy :-) */ + if(book->c->lengthlist[index]>0) /* is this unused? If so, we'll + use a decision tree after all + and fall through*/ + return(index); + } + +#if 0 + /* do we have a pigeonhole encode hint? */ + if(pt){ + const static_codebook *c=book->c; + int i,besti=-1; + float best=0.f; + int entry=0; + + /* dealing with sequentialness is a pain in the ass */ + if(c->q_sequencep){ + int pv; + long mul=1; + float qlast=0; + for(k=0,o=0;kmin)/pt->del); + if(pv<0 || pv>=pt->mapentries)break; + entry+=pt->pigeonmap[pv]*mul; + mul*=pt->quantvals; + qlast+=pv*pt->del+pt->min; + } + }else{ + for(k=0,o=step*(dim-1);kmin)/pt->del); + if(pv<0 || pv>=pt->mapentries)break; + entry=entry*pt->quantvals+pt->pigeonmap[pv]; + } + } + + /* must be within the pigeonholable range; if we quant outside (or + in an entry that we define no list for), brute force it */ + if(k==dim && pt->fitlength[entry]){ + /* search the abbreviated list */ + long *list=pt->fitlist+pt->fitmap[entry]; + for(i=0;ifitlength[entry];i++){ + float this=_dist(dim,book->valuelist+list[i]*dim,a,step); + if(besti==-1 || thisvaluelist+nt->p[ptr]; + float *q=book->valuelist+nt->q[ptr]; + + for(k=0,o=0;k0.f) /* in A */ + ptr= -nt->ptr0[ptr]; + else /* in B */ + ptr= -nt->ptr1[ptr]; + if(ptr<=0)break; + } + return(-ptr); + } +#endif + + /* brute force it! */ + { + const static_codebook *c=book->c; + int i,besti=-1; + float best=0.f; + float *e=book->valuelist; + for(i=0;ientries;i++){ + if(c->lengthlist[i]>0){ + float this=_dist(dim,e,a,step); + if(besti==-1 || thisvaluelist+savebest*dim)[i]); + fprintf(stderr,"\n" + "bruteforce (entry %d, err %g):",besti,best); + for(i=0;ivaluelist+besti*dim)[i]); + fprintf(stderr,"\n"); + }*/ + return(besti); + } +} + +long vorbis_book_codeword(codebook *book,int entry){ + if(book->c) /* only use with encode; decode optimizations are + allowed to break this */ + return book->codelist[entry]; + return -1; +} + +long vorbis_book_codelen(codebook *book,int entry){ + if(book->c) /* only use with encode; decode optimizations are + allowed to break this */ + return book->c->lengthlist[entry]; + return -1; +} + +#ifdef _V_SELFTEST + +/* Unit tests of the dequantizer; this stuff will be OK + cross-platform, I simply want to be sure that special mapping cases + actually work properly; a bug could go unnoticed for a while */ + +#include + +/* cases: + + no mapping + full, explicit mapping + algorithmic mapping + + nonsequential + sequential +*/ + +static long full_quantlist1[]={0,1,2,3, 4,5,6,7, 8,3,6,1}; +static long partial_quantlist1[]={0,7,2}; + +/* no mapping */ +static_codebook test1={ + 4,16, + NULL, + 0, + 0,0,0,0, + NULL, + NULL,NULL +}; +static float *test1_result=NULL; + +/* linear, full mapping, nonsequential */ +static_codebook test2={ + 4,3, + NULL, + 2, + -533200896,1611661312,4,0, + full_quantlist1, + NULL,NULL +}; +static float test2_result[]={-3,-2,-1,0, 1,2,3,4, 5,0,3,-2}; + +/* linear, full mapping, sequential */ +static_codebook test3={ + 4,3, + NULL, + 2, + -533200896,1611661312,4,1, + full_quantlist1, + NULL,NULL +}; +static float test3_result[]={-3,-5,-6,-6, 1,3,6,10, 5,5,8,6}; + +/* linear, algorithmic mapping, nonsequential */ +static_codebook test4={ + 3,27, + NULL, + 1, + -533200896,1611661312,4,0, + partial_quantlist1, + NULL,NULL +}; +static float test4_result[]={-3,-3,-3, 4,-3,-3, -1,-3,-3, + -3, 4,-3, 4, 4,-3, -1, 4,-3, + -3,-1,-3, 4,-1,-3, -1,-1,-3, + -3,-3, 4, 4,-3, 4, -1,-3, 4, + -3, 4, 4, 4, 4, 4, -1, 4, 4, + -3,-1, 4, 4,-1, 4, -1,-1, 4, + -3,-3,-1, 4,-3,-1, -1,-3,-1, + -3, 4,-1, 4, 4,-1, -1, 4,-1, + -3,-1,-1, 4,-1,-1, -1,-1,-1}; + +/* linear, algorithmic mapping, sequential */ +static_codebook test5={ + 3,27, + NULL, + 1, + -533200896,1611661312,4,1, + partial_quantlist1, + NULL,NULL +}; +static float test5_result[]={-3,-6,-9, 4, 1,-2, -1,-4,-7, + -3, 1,-2, 4, 8, 5, -1, 3, 0, + -3,-4,-7, 4, 3, 0, -1,-2,-5, + -3,-6,-2, 4, 1, 5, -1,-4, 0, + -3, 1, 5, 4, 8,12, -1, 3, 7, + -3,-4, 0, 4, 3, 7, -1,-2, 2, + -3,-6,-7, 4, 1, 0, -1,-4,-5, + -3, 1, 0, 4, 8, 7, -1, 3, 2, + -3,-4,-5, 4, 3, 2, -1,-2,-3}; + +void run_test(static_codebook *b,float *comp){ + float *out=_book_unquantize(b,b->entries,NULL); + int i; + + if(comp){ + if(!out){ + fprintf(stderr,"_book_unquantize incorrectly returned NULL\n"); + exit(1); + } + + for(i=0;ientries*b->dim;i++) + if(fabs(out[i]-comp[i])>.0001){ + fprintf(stderr,"disagreement in unquantized and reference data:\n" + "position %d, %g != %g\n",i,out[i],comp[i]); + exit(1); + } + + }else{ + if(out){ + fprintf(stderr,"_book_unquantize returned a value array: \n" + " correct result should have been NULL\n"); + exit(1); + } + } +} + +int main(){ + /* run the nine dequant tests, and compare to the hand-rolled results */ + fprintf(stderr,"Dequant test 1... "); + run_test(&test1,test1_result); + fprintf(stderr,"OK\nDequant test 2... "); + run_test(&test2,test2_result); + fprintf(stderr,"OK\nDequant test 3... "); + run_test(&test3,test3_result); + fprintf(stderr,"OK\nDequant test 4... "); + run_test(&test4,test4_result); + fprintf(stderr,"OK\nDequant test 5... "); + run_test(&test5,test5_result); + fprintf(stderr,"OK\n\n"); + + return(0); +} + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/smallft.c b/src/add-ons/media/plugins/vorbis/libvorbis/smallft.c new file mode 100644 index 0000000000..f96dc496ab --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/smallft.c @@ -0,0 +1,1254 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: *unnormalized* fft transform + last mod: $Id: smallft.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +/* FFT implementation from OggSquish, minus cosine transforms, + * minus all but radix 2/4 case. In Vorbis we only need this + * cut-down version. + * + * To do more than just power-of-two sized vectors, see the full + * version I wrote for NetLib. + * + * Note that the packing is a little strange; rather than the FFT r/i + * packing following R_0, I_n, R_1, I_1, R_2, I_2 ... R_n-1, I_n-1, + * it follows R_0, R_1, I_1, R_2, I_2 ... R_n-1, I_n-1, I_n like the + * FORTRAN version + */ + +#include +#include +#include +#include "smallft.h" +#include "misc.h" + +static void drfti1(int n, float *wa, int *ifac){ + static int ntryh[4] = { 4,2,3,5 }; + static float tpi = 6.28318530717958648f; + float arg,argh,argld,fi; + int ntry=0,i,j=-1; + int k1, l1, l2, ib; + int ld, ii, ip, is, nq, nr; + int ido, ipm, nfm1; + int nl=n; + int nf=0; + + L101: + j++; + if (j < 4) + ntry=ntryh[j]; + else + ntry+=2; + + L104: + nq=nl/ntry; + nr=nl-ntry*nq; + if (nr!=0) goto L101; + + nf++; + ifac[nf+1]=ntry; + nl=nq; + if(ntry!=2)goto L107; + if(nf==1)goto L107; + + for (i=1;i>1; + ipp2=ip; + idp2=ido; + nbd=(ido-1)>>1; + t0=l1*ido; + t10=ip*ido; + + if(ido==1)goto L119; + for(ik=0;ikl1){ + for(j=1;j>1; + ipp2=ip; + ipph=(ip+1)>>1; + if(idol1)goto L139; + + is= -ido-1; + t1=0; + for(j=1;jn==1)return; + drftf1(l->n,data,l->trigcache,l->trigcache+l->n,l->splitcache); +} + +void drft_backward(drft_lookup *l,float *data){ + if (l->n==1)return; + drftb1(l->n,data,l->trigcache,l->trigcache+l->n,l->splitcache); +} + +void drft_init(drft_lookup *l,int n){ + l->n=n; + l->trigcache=_ogg_calloc(3*n,sizeof(*l->trigcache)); + l->splitcache=_ogg_calloc(32,sizeof(*l->splitcache)); + fdrffti(n, l->trigcache, l->splitcache); +} + +void drft_clear(drft_lookup *l){ + if(l){ + if(l->trigcache)_ogg_free(l->trigcache); + if(l->splitcache)_ogg_free(l->splitcache); + memset(l,0,sizeof(*l)); + } +} diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/smallft.h b/src/add-ons/media/plugins/vorbis/libvorbis/smallft.h new file mode 100644 index 0000000000..1a1e38763a --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/smallft.h @@ -0,0 +1,34 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: fft transform + last mod: $Id: smallft.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _V_SMFT_H_ +#define _V_SMFT_H_ + +#include "vorbis/codec.h" + +typedef struct { + int n; + float *trigcache; + int *splitcache; +} drft_lookup; + +extern void drft_forward(drft_lookup *l,float *data); +extern void drft_backward(drft_lookup *l,float *data); +extern void drft_init(drft_lookup *l,int n); +extern void drft_clear(drft_lookup *l); + +#endif diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/synthesis.c b/src/add-ons/media/plugins/vorbis/libvorbis/synthesis.c new file mode 100644 index 0000000000..1742c27d05 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/synthesis.c @@ -0,0 +1,155 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: single-block PCM synthesis + last mod: $Id: synthesis.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include "vorbis/codec.h" +#include "codec_internal.h" +#include "registry.h" +#include "misc.h" +#include "os.h" + +int vorbis_synthesis(vorbis_block *vb,ogg_packet *op){ + vorbis_dsp_state *vd=vb->vd; + backend_lookup_state *b=vd->backend_state; + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + oggpack_buffer *opb=&vb->opb; + int type,mode,i; + + /* first things first. Make sure decode is ready */ + _vorbis_block_ripcord(vb); + oggpack_readinit(opb,op->packet,op->bytes); + + /* Check the packet type */ + if(oggpack_read(opb,1)!=0){ + /* Oops. This is not an audio data packet */ + return(OV_ENOTAUDIO); + } + + /* read our mode and pre/post windowsize */ + mode=oggpack_read(opb,b->modebits); + if(mode==-1)return(OV_EBADPACKET); + + vb->mode=mode; + vb->W=ci->mode_param[mode]->blockflag; + if(vb->W){ + + /* this doesn;t get mapped through mode selection as it's used + only for window selection */ + vb->lW=oggpack_read(opb,1); + vb->nW=oggpack_read(opb,1); + if(vb->nW==-1) return(OV_EBADPACKET); + }else{ + vb->lW=0; + vb->nW=0; + } + + /* more setup */ + vb->granulepos=op->granulepos; + vb->sequence=op->packetno-3; /* first block is third packet */ + vb->eofflag=op->e_o_s; + + /* alloc pcm passback storage */ + vb->pcmend=ci->blocksizes[vb->W]; + vb->pcm=_vorbis_block_alloc(vb,sizeof(*vb->pcm)*vi->channels); + for(i=0;ichannels;i++) + vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i])); + + /* unpack_header enforces range checking */ + type=ci->map_type[ci->mode_param[mode]->mapping]; + + return(_mapping_P[type]->inverse(vb,ci->map_param[ci->mode_param[mode]-> + mapping])); +} + +/* used to track pcm position without actually performing decode. + Useful for sequential 'fast forward' */ +int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op){ + vorbis_dsp_state *vd=vb->vd; + backend_lookup_state *b=vd->backend_state; + vorbis_info *vi=vd->vi; + codec_setup_info *ci=vi->codec_setup; + oggpack_buffer *opb=&vb->opb; + int mode; + + /* first things first. Make sure decode is ready */ + _vorbis_block_ripcord(vb); + oggpack_readinit(opb,op->packet,op->bytes); + + /* Check the packet type */ + if(oggpack_read(opb,1)!=0){ + /* Oops. This is not an audio data packet */ + return(OV_ENOTAUDIO); + } + + /* read our mode and pre/post windowsize */ + mode=oggpack_read(opb,b->modebits); + if(mode==-1)return(OV_EBADPACKET); + + vb->mode=mode; + vb->W=ci->mode_param[mode]->blockflag; + if(vb->W){ + vb->lW=oggpack_read(opb,1); + vb->nW=oggpack_read(opb,1); + if(vb->nW==-1) return(OV_EBADPACKET); + }else{ + vb->lW=0; + vb->nW=0; + } + + /* more setup */ + vb->granulepos=op->granulepos; + vb->sequence=op->packetno-3; /* first block is third packet */ + vb->eofflag=op->e_o_s; + + /* no pcm */ + vb->pcmend=0; + vb->pcm=NULL; + + return(0); +} + +long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op){ + codec_setup_info *ci=vi->codec_setup; + oggpack_buffer opb; + int mode; + + oggpack_readinit(&opb,op->packet,op->bytes); + + /* Check the packet type */ + if(oggpack_read(&opb,1)!=0){ + /* Oops. This is not an audio data packet */ + return(OV_ENOTAUDIO); + } + + { + int modebits=0; + int v=ci->modes; + 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]); +} + + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/tone.c b/src/add-ons/media/plugins/vorbis/libvorbis/tone.c new file mode 100644 index 0000000000..daf2f93137 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/tone.c @@ -0,0 +1,54 @@ +#include +#include +#include +#include + +void usage(){ + fprintf(stderr,"tone ,[] [,[]...]\n"); + exit(1); +} + +int main (int argc,char *argv[]){ + int i,j; + double *f; + double *amp; + + if(argc<2)usage(); + + f=alloca(sizeof(*f)*(argc-1)); + amp=alloca(sizeof(*amp)*(argc-1)); + + i=0; + while(argv[i+1]){ + char *pos=strchr(argv[i+1],','); + + f[i]=atof(argv[i+1]); + if(pos) + amp[i]=atof(pos+1)*32767.f; + else + amp[i]=32767.f; + + fprintf(stderr,"%g Hz, %g amp\n",f[i],amp[i]); + + i++; + } + + for(i=0;i<44100*10;i++){ + float val=0; + int ival; + for(j=0;j32767.f)ival=32767.f; + if(ival<-32768.f)ival=-32768.f; + + fprintf(stdout,"%c%c%c%c", + (char)(ival&0xff), + (char)((ival>>8)&0xff), + (char)(ival&0xff), + (char)((ival>>8)&0xff)); + } + return(0); +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/vorbis/codec.h b/src/add-ons/media/plugins/vorbis/libvorbis/vorbis/codec.h new file mode 100644 index 0000000000..dfaa9a2691 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/vorbis/codec.h @@ -0,0 +1,233 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2001 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + + ******************************************************************** + + function: libvorbis codec headers + last mod: $Id: codec.h,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#ifndef _vorbis_codec_h_ +#define _vorbis_codec_h_ + +#ifdef __cplusplus +extern "C" +{ +#endif /* __cplusplus */ + +#include + +typedef struct vorbis_info{ + int version; + int channels; + long rate; + + /* The below bitrate declarations are *hints*. + Combinations of the three values carry the following implications: + + all three set to the same value: + implies a fixed rate bitstream + only nominal set: + implies a VBR stream that averages the nominal bitrate. No hard + upper/lower limit + upper and or lower set: + implies a VBR bitstream that obeys the bitrate limits. nominal + may also be set to give a nominal rate. + none set: + the coder does not care to speculate. + */ + + long bitrate_upper; + long bitrate_nominal; + long bitrate_lower; + long bitrate_window; + + void *codec_setup; +} vorbis_info; + +/* vorbis_dsp_state buffers the current vorbis audio + analysis/synthesis state. The DSP state belongs to a specific + logical bitstream ****************************************************/ +typedef struct vorbis_dsp_state{ + int analysisp; + vorbis_info *vi; + + float **pcm; + float **pcmret; + int pcm_storage; + int pcm_current; + int pcm_returned; + + int preextrapolate; + int eofflag; + + long lW; + long W; + long nW; + long centerW; + + ogg_int64_t granulepos; + ogg_int64_t sequence; + + ogg_int64_t glue_bits; + ogg_int64_t time_bits; + ogg_int64_t floor_bits; + ogg_int64_t res_bits; + + void *backend_state; +} vorbis_dsp_state; + +typedef struct vorbis_block{ + /* necessary stream state for linking to the framing abstraction */ + float **pcm; /* this is a pointer into local storage */ + oggpack_buffer opb; + + long lW; + long W; + long nW; + int pcmend; + int mode; + + int eofflag; + ogg_int64_t granulepos; + ogg_int64_t sequence; + vorbis_dsp_state *vd; /* For read-only access of configuration */ + + /* local storage to avoid remallocing; it's up to the mapping to + structure it */ + void *localstore; + long localtop; + long localalloc; + long totaluse; + struct alloc_chain *reap; + + /* bitmetrics for the frame */ + long glue_bits; + long time_bits; + long floor_bits; + long res_bits; + + void *internal; + +} vorbis_block; + +/* vorbis_block is a single block of data to be processed as part of +the analysis/synthesis stream; it belongs to a specific logical +bitstream, but is independant from other vorbis_blocks belonging to +that logical bitstream. *************************************************/ + +struct alloc_chain{ + void *ptr; + struct alloc_chain *next; +}; + +/* vorbis_info contains all the setup information specific to the + specific compression/decompression mode in progress (eg, + psychoacoustic settings, channel setup, options, codebook + etc). vorbis_info and substructures are in backends.h. +*********************************************************************/ + +/* the comments are not part of vorbis_info so that vorbis_info can be + static storage */ +typedef struct vorbis_comment{ + /* unlimited user comment fields. libvorbis writes 'libvorbis' + whatever vendor is set to in encode */ + char **user_comments; + int *comment_lengths; + int comments; + char *vendor; + +} vorbis_comment; + + +/* libvorbis encodes in two abstraction layers; first we perform DSP + and produce a packet (see docs/analysis.txt). The packet is then + coded into a framed OggSquish bitstream by the second layer (see + docs/framing.txt). Decode is the reverse process; we sync/frame + the bitstream and extract individual packets, then decode the + packet back into PCM audio. + + The extra framing/packetizing is used in streaming formats, such as + files. Over the net (such as with UDP), the framing and + packetization aren't necessary as they're provided by the transport + and the streaming layer is not used */ + +/* Vorbis PRIMITIVES: general ***************************************/ + +extern void vorbis_info_init(vorbis_info *vi); +extern void vorbis_info_clear(vorbis_info *vi); +extern int vorbis_info_blocksize(vorbis_info *vi,int zo); +extern void vorbis_comment_init(vorbis_comment *vc); +extern void vorbis_comment_add(vorbis_comment *vc, char *comment); +extern void vorbis_comment_add_tag(vorbis_comment *vc, + char *tag, char *contents); +extern char *vorbis_comment_query(vorbis_comment *vc, char *tag, int count); +extern int vorbis_comment_query_count(vorbis_comment *vc, char *tag); +extern void vorbis_comment_clear(vorbis_comment *vc); + +extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb); +extern int vorbis_block_clear(vorbis_block *vb); +extern void vorbis_dsp_clear(vorbis_dsp_state *v); + +/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/ + +extern int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi); +extern int vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op); +extern int vorbis_analysis_headerout(vorbis_dsp_state *v, + vorbis_comment *vc, + ogg_packet *op, + ogg_packet *op_comm, + ogg_packet *op_code); +extern float **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals); +extern int vorbis_analysis_wrote(vorbis_dsp_state *v,int vals); +extern int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb); +extern int vorbis_analysis(vorbis_block *vb,ogg_packet *op); + +extern int vorbis_bitrate_addblock(vorbis_block *vb); +extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, + ogg_packet *op); + +/* Vorbis PRIMITIVES: synthesis layer *******************************/ +extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc, + ogg_packet *op); + +extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi); +extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op); +extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op); +extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb); +extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm); +extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples); +extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op); + +/* Vorbis ERRORS and return codes ***********************************/ + +#define OV_FALSE -1 +#define OV_EOF -2 +#define OV_HOLE -3 + +#define OV_EREAD -128 +#define OV_EFAULT -129 +#define OV_EIMPL -130 +#define OV_EINVAL -131 +#define OV_ENOTVORBIS -132 +#define OV_EBADHEADER -133 +#define OV_EVERSION -134 +#define OV_ENOTAUDIO -135 +#define OV_EBADPACKET -136 +#define OV_EBADLINK -137 +#define OV_ENOSEEK -138 + +#ifdef __cplusplus +} +#endif /* __cplusplus */ + +#endif + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/vorbisenc.c b/src/add-ons/media/plugins/vorbis/libvorbis/vorbisenc.c new file mode 100644 index 0000000000..2da0afe93b --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/vorbisenc.c @@ -0,0 +1,1134 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: simple programmatic interface for encoder mode setup + last mod: $Id: vorbisenc.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include + +#include "vorbis/codec.h" +#include "vorbis/vorbisenc.h" + +#include "codec_internal.h" + +#include "os.h" +#include "misc.h" + +/* careful with this; it's using static array sizing to make managing + all the modes a little less annoying. If we use a residue backend + with > 12 partition types, or a different division of iteration, + this needs to be updated. */ +typedef struct { + static_codebook *books[12][3]; +} static_bookblock; + +typedef struct { + int res_type; + int limit_type; /* 0 lowpass limited, 1 point stereo limited */ + vorbis_info_residue0 *res; + static_codebook *book_aux; + static_codebook *book_aux_managed; + static_bookblock *books_base; + static_bookblock *books_base_managed; +} vorbis_residue_template; + +typedef struct { + vorbis_info_mapping0 *map; + vorbis_residue_template *res; +} vorbis_mapping_template; + +typedef struct vp_adjblock{ + int block[P_BANDS]; +} vp_adjblock; + +typedef struct { + int data[NOISE_COMPAND_LEVELS]; +} compandblock; + +/* high level configuration information for setting things up + step-by-step with the detailed vorbis_encode_ctl interface. + There's a fair amount of redundancy such that interactive setup + does not directly deal with any vorbis_info or codec_setup_info + initialization; it's all stored (until full init) in this highlevel + setup, then flushed out to the real codec setup structs later. */ + +typedef struct { + int att[P_NOISECURVES]; + float boost; + float decay; +} att3; +typedef struct { int data[P_NOISECURVES]; } adj3; + +typedef struct { + int pre[PACKETBLOBS]; + int post[PACKETBLOBS]; + float kHz[PACKETBLOBS]; + float lowpasskHz[PACKETBLOBS]; +} adj_stereo; + +typedef struct { + int lo; + int hi; + int fixed; +} noiseguard; +typedef struct { + int data[P_NOISECURVES][17]; +} noise3; + +typedef struct { + int mappings; + double *rate_mapping; + double *quality_mapping; + int coupling_restriction; + long samplerate_min_restriction; + long samplerate_max_restriction; + + + int *blocksize_short; + int *blocksize_long; + + att3 *psy_tone_masteratt; + int *psy_tone_0dB; + int *psy_tone_dBsuppress; + + vp_adjblock *psy_tone_adj_impulse; + vp_adjblock *psy_tone_adj_long; + vp_adjblock *psy_tone_adj_other; + + noiseguard *psy_noiseguards; + noise3 *psy_noise_bias_impulse; + noise3 *psy_noise_bias_padding; + noise3 *psy_noise_bias_trans; + noise3 *psy_noise_bias_long; + int *psy_noise_dBsuppress; + + compandblock *psy_noise_compand; + double *psy_noise_compand_short_mapping; + double *psy_noise_compand_long_mapping; + + int *psy_noise_normal_start[2]; + int *psy_noise_normal_partition[2]; + double *psy_noise_normal_thresh; + + int *psy_ath_float; + int *psy_ath_abs; + + double *psy_lowpass; + + vorbis_info_psy_global *global_params; + double *global_mapping; + adj_stereo *stereo_modes; + + static_codebook ***floor_books; + vorbis_info_floor1 *floor_params; + int *floor_short_mapping; + int *floor_long_mapping; + + vorbis_mapping_template *maps; +} ve_setup_data_template; + +/* a few static coder conventions */ +static vorbis_info_mode _mode_template[2]={ + {0,0,0,0}, + {1,0,0,1} +}; + +static vorbis_info_mapping0 _map_nominal[2]={ + {1, {0,0}, {0}, {0}, 1,{0},{1}}, + {1, {0,0}, {1}, {1}, 1,{0},{1}} +}; + +#include "modes/setup_44.h" +#include "modes/setup_44u.h" +#include "modes/setup_32.h" +#include "modes/setup_8.h" +#include "modes/setup_11.h" +#include "modes/setup_16.h" +#include "modes/setup_22.h" +#include "modes/setup_X.h" + +static ve_setup_data_template *setup_list[]={ + &ve_setup_44_stereo, + &ve_setup_44_stereo_low, + &ve_setup_44_uncoupled, + &ve_setup_44_uncoupled_low, + + &ve_setup_32_stereo, + &ve_setup_32_stereo_low, + &ve_setup_32_uncoupled, + &ve_setup_32_uncoupled_low, + + &ve_setup_22_stereo, + &ve_setup_22_uncoupled, + &ve_setup_16_stereo, + &ve_setup_16_uncoupled, + + &ve_setup_11_stereo, + &ve_setup_11_uncoupled, + &ve_setup_8_stereo, + &ve_setup_8_uncoupled, + + &ve_setup_X_stereo, + &ve_setup_X_uncoupled, + &ve_setup_X_stereo_low, + &ve_setup_X_uncoupled_low, + &ve_setup_XX_stereo, + &ve_setup_XX_uncoupled, + 0 +}; + +static int vorbis_encode_toplevel_setup(vorbis_info *vi,int ch,long rate){ + if(vi && vi->codec_setup){ + + vi->version=0; + vi->channels=ch; + vi->rate=rate; + + return(0); + } + return(OV_EINVAL); +} + +static void vorbis_encode_floor_setup(vorbis_info *vi,double s,int block, + static_codebook ***books, + vorbis_info_floor1 *in, + int *x){ + int i,k,is=s; + vorbis_info_floor1 *f=_ogg_calloc(1,sizeof(*f)); + codec_setup_info *ci=vi->codec_setup; + + memcpy(f,in+x[is],sizeof(*f)); + /* fill in the lowpass field, even if it's temporary */ + f->n=ci->blocksizes[block]>>1; + + /* books */ + { + int partitions=f->partitions; + int maxclass=-1; + int maxbook=-1; + for(i=0;ipartitionclass[i]>maxclass)maxclass=f->partitionclass[i]; + for(i=0;i<=maxclass;i++){ + if(f->class_book[i]>maxbook)maxbook=f->class_book[i]; + f->class_book[i]+=ci->books; + for(k=0;k<(1<class_subs[i]);k++){ + if(f->class_subbook[i][k]>maxbook)maxbook=f->class_subbook[i][k]; + if(f->class_subbook[i][k]>=0)f->class_subbook[i][k]+=ci->books; + } + } + + for(i=0;i<=maxbook;i++) + ci->book_param[ci->books++]=books[x[is]][i]; + } + + /* for now, we're only using floor 1 */ + ci->floor_type[ci->floors]=1; + ci->floor_param[ci->floors]=f; + ci->floors++; + + return; +} + +static void vorbis_encode_global_psych_setup(vorbis_info *vi,double s, + vorbis_info_psy_global *in, + double *x){ + int i,is=s; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *g=&ci->psy_g_param; + + memcpy(g,in+(int)x[is],sizeof(*g)); + + ds=x[is]*(1.-ds)+x[is+1]*ds; + is=(int)ds; + ds-=is; + if(ds==0 && is>0){ + is--; + ds=1.; + } + + /* interpolate the trigger threshholds */ + for(i=0;i<4;i++){ + g->preecho_thresh[i]=in[is].preecho_thresh[i]*(1.-ds)+in[is+1].preecho_thresh[i]*ds; + g->postecho_thresh[i]=in[is].postecho_thresh[i]*(1.-ds)+in[is+1].postecho_thresh[i]*ds; + } + g->ampmax_att_per_sec=ci->hi.amplitude_track_dBpersec; + return; +} + +static void vorbis_encode_global_stereo(vorbis_info *vi, + highlevel_encode_setup *hi, + adj_stereo *p){ + float s=hi->stereo_point_setting; + int i,is=s; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy_global *g=&ci->psy_g_param; + + if(p){ + memcpy(g->coupling_prepointamp,p[is].pre,sizeof(*p[is].pre)*PACKETBLOBS); + memcpy(g->coupling_postpointamp,p[is].post,sizeof(*p[is].post)*PACKETBLOBS); + + if(hi->managed){ + /* interpolate the kHz threshholds */ + for(i=0;icoupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0]; + g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1]; + g->coupling_pkHz[i]=kHz; + + kHz=p[is].lowpasskHz[i]*(1.-ds)+p[is+1].lowpasskHz[i]*ds; + g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0]; + g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1]; + + } + }else{ + float kHz=p[is].kHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].kHz[PACKETBLOBS/2]*ds; + for(i=0;icoupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0]; + g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1]; + g->coupling_pkHz[i]=kHz; + } + + kHz=p[is].lowpasskHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].lowpasskHz[PACKETBLOBS/2]*ds; + for(i=0;isliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0]; + g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1]; + } + } + }else{ + for(i=0;isliding_lowpass[0][i]=ci->blocksizes[0]; + g->sliding_lowpass[1][i]=ci->blocksizes[1]; + } + } + return; +} + +static void vorbis_encode_psyset_setup(vorbis_info *vi,double s, + int *nn_start, + int *nn_partition, + double *nn_thresh, + int block){ + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + highlevel_encode_setup *hi=&ci->hi; + int is=s; + + if(block>=ci->psys) + ci->psys=block+1; + if(!p){ + p=_ogg_calloc(1,sizeof(*p)); + ci->psy_param[block]=p; + } + + memcpy(p,&_psy_info_template,sizeof(*p)); + p->blockflag=block>>1; + + if(hi->noise_normalize_p){ + p->normal_channel_p=1; + p->normal_point_p=1; + p->normal_start=nn_start[is]; + p->normal_partition=nn_partition[is]; + p->normal_thresh=nn_thresh[is]; + } + + return; +} + +static void vorbis_encode_tonemask_setup(vorbis_info *vi,double s,int block, + att3 *att, + int *max, + vp_adjblock *in){ + int i,is=s; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + + /* 0 and 2 are only used by bitmanagement, but there's no harm to always + filling the values in here */ + p->tone_masteratt[0]=att[is].att[0]*(1.-ds)+att[is+1].att[0]*ds; + p->tone_masteratt[1]=att[is].att[1]*(1.-ds)+att[is+1].att[1]*ds; + p->tone_masteratt[2]=att[is].att[2]*(1.-ds)+att[is+1].att[2]*ds; + p->tone_centerboost=att[is].boost*(1.-ds)+att[is+1].boost*ds; + p->tone_decay=att[is].decay*(1.-ds)+att[is+1].decay*ds; + + p->max_curve_dB=max[is]*(1.-ds)+max[is+1]*ds; + + for(i=0;itoneatt[i]=in[is].block[i]*(1.-ds)+in[is+1].block[i]*ds; + return; +} + + +static void vorbis_encode_compand_setup(vorbis_info *vi,double s,int block, + compandblock *in, double *x){ + int i,is=s; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + + ds=x[is]*(1.-ds)+x[is+1]*ds; + is=(int)ds; + ds-=is; + if(ds==0 && is>0){ + is--; + ds=1.; + } + + /* interpolate the compander settings */ + for(i=0;inoisecompand[i]=in[is].data[i]*(1.-ds)+in[is+1].data[i]*ds; + return; +} + +static void vorbis_encode_peak_setup(vorbis_info *vi,double s,int block, + int *suppress){ + int is=s; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + + p->tone_abs_limit=suppress[is]*(1.-ds)+suppress[is+1]*ds; + + return; +} + +static void vorbis_encode_noisebias_setup(vorbis_info *vi,double s,int block, + int *suppress, + noise3 *in, + noiseguard *guard, + double userbias){ + int i,is=s,j; + double ds=s-is; + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + + p->noisemaxsupp=suppress[is]*(1.-ds)+suppress[is+1]*ds; + p->noisewindowlomin=guard[block].lo; + p->noisewindowhimin=guard[block].hi; + p->noisewindowfixed=guard[block].fixed; + + for(j=0;jnoiseoff[j][i]=in[is].data[j][i]*(1.-ds)+in[is+1].data[j][i]*ds; + + /* impulse blocks may take a user specified bias to boost the + nominal/high noise encoding depth */ + for(j=0;jnoiseoff[j][0]+6; /* the lowest it can go */ + for(i=0;inoiseoff[j][i]+=userbias; + if(p->noiseoff[j][i]noiseoff[j][i]=min; + } + } + + return; +} + +static void vorbis_encode_ath_setup(vorbis_info *vi,int block){ + codec_setup_info *ci=vi->codec_setup; + vorbis_info_psy *p=ci->psy_param[block]; + + p->ath_adjatt=ci->hi.ath_floating_dB; + p->ath_maxatt=ci->hi.ath_absolute_dB; + return; +} + + +static int book_dup_or_new(codec_setup_info *ci,static_codebook *book){ + int i; + for(i=0;ibooks;i++) + if(ci->book_param[i]==book)return(i); + + return(ci->books++); +} + +static void vorbis_encode_blocksize_setup(vorbis_info *vi,double s, + int *shortb,int *longb){ + + codec_setup_info *ci=vi->codec_setup; + int is=s; + + int blockshort=shortb[is]; + int blocklong=longb[is]; + ci->blocksizes[0]=blockshort; + ci->blocksizes[1]=blocklong; + +} + +static void vorbis_encode_residue_setup(vorbis_info *vi, + int number, int block, + vorbis_residue_template *res){ + + codec_setup_info *ci=vi->codec_setup; + int i,n; + + vorbis_info_residue0 *r=ci->residue_param[number]= + _ogg_malloc(sizeof(*r)); + + memcpy(r,res->res,sizeof(*r)); + if(ci->residues<=number)ci->residues=number+1; + + switch(ci->blocksizes[block]){ + case 64:case 128:case 256: + r->grouping=16; + break; + default: + r->grouping=32; + break; + } + ci->residue_type[number]=res->res_type; + + /* to be adjusted by lowpass/pointlimit later */ + n=r->end=ci->blocksizes[block]>>1; + if(res->res_type==2) + n=r->end*=vi->channels; + + /* fill in all the books */ + { + int booklist=0,k; + + if(ci->hi.managed){ + for(i=0;ipartitions;i++) + for(k=0;k<3;k++) + if(res->books_base_managed->books[i][k]) + r->secondstages[i]|=(1<groupbook=book_dup_or_new(ci,res->book_aux_managed); + ci->book_param[r->groupbook]=res->book_aux_managed; + + for(i=0;ipartitions;i++){ + for(k=0;k<3;k++){ + if(res->books_base_managed->books[i][k]){ + int bookid=book_dup_or_new(ci,res->books_base_managed->books[i][k]); + r->booklist[booklist++]=bookid; + ci->book_param[bookid]=res->books_base_managed->books[i][k]; + } + } + } + + }else{ + + for(i=0;ipartitions;i++) + for(k=0;k<3;k++) + if(res->books_base->books[i][k]) + r->secondstages[i]|=(1<groupbook=book_dup_or_new(ci,res->book_aux); + ci->book_param[r->groupbook]=res->book_aux; + + for(i=0;ipartitions;i++){ + for(k=0;k<3;k++){ + if(res->books_base->books[i][k]){ + int bookid=book_dup_or_new(ci,res->books_base->books[i][k]); + r->booklist[booklist++]=bookid; + ci->book_param[bookid]=res->books_base->books[i][k]; + } + } + } + } + } + + /* lowpass setup/pointlimit */ + { + double freq=ci->hi.lowpass_kHz*1000.; + vorbis_info_floor1 *f=ci->floor_param[block]; /* by convention */ + double nyq=vi->rate/2.; + long blocksize=ci->blocksizes[block]>>1; + + /* lowpass needs to be set in the floor and the residue. */ + if(freq>nyq)freq=nyq; + /* in the floor, the granularity can be very fine; it doesn't alter + the encoding structure, only the samples used to fit the floor + approximation */ + f->n=freq/nyq*blocksize; + + /* this res may by limited by the maximum pointlimit of the mode, + not the lowpass. the floor is always lowpass limited. */ + if(res->limit_type){ + if(ci->hi.managed) + freq=ci->psy_g_param.coupling_pkHz[PACKETBLOBS-1]*1000.; + else + freq=ci->psy_g_param.coupling_pkHz[PACKETBLOBS/2]*1000.; + if(freq>nyq)freq=nyq; + } + + /* in the residue, we're constrained, physically, by partition + boundaries. We still lowpass 'wherever', but we have to round up + here to next boundary, or the vorbis spec will round it *down* to + previous boundary in encode/decode */ + if(ci->residue_type[block]==2) + r->end=(int)((freq/nyq*blocksize*2)/r->grouping+.9)* /* round up only if we're well past */ + r->grouping; + else + r->end=(int)((freq/nyq*blocksize)/r->grouping+.9)* /* round up only if we're well past */ + r->grouping; + } +} + +/* we assume two maps in this encoder */ +static void vorbis_encode_map_n_res_setup(vorbis_info *vi,double s, + vorbis_mapping_template *maps){ + + codec_setup_info *ci=vi->codec_setup; + int i,j,is=s,modes=2; + vorbis_info_mapping0 *map=maps[is].map; + vorbis_info_mode *mode=_mode_template; + vorbis_residue_template *res=maps[is].res; + + if(ci->blocksizes[0]==ci->blocksizes[1])modes=1; + + for(i=0;imap_param[i]=_ogg_calloc(1,sizeof(*map)); + ci->mode_param[i]=_ogg_calloc(1,sizeof(*mode)); + + memcpy(ci->mode_param[i],mode+i,sizeof(*_mode_template)); + if(i>=ci->modes)ci->modes=i+1; + + ci->map_type[i]=0; + memcpy(ci->map_param[i],map+i,sizeof(*map)); + if(i>=ci->maps)ci->maps=i+1; + + for(j=0;jcodec_setup; + highlevel_encode_setup *hi=&ci->hi; + ve_setup_data_template *setup=(ve_setup_data_template *)hi->setup; + int is=hi->base_setting; + double ds=hi->base_setting-is; + int ch=vi->channels; + double *r=setup->rate_mapping; + + if(r==NULL) + return(-1); + + return((r[is]*(1.-ds)+r[is+1]*ds)*ch); +} + +static void get_setup_template(vorbis_info *vi, + long ch,long srate, + double req,int q_or_bitrate){ + int i=0,j; + codec_setup_info *ci=vi->codec_setup; + highlevel_encode_setup *hi=&ci->hi; + if(q_or_bitrate)req/=ch; + + while(setup_list[i]){ + if(setup_list[i]->coupling_restriction==-1 || + setup_list[i]->coupling_restriction==ch){ + if(srate>=setup_list[i]->samplerate_min_restriction && + srate<=setup_list[i]->samplerate_max_restriction){ + int mappings=setup_list[i]->mappings; + double *map=(q_or_bitrate? + setup_list[i]->rate_mapping: + setup_list[i]->quality_mapping); + + /* the template matches. Does the requested quality mode + fall within this template's modes? */ + if(reqmap[setup_list[i]->mappings]){++i;continue;} + for(j=0;j=map[j] && reqsetup=setup_list[i]; + if(j==mappings) + hi->base_setting=j-.001; + else{ + float low=map[j]; + float high=map[j+1]; + float del=(req-low)/(high-low); + hi->base_setting=j+del; + } + return; + } + } + i++; + } + + hi->setup=NULL; +} + +/* encoders will need to use vorbis_info_init beforehand and call + vorbis_info clear when all done */ + +/* two interfaces; this, more detailed one, and later a convenience + layer on top */ + +/* the final setup call */ +int vorbis_encode_setup_init(vorbis_info *vi){ + int i0=0,singleblock=0; + codec_setup_info *ci=vi->codec_setup; + ve_setup_data_template *setup=NULL; + highlevel_encode_setup *hi=&ci->hi; + + if(ci==NULL)return(OV_EINVAL); + if(!hi->impulse_block_p)i0=1; + + /* too low/high an ATH floater is nonsensical, but doesn't break anything */ + if(hi->ath_floating_dB>-80)hi->ath_floating_dB=-80; + if(hi->ath_floating_dB<-200)hi->ath_floating_dB=-200; + + /* again, bound this to avoid the app shooting itself int he foot + too badly */ + if(hi->amplitude_track_dBpersec>0.)hi->amplitude_track_dBpersec=0.; + if(hi->amplitude_track_dBpersec<-99999.)hi->amplitude_track_dBpersec=-99999.; + + /* get the appropriate setup template; matches the fetch in previous + stages */ + setup=(ve_setup_data_template *)hi->setup; + if(setup==NULL)return(OV_EINVAL); + + hi->set_in_stone=1; + /* choose block sizes from configured sizes as well as paying + attention to long_block_p and short_block_p. If the configured + short and long blocks are the same length, we set long_block_p + and unset short_block_p */ + vorbis_encode_blocksize_setup(vi,hi->base_setting, + setup->blocksize_short, + setup->blocksize_long); + if(ci->blocksizes[0]==ci->blocksizes[1])singleblock=1; + + /* floor setup; choose proper floor params. Allocated on the floor + stack in order; if we alloc only long floor, it's 0 */ + vorbis_encode_floor_setup(vi,hi->short_setting,0, + setup->floor_books, + setup->floor_params, + setup->floor_short_mapping); + if(!singleblock) + vorbis_encode_floor_setup(vi,hi->long_setting,1, + setup->floor_books, + setup->floor_params, + setup->floor_long_mapping); + + /* setup of [mostly] short block detection and stereo*/ + vorbis_encode_global_psych_setup(vi,hi->trigger_setting, + setup->global_params, + setup->global_mapping); + vorbis_encode_global_stereo(vi,hi,setup->stereo_modes); + + /* basic psych setup and noise normalization */ + vorbis_encode_psyset_setup(vi,hi->short_setting, + setup->psy_noise_normal_start[0], + setup->psy_noise_normal_partition[0], + setup->psy_noise_normal_thresh, + 0); + vorbis_encode_psyset_setup(vi,hi->short_setting, + setup->psy_noise_normal_start[0], + setup->psy_noise_normal_partition[0], + setup->psy_noise_normal_thresh, + 1); + if(!singleblock){ + vorbis_encode_psyset_setup(vi,hi->long_setting, + setup->psy_noise_normal_start[1], + setup->psy_noise_normal_partition[1], + setup->psy_noise_normal_thresh, + 2); + vorbis_encode_psyset_setup(vi,hi->long_setting, + setup->psy_noise_normal_start[1], + setup->psy_noise_normal_partition[1], + setup->psy_noise_normal_thresh, + 3); + } + + /* tone masking setup */ + vorbis_encode_tonemask_setup(vi,hi->block[i0].tone_mask_setting,0, + setup->psy_tone_masteratt, + setup->psy_tone_0dB, + setup->psy_tone_adj_impulse); + vorbis_encode_tonemask_setup(vi,hi->block[1].tone_mask_setting,1, + setup->psy_tone_masteratt, + setup->psy_tone_0dB, + setup->psy_tone_adj_other); + if(!singleblock){ + vorbis_encode_tonemask_setup(vi,hi->block[2].tone_mask_setting,2, + setup->psy_tone_masteratt, + setup->psy_tone_0dB, + setup->psy_tone_adj_other); + vorbis_encode_tonemask_setup(vi,hi->block[3].tone_mask_setting,3, + setup->psy_tone_masteratt, + setup->psy_tone_0dB, + setup->psy_tone_adj_long); + } + + /* noise companding setup */ + vorbis_encode_compand_setup(vi,hi->block[i0].noise_compand_setting,0, + setup->psy_noise_compand, + setup->psy_noise_compand_short_mapping); + vorbis_encode_compand_setup(vi,hi->block[1].noise_compand_setting,1, + setup->psy_noise_compand, + setup->psy_noise_compand_short_mapping); + if(!singleblock){ + vorbis_encode_compand_setup(vi,hi->block[2].noise_compand_setting,2, + setup->psy_noise_compand, + setup->psy_noise_compand_long_mapping); + vorbis_encode_compand_setup(vi,hi->block[3].noise_compand_setting,3, + setup->psy_noise_compand, + setup->psy_noise_compand_long_mapping); + } + + /* peak guarding setup */ + vorbis_encode_peak_setup(vi,hi->block[i0].tone_peaklimit_setting,0, + setup->psy_tone_dBsuppress); + vorbis_encode_peak_setup(vi,hi->block[1].tone_peaklimit_setting,1, + setup->psy_tone_dBsuppress); + if(!singleblock){ + vorbis_encode_peak_setup(vi,hi->block[2].tone_peaklimit_setting,2, + setup->psy_tone_dBsuppress); + vorbis_encode_peak_setup(vi,hi->block[3].tone_peaklimit_setting,3, + setup->psy_tone_dBsuppress); + } + + /* noise bias setup */ + vorbis_encode_noisebias_setup(vi,hi->block[i0].noise_bias_setting,0, + setup->psy_noise_dBsuppress, + setup->psy_noise_bias_impulse, + setup->psy_noiseguards, + (i0==0?hi->impulse_noisetune:0.)); + vorbis_encode_noisebias_setup(vi,hi->block[1].noise_bias_setting,1, + setup->psy_noise_dBsuppress, + setup->psy_noise_bias_padding, + setup->psy_noiseguards,0.); + if(!singleblock){ + vorbis_encode_noisebias_setup(vi,hi->block[2].noise_bias_setting,2, + setup->psy_noise_dBsuppress, + setup->psy_noise_bias_trans, + setup->psy_noiseguards,0.); + vorbis_encode_noisebias_setup(vi,hi->block[3].noise_bias_setting,3, + setup->psy_noise_dBsuppress, + setup->psy_noise_bias_long, + setup->psy_noiseguards,0.); + } + + vorbis_encode_ath_setup(vi,0); + vorbis_encode_ath_setup(vi,1); + if(!singleblock){ + vorbis_encode_ath_setup(vi,2); + vorbis_encode_ath_setup(vi,3); + } + + vorbis_encode_map_n_res_setup(vi,hi->base_setting,setup->maps); + + /* set bitrate readonlies and management */ + vi->bitrate_nominal=setting_to_approx_bitrate(vi); + vi->bitrate_lower=hi->bitrate_min; + vi->bitrate_upper=hi->bitrate_max; + vi->bitrate_window=hi->bitrate_limit_window; + + if(hi->managed){ + ci->bi.queue_avg_time=hi->bitrate_av_window; + ci->bi.queue_avg_center=hi->bitrate_av_window_center; + ci->bi.queue_minmax_time=hi->bitrate_limit_window; + ci->bi.queue_hardmin=hi->bitrate_min; + ci->bi.queue_hardmax=hi->bitrate_max; + ci->bi.queue_avgmin=hi->bitrate_av_lo; + ci->bi.queue_avgmax=hi->bitrate_av_hi; + ci->bi.avgfloat_downslew_max=-999999.f; + ci->bi.avgfloat_upslew_max=999999.f; + } + + return(0); + +} + +static int vorbis_encode_setup_setting(vorbis_info *vi, + long channels, + long rate){ + int ret=0,i,is; + codec_setup_info *ci=vi->codec_setup; + highlevel_encode_setup *hi=&ci->hi; + ve_setup_data_template *setup=hi->setup; + double ds; + + ret=vorbis_encode_toplevel_setup(vi,channels,rate); + if(ret)return(ret); + + is=hi->base_setting; + ds=hi->base_setting-is; + + hi->short_setting=hi->base_setting; + hi->long_setting=hi->base_setting; + + hi->managed=0; + + hi->impulse_block_p=1; + hi->noise_normalize_p=1; + + hi->stereo_point_setting=hi->base_setting; + hi->lowpass_kHz= + setup->psy_lowpass[is]*(1.-ds)+setup->psy_lowpass[is+1]*ds; + + hi->ath_floating_dB=setup->psy_ath_float[is]*(1.-ds)+ + setup->psy_ath_float[is+1]*ds; + hi->ath_absolute_dB=setup->psy_ath_abs[is]*(1.-ds)+ + setup->psy_ath_abs[is+1]*ds; + + hi->amplitude_track_dBpersec=-6.; + hi->trigger_setting=hi->base_setting; + + for(i=0;i<4;i++){ + hi->block[i].tone_mask_setting=hi->base_setting; + hi->block[i].tone_peaklimit_setting=hi->base_setting; + hi->block[i].noise_bias_setting=hi->base_setting; + hi->block[i].noise_compand_setting=hi->base_setting; + } + + return(ret); +} + +int vorbis_encode_setup_vbr(vorbis_info *vi, + long channels, + long rate, + float quality){ + codec_setup_info *ci=vi->codec_setup; + highlevel_encode_setup *hi=&ci->hi; + + quality+=.00001; + if(quality>=1.)quality=.9999; + + get_setup_template(vi,channels,rate,quality,0); + if(!hi->setup)return OV_EIMPL; + + return vorbis_encode_setup_setting(vi,channels,rate); +} + +int vorbis_encode_init_vbr(vorbis_info *vi, + long channels, + long rate, + + float base_quality /* 0. to 1. */ + ){ + int ret=0; + + ret=vorbis_encode_setup_vbr(vi,channels,rate,base_quality); + + if(ret){ + vorbis_info_clear(vi); + return ret; + } + ret=vorbis_encode_setup_init(vi); + if(ret) + vorbis_info_clear(vi); + return(ret); +} + +int vorbis_encode_setup_managed(vorbis_info *vi, + long channels, + long rate, + + long max_bitrate, + long nominal_bitrate, + long min_bitrate){ + + codec_setup_info *ci=vi->codec_setup; + highlevel_encode_setup *hi=&ci->hi; + double tnominal=nominal_bitrate; + int ret=0; + + if(nominal_bitrate<=0.){ + if(max_bitrate>0.){ + nominal_bitrate=max_bitrate*.875; + }else{ + if(min_bitrate>0.){ + nominal_bitrate=min_bitrate; + }else{ + return(OV_EINVAL); + } + } + } + + get_setup_template(vi,channels,rate,nominal_bitrate,1); + if(!hi->setup)return OV_EIMPL; + + ret=vorbis_encode_setup_setting(vi,channels,rate); + if(ret){ + vorbis_info_clear(vi); + return ret; + } + + /* initialize management with sane defaults */ + /* initialize management with sane defaults */ + hi->managed=1; + hi->bitrate_av_window=4.; + hi->bitrate_av_window_center=.5; + hi->bitrate_limit_window=2.; + hi->bitrate_min=min_bitrate; + hi->bitrate_max=max_bitrate; + hi->bitrate_av_lo=tnominal; + hi->bitrate_av_hi=tnominal; + + return(ret); + +} + +int vorbis_encode_init(vorbis_info *vi, + long channels, + long rate, + + long max_bitrate, + long nominal_bitrate, + long min_bitrate){ + + int ret=vorbis_encode_setup_managed(vi,channels,rate, + max_bitrate, + nominal_bitrate, + min_bitrate); + if(ret){ + vorbis_info_clear(vi); + return(ret); + } + + ret=vorbis_encode_setup_init(vi); + if(ret) + vorbis_info_clear(vi); + return(ret); +} + +int vorbis_encode_ctl(vorbis_info *vi,int number,void *arg){ + if(vi){ + codec_setup_info *ci=vi->codec_setup; + highlevel_encode_setup *hi=&ci->hi; + int setp=(number&0xf); /* a read request has a low nibble of 0 */ + + if(setp && hi->set_in_stone)return(OV_EINVAL); + + switch(number){ + case OV_ECTL_RATEMANAGE_GET: + { + + struct ovectl_ratemanage_arg *ai= + (struct ovectl_ratemanage_arg *)arg; + + ai->management_active=hi->managed; + ai->bitrate_av_window=hi->bitrate_av_window; + ai->bitrate_av_window_center=hi->bitrate_av_window_center; + ai->bitrate_hard_window=hi->bitrate_limit_window; + ai->bitrate_hard_min=hi->bitrate_min; + ai->bitrate_hard_max=hi->bitrate_max; + ai->bitrate_av_lo=hi->bitrate_av_lo; + ai->bitrate_av_hi=hi->bitrate_av_hi; + + } + return(0); + + case OV_ECTL_RATEMANAGE_SET: + { + struct ovectl_ratemanage_arg *ai= + (struct ovectl_ratemanage_arg *)arg; + if(ai==NULL){ + hi->managed=0; + }else{ + hi->managed=ai->management_active; + vorbis_encode_ctl(vi,OV_ECTL_RATEMANAGE_AVG,arg); + vorbis_encode_ctl(vi,OV_ECTL_RATEMANAGE_HARD,arg); + } + } + return 0; + + case OV_ECTL_RATEMANAGE_AVG: + { + struct ovectl_ratemanage_arg *ai= + (struct ovectl_ratemanage_arg *)arg; + if(ai==NULL){ + hi->bitrate_av_lo=0; + hi->bitrate_av_hi=0; + hi->bitrate_av_window=0; + }else{ + hi->bitrate_av_window=ai->bitrate_av_window; + hi->bitrate_av_window_center=ai->bitrate_av_window_center; + hi->bitrate_av_lo=ai->bitrate_av_lo; + hi->bitrate_av_hi=ai->bitrate_av_hi; + } + + if(hi->bitrate_av_window<.25)hi->bitrate_av_window=.25; + if(hi->bitrate_av_window>10.)hi->bitrate_av_window=10.; + if(hi->bitrate_av_window_center<0.)hi->bitrate_av_window=0.; + if(hi->bitrate_av_window_center>1.)hi->bitrate_av_window=1.; + + if( ( (hi->bitrate_av_lo<=0 && hi->bitrate_av_hi<=0)|| + (hi->bitrate_av_window<=0) ) && + ( (hi->bitrate_min<=0 && hi->bitrate_max<=0)|| + (hi->bitrate_limit_window<=0) )) + hi->managed=0; + } + return(0); + case OV_ECTL_RATEMANAGE_HARD: + { + struct ovectl_ratemanage_arg *ai= + (struct ovectl_ratemanage_arg *)arg; + if(ai==NULL){ + hi->bitrate_min=0; + hi->bitrate_max=0; + hi->bitrate_limit_window=0; + }else{ + hi->bitrate_limit_window=ai->bitrate_hard_window; + hi->bitrate_min=ai->bitrate_hard_min; + hi->bitrate_max=ai->bitrate_hard_max; + } + if(hi->bitrate_limit_window<0.)hi->bitrate_limit_window=0.; + if(hi->bitrate_limit_window>10.)hi->bitrate_limit_window=10.; + + if( ( (hi->bitrate_av_lo<=0 && hi->bitrate_av_hi<=0)|| + (hi->bitrate_av_window<=0) ) && + ( (hi->bitrate_min<=0 && hi->bitrate_max<=0)|| + (hi->bitrate_limit_window<=0) )) + hi->managed=0; + } + return(0); + + case OV_ECTL_LOWPASS_GET: + { + double *farg=(double *)arg; + *farg=hi->lowpass_kHz; + } + return(0); + case OV_ECTL_LOWPASS_SET: + { + double *farg=(double *)arg; + hi->lowpass_kHz=*farg; + + if(hi->lowpass_kHz<2.)hi->lowpass_kHz=2.; + if(hi->lowpass_kHz>99.)hi->lowpass_kHz=99.; + } + return(0); + case OV_ECTL_IBLOCK_GET: + { + double *farg=(double *)arg; + *farg=hi->impulse_noisetune; + } + return(0); + case OV_ECTL_IBLOCK_SET: + { + double *farg=(double *)arg; + hi->impulse_noisetune=*farg; + + if(hi->impulse_noisetune>0.)hi->impulse_noisetune=0.; + if(hi->impulse_noisetune<-15.)hi->impulse_noisetune=-15.; + } + return(0); + } + + + return(OV_EIMPL); + } + return(OV_EINVAL); +} diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/vorbisfile.c b/src/add-ons/media/plugins/vorbis/libvorbis/vorbisfile.c new file mode 100644 index 0000000000..d5e9812ce5 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/vorbisfile.c @@ -0,0 +1,1619 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: stdio-based convenience library for opening/seeking/decoding + last mod: $Id: vorbisfile.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include +#include +#include + +#include "vorbis/codec.h" +#include "vorbis/vorbisfile.h" + +#include "os.h" +#include "misc.h" + +/* A 'chained bitstream' is a Vorbis bitstream that contains more than + one logical bitstream arranged end to end (the only form of Ogg + multiplexing allowed in a Vorbis bitstream; grouping [parallel + multiplexing] is not allowed in Vorbis) */ + +/* A Vorbis file can be played beginning to end (streamed) without + worrying ahead of time about chaining (see decoder_example.c). If + we have the whole file, however, and want random access + (seeking/scrubbing) or desire to know the total length/time of a + file, we need to account for the possibility of chaining. */ + +/* We can handle things a number of ways; we can determine the entire + bitstream structure right off the bat, or find pieces on demand. + This example determines and caches structure for the entire + bitstream, but builds a virtual decoder on the fly when moving + between links in the chain. */ + +/* There are also different ways to implement seeking. Enough + information exists in an Ogg bitstream to seek to + sample-granularity positions in the output. Or, one can seek by + picking some portion of the stream roughly in the desired area if + we only want coarse navigation through the stream. */ + +/************************************************************************* + * Many, many internal helpers. The intention is not to be confusing; + * rampant duplication and monolithic function implementation would be + * harder to understand anyway. The high level functions are last. Begin + * grokking near the end of the file */ + +/* read a little more data from the file/pipe into the ogg_sync framer +*/ +#define CHUNKSIZE 8500 /* a shade over 8k; anyone using pages well + over 8k gets what they deserve */ +static long _get_data(OggVorbis_File *vf){ + errno=0; + if(vf->datasource){ + char *buffer=ogg_sync_buffer(&vf->oy,CHUNKSIZE); + long bytes=(vf->callbacks.read_func)(buffer,1,CHUNKSIZE,vf->datasource); + if(bytes>0)ogg_sync_wrote(&vf->oy,bytes); + if(bytes==0 && errno)return(-1); + return(bytes); + }else + return(0); +} + +/* save a tiny smidge of verbosity to make the code more readable */ +static void _seek_helper(OggVorbis_File *vf,ogg_int64_t offset){ + if(vf->datasource){ + (vf->callbacks.seek_func)(vf->datasource, offset, SEEK_SET); + vf->offset=offset; + ogg_sync_reset(&vf->oy); + }else{ + /* shouldn't happen unless someone writes a broken callback */ + return; + } +} + +/* The read/seek functions track absolute position within the stream */ + +/* from the head of the stream, get the next page. boundary specifies + if the function is allowed to fetch more data from the stream (and + how much) or only use internally buffered data. + + boundary: -1) unbounded search + 0) read no additional data; use cached only + n) search for a new page beginning for n bytes + + return: <0) did not find a page (OV_FALSE, OV_EOF, OV_EREAD) + n) found a page at absolute offset n */ + +static ogg_int64_t _get_next_page(OggVorbis_File *vf,ogg_page *og, + ogg_int64_t boundary){ + if(boundary>0)boundary+=vf->offset; + while(1){ + long more; + + if(boundary>0 && vf->offset>=boundary)return(OV_FALSE); + more=ogg_sync_pageseek(&vf->oy,og); + + if(more<0){ + /* skipped n bytes */ + vf->offset-=more; + }else{ + if(more==0){ + /* send more paramedics */ + if(!boundary)return(OV_FALSE); + { + long ret=_get_data(vf); + if(ret==0)return(OV_EOF); + if(ret<0)return(OV_EREAD); + } + }else{ + /* got a page. Return the offset at the page beginning, + advance the internal offset past the page end */ + ogg_int64_t ret=vf->offset; + vf->offset+=more; + return(ret); + + } + } + } +} + +/* find the latest page beginning before the current stream cursor + position. Much dirtier than the above as Ogg doesn't have any + backward search linkage. no 'readp' as it will certainly have to + read. */ +/* returns offset or OV_EREAD, OV_FAULT */ +static ogg_int64_t _get_prev_page(OggVorbis_File *vf,ogg_page *og){ + ogg_int64_t begin=vf->offset; + ogg_int64_t end=begin; + ogg_int64_t ret; + ogg_int64_t offset=-1; + + while(offset==-1){ + begin-=CHUNKSIZE; + if(begin<0) + begin=0; + _seek_helper(vf,begin); + while(vf->offsetoffset); + if(ret==OV_EREAD)return(OV_EREAD); + if(ret<0){ + break; + }else{ + offset=ret; + } + } + } + + /* we have the offset. Actually snork and hold the page now */ + _seek_helper(vf,offset); + ret=_get_next_page(vf,og,CHUNKSIZE); + if(ret<0) + /* this shouldn't be possible */ + return(OV_EFAULT); + + return(offset); +} + +/* finds each bitstream link one at a time using a bisection search + (has to begin by knowing the offset of the lb's initial page). + Recurses for each link so it can alloc the link storage after + finding them all, then unroll and fill the cache at the same time */ +static int _bisect_forward_serialno(OggVorbis_File *vf, + ogg_int64_t begin, + ogg_int64_t searched, + ogg_int64_t end, + long currentno, + long m){ + ogg_int64_t endsearched=end; + ogg_int64_t next=end; + ogg_page og; + ogg_int64_t ret; + + /* the below guards against garbage seperating the last and + first pages of two links. */ + while(searched=0)next=ret; + }else{ + searched=ret+og.header_len+og.body_len; + } + } + + _seek_helper(vf,next); + ret=_get_next_page(vf,&og,-1); + if(ret==OV_EREAD)return(OV_EREAD); + + if(searched>=end || ret<0){ + vf->links=m+1; + vf->offsets=_ogg_malloc((vf->links+1)*sizeof(*vf->offsets)); + vf->serialnos=_ogg_malloc(vf->links*sizeof(*vf->serialnos)); + vf->offsets[m+1]=searched; + }else{ + ret=_bisect_forward_serialno(vf,next,vf->offset, + end,ogg_page_serialno(&og),m+1); + if(ret==OV_EREAD)return(OV_EREAD); + } + + vf->offsets[m]=begin; + vf->serialnos[m]=currentno; + return(0); +} + +/* uses the local ogg_stream storage in vf; this is important for + non-streaming input sources */ +static int _fetch_headers(OggVorbis_File *vf,vorbis_info *vi,vorbis_comment *vc, + long *serialno,ogg_page *og_ptr){ + ogg_page og; + ogg_packet op; + int i,ret; + + if(!og_ptr){ + ogg_int64_t llret=_get_next_page(vf,&og,CHUNKSIZE); + if(llret==OV_EREAD)return(OV_EREAD); + if(llret<0)return OV_ENOTVORBIS; + og_ptr=&og; + } + + ogg_stream_reset_serialno(&vf->os,ogg_page_serialno(og_ptr)); + if(serialno)*serialno=vf->os.serialno; + vf->ready_state=STREAMSET; + + /* extract the initial header from the first page and verify that the + Ogg bitstream is in fact Vorbis data */ + + vorbis_info_init(vi); + vorbis_comment_init(vc); + + i=0; + while(i<3){ + ogg_stream_pagein(&vf->os,og_ptr); + while(i<3){ + int result=ogg_stream_packetout(&vf->os,&op); + if(result==0)break; + if(result==-1){ + ret=OV_EBADHEADER; + goto bail_header; + } + if((ret=vorbis_synthesis_headerin(vi,vc,&op))){ + goto bail_header; + } + i++; + } + if(i<3) + if(_get_next_page(vf,og_ptr,CHUNKSIZE)<0){ + ret=OV_EBADHEADER; + goto bail_header; + } + } + return 0; + + bail_header: + vorbis_info_clear(vi); + vorbis_comment_clear(vc); + vf->ready_state=OPENED; + + return ret; +} + +/* last step of the OggVorbis_File initialization; get all the + vorbis_info structs and PCM positions. Only called by the seekable + initialization (local stream storage is hacked slightly; pay + attention to how that's done) */ + +/* this is void and does not propogate errors up because we want to be + able to open and use damaged bitstreams as well as we can. Just + watch out for missing information for links in the OggVorbis_File + struct */ +static void _prefetch_all_headers(OggVorbis_File *vf, ogg_int64_t dataoffset){ + ogg_page og; + int i; + ogg_int64_t ret; + + vf->vi=_ogg_realloc(vf->vi,vf->links*sizeof(*vf->vi)); + vf->vc=_ogg_realloc(vf->vc,vf->links*sizeof(*vf->vc)); + vf->dataoffsets=_ogg_malloc(vf->links*sizeof(*vf->dataoffsets)); + vf->pcmlengths=_ogg_malloc(vf->links*2*sizeof(*vf->pcmlengths)); + + for(i=0;ilinks;i++){ + if(i==0){ + /* we already grabbed the initial header earlier. Just set the offset */ + vf->dataoffsets[i]=dataoffset; + _seek_helper(vf,dataoffset); + + }else{ + + /* seek to the location of the initial header */ + + _seek_helper(vf,vf->offsets[i]); + if(_fetch_headers(vf,vf->vi+i,vf->vc+i,NULL,NULL)<0){ + vf->dataoffsets[i]=-1; + }else{ + vf->dataoffsets[i]=vf->offset; + } + } + + /* fetch beginning PCM offset */ + + if(vf->dataoffsets[i]!=-1){ + ogg_int64_t accumulated=0; + long lastblock=-1; + int result; + + ogg_stream_reset_serialno(&vf->os,vf->serialnos[i]); + + while(1){ + ogg_packet op; + + ret=_get_next_page(vf,&og,-1); + if(ret<0) + /* this should not be possible unless the file is + truncated/mangled */ + break; + + if(ogg_page_serialno(&og)!=vf->serialnos[i]) + break; + + /* count blocksizes of all frames in the page */ + ogg_stream_pagein(&vf->os,&og); + while((result=ogg_stream_packetout(&vf->os,&op))){ + if(result>0){ /* ignore holes */ + long thisblock=vorbis_packet_blocksize(vf->vi+i,&op); + if(lastblock!=-1) + accumulated+=(lastblock+thisblock)>>2; + lastblock=thisblock; + } + } + + if(ogg_page_granulepos(&og)!=-1){ + /* pcm offset of last packet on the first audio page */ + accumulated= ogg_page_granulepos(&og)-accumulated; + break; + } + } + + /* less than zero? This is a stream with samples trimmed off + the beginning, a normal occurrence; set the offset to zero */ + if(accumulated<0)accumulated=0; + + vf->pcmlengths[i*2]=accumulated; + } + + /* get the PCM length of this link. To do this, + get the last page of the stream */ + { + ogg_int64_t end=vf->offsets[i+1]; + _seek_helper(vf,end); + + while(1){ + ret=_get_prev_page(vf,&og); + if(ret<0){ + /* this should not be possible */ + vorbis_info_clear(vf->vi+i); + vorbis_comment_clear(vf->vc+i); + break; + } + if(ogg_page_granulepos(&og)!=-1){ + vf->pcmlengths[i*2+1]=ogg_page_granulepos(&og)-vf->pcmlengths[i*2]; + break; + } + vf->offset=ret; + } + } + } +} + +static void _make_decode_ready(OggVorbis_File *vf){ + if(vf->ready_state!=STREAMSET)return; + if(vf->seekable){ + vorbis_synthesis_init(&vf->vd,vf->vi+vf->current_link); + }else{ + vorbis_synthesis_init(&vf->vd,vf->vi); + } + vorbis_block_init(&vf->vd,&vf->vb); + vf->ready_state=INITSET; + return; +} + +static int _open_seekable2(OggVorbis_File *vf){ + long serialno=vf->current_serialno; + ogg_int64_t dataoffset=vf->offset, end; + ogg_page og; + + /* we're partially open and have a first link header state in + storage in vf */ + /* we can seek, so set out learning all about this file */ + (vf->callbacks.seek_func)(vf->datasource,0,SEEK_END); + vf->offset=vf->end=(vf->callbacks.tell_func)(vf->datasource); + + /* We get the offset for the last page of the physical bitstream. + Most OggVorbis files will contain a single logical bitstream */ + end=_get_prev_page(vf,&og); + if(end<0)return(end); + + /* more than one logical bitstream? */ + if(ogg_page_serialno(&og)!=serialno){ + + /* Chained bitstream. Bisect-search each logical bitstream + section. Do so based on serial number only */ + if(_bisect_forward_serialno(vf,0,0,end+1,serialno,0)<0)return(OV_EREAD); + + }else{ + + /* Only one logical bitstream */ + if(_bisect_forward_serialno(vf,0,end,end+1,serialno,0))return(OV_EREAD); + + } + + /* the initial header memory is referenced by vf after; don't free it */ + _prefetch_all_headers(vf,dataoffset); + return(ov_raw_seek(vf,0)); +} + +/* clear out the current logical bitstream decoder */ +static void _decode_clear(OggVorbis_File *vf){ + vorbis_dsp_clear(&vf->vd); + vorbis_block_clear(&vf->vb); + vf->ready_state=OPENED; + + vf->bittrack=0.f; + vf->samptrack=0.f; +} + +/* fetch and process a packet. Handles the case where we're at a + bitstream boundary and dumps the decoding machine. If the decoding + machine is unloaded, it loads it. It also keeps pcm_offset up to + date (seek and read both use this. seek uses a special hack with + readp). + + return: <0) error, OV_HOLE (lost packet) or OV_EOF + 0) need more data (only if readp==0) + 1) got a packet +*/ + +static int _fetch_and_process_packet(OggVorbis_File *vf, + ogg_packet *op_in, + int readp){ + ogg_page og; + + /* handle one packet. Try to fetch it from current stream state */ + /* extract packets from page */ + while(1){ + + /* process a packet if we can. If the machine isn't loaded, + neither is a page */ + if(vf->ready_state==INITSET){ + while(1) { + ogg_packet op; + ogg_packet *op_ptr=(op_in?op_in:&op); + int result=ogg_stream_packetout(&vf->os,op_ptr); + ogg_int64_t granulepos; + + op_in=NULL; + if(result==-1)return(OV_HOLE); /* hole in the data. */ + if(result>0){ + /* got a packet. process it */ + granulepos=op_ptr->granulepos; + if(!vorbis_synthesis(&vf->vb,op_ptr)){ /* lazy check for lazy + header handling. The + header packets aren't + audio, so if/when we + submit them, + vorbis_synthesis will + reject them */ + + /* suck in the synthesis data and track bitrate */ + { + int oldsamples=vorbis_synthesis_pcmout(&vf->vd,NULL); + /* for proper use of libvorbis within libvorbisfile, + oldsamples will always be zero. */ + if(oldsamples)return(OV_EFAULT); + + vorbis_synthesis_blockin(&vf->vd,&vf->vb); + vf->samptrack+=vorbis_synthesis_pcmout(&vf->vd,NULL)-oldsamples; + vf->bittrack+=op_ptr->bytes*8; + } + + /* update the pcm offset. */ + if(granulepos!=-1 && !op_ptr->e_o_s){ + int link=(vf->seekable?vf->current_link:0); + int i,samples; + + /* this packet has a pcm_offset on it (the last packet + completed on a page carries the offset) After processing + (above), we know the pcm position of the *last* sample + ready to be returned. Find the offset of the *first* + + As an aside, this trick is inaccurate if we begin + reading anew right at the last page; the end-of-stream + granulepos declares the last frame in the stream, and the + last packet of the last page may be a partial frame. + So, we need a previous granulepos from an in-sequence page + to have a reference point. Thus the !op_ptr->e_o_s clause + above */ + + if(vf->seekable && link>0) + granulepos-=vf->pcmlengths[link*2]; + if(granulepos<0)granulepos=0; /* actually, this + shouldn't be possible + here unless the stream + is very broken */ + + samples=vorbis_synthesis_pcmout(&vf->vd,NULL); + + granulepos-=samples; + for(i=0;ipcmlengths[i*2+1]; + vf->pcm_offset=granulepos; + } + return(1); + } + } + else + break; + } + } + + if(vf->ready_state>=OPENED){ + if(!readp)return(0); + if(_get_next_page(vf,&og,-1)<0)return(OV_EOF); /* eof. + leave unitialized */ + /* bitrate tracking; add the header's bytes here, the body bytes + are done by packet above */ + vf->bittrack+=og.header_len*8; + + /* has our decoding just traversed a bitstream boundary? */ + if(vf->ready_state==INITSET){ + if(vf->current_serialno!=ogg_page_serialno(&og)){ + _decode_clear(vf); + + if(!vf->seekable){ + vorbis_info_clear(vf->vi); + vorbis_comment_clear(vf->vc); + } + } + } + } + + /* Do we need to load a new machine before submitting the page? */ + /* This is different in the seekable and non-seekable cases. + + In the seekable case, we already have all the header + information loaded and cached; we just initialize the machine + with it and continue on our merry way. + + In the non-seekable (streaming) case, we'll only be at a + boundary if we just left the previous logical bitstream and + we're now nominally at the header of the next bitstream + */ + + if(vf->ready_state!=INITSET){ + int link; + + if(vf->ready_stateseekable){ + vf->current_serialno=ogg_page_serialno(&og); + + /* match the serialno to bitstream section. We use this rather than + offset positions to avoid problems near logical bitstream + boundaries */ + for(link=0;linklinks;link++) + if(vf->serialnos[link]==vf->current_serialno)break; + if(link==vf->links)return(OV_EBADLINK); /* sign of a bogus + stream. error out, + leave machine + uninitialized */ + + vf->current_link=link; + + ogg_stream_reset_serialno(&vf->os,vf->current_serialno); + vf->ready_state=STREAMSET; + + }else{ + /* we're streaming */ + /* fetch the three header packets, build the info struct */ + + int ret=_fetch_headers(vf,vf->vi,vf->vc,&vf->current_serialno,&og); + if(ret)return(ret); + vf->current_link++; + link=0; + } + } + + _make_decode_ready(vf); + } + ogg_stream_pagein(&vf->os,&og); + } +} + +/* if, eg, 64 bit stdio is configured by default, this will build with + fseek64 */ +static int _fseek64_wrap(FILE *f,ogg_int64_t off,int whence){ + if(f==NULL)return(-1); + return fseek(f,off,whence); +} + +static int _ov_open1(void *f,OggVorbis_File *vf,char *initial, + long ibytes, ov_callbacks callbacks){ + int offsettest=(f?callbacks.seek_func(f,0,SEEK_CUR):-1); + int ret; + + memset(vf,0,sizeof(*vf)); + vf->datasource=f; + vf->callbacks = callbacks; + + /* init the framing state */ + ogg_sync_init(&vf->oy); + + /* perhaps some data was previously read into a buffer for testing + against other stream types. Allow initialization from this + previously read data (as we may be reading from a non-seekable + stream) */ + if(initial){ + char *buffer=ogg_sync_buffer(&vf->oy,ibytes); + memcpy(buffer,initial,ibytes); + ogg_sync_wrote(&vf->oy,ibytes); + } + + /* can we seek? Stevens suggests the seek test was portable */ + if(offsettest!=-1)vf->seekable=1; + + /* No seeking yet; Set up a 'single' (current) logical bitstream + entry for partial open */ + vf->links=1; + vf->vi=_ogg_calloc(vf->links,sizeof(*vf->vi)); + vf->vc=_ogg_calloc(vf->links,sizeof(*vf->vc)); + ogg_stream_init(&vf->os,-1); /* fill in the serialno later */ + + /* Try to fetch the headers, maintaining all the storage */ + if((ret=_fetch_headers(vf,vf->vi,vf->vc,&vf->current_serialno,NULL))<0){ + vf->datasource=NULL; + ov_clear(vf); + }else if(vf->ready_state < PARTOPEN) + vf->ready_state=PARTOPEN; + return(ret); +} + +static int _ov_open2(OggVorbis_File *vf){ + if(vf->ready_state < OPENED) + vf->ready_state=OPENED; + if(vf->seekable){ + int ret=_open_seekable2(vf); + if(ret){ + vf->datasource=NULL; + ov_clear(vf); + } + return(ret); + } + return 0; +} + + +/* clear out the OggVorbis_File struct */ +int ov_clear(OggVorbis_File *vf){ + if(vf){ + vorbis_block_clear(&vf->vb); + vorbis_dsp_clear(&vf->vd); + ogg_stream_clear(&vf->os); + + if(vf->vi && vf->links){ + int i; + for(i=0;ilinks;i++){ + vorbis_info_clear(vf->vi+i); + vorbis_comment_clear(vf->vc+i); + } + _ogg_free(vf->vi); + _ogg_free(vf->vc); + } + if(vf->dataoffsets)_ogg_free(vf->dataoffsets); + if(vf->pcmlengths)_ogg_free(vf->pcmlengths); + if(vf->serialnos)_ogg_free(vf->serialnos); + if(vf->offsets)_ogg_free(vf->offsets); + ogg_sync_clear(&vf->oy); + if(vf->datasource)(vf->callbacks.close_func)(vf->datasource); + memset(vf,0,sizeof(*vf)); + } +#ifdef DEBUG_LEAKS + _VDBG_dump(); +#endif + return(0); +} + +/* inspects the OggVorbis file and finds/documents all the logical + bitstreams contained in it. Tries to be tolerant of logical + bitstream sections that are truncated/woogie. + + return: -1) error + 0) OK +*/ + +int ov_open_callbacks(void *f,OggVorbis_File *vf,char *initial,long ibytes, + ov_callbacks callbacks){ + int ret=_ov_open1(f,vf,initial,ibytes,callbacks); + if(ret)return ret; + return _ov_open2(vf); +} + +int ov_open(FILE *f,OggVorbis_File *vf,char *initial,long ibytes){ + ov_callbacks callbacks = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) _fseek64_wrap, + (int (*)(void *)) fclose, + (long (*)(void *)) ftell + }; + + return ov_open_callbacks((void *)f, vf, initial, ibytes, callbacks); +} + +/* Only partially open the vorbis file; test for Vorbisness, and load + the headers for the first chain. Do not seek (although test for + seekability). Use ov_test_open to finish opening the file, else + ov_clear to close/free it. Same return codes as open. */ + +int ov_test_callbacks(void *f,OggVorbis_File *vf,char *initial,long ibytes, + ov_callbacks callbacks) +{ + return _ov_open1(f,vf,initial,ibytes,callbacks); +} + +int ov_test(FILE *f,OggVorbis_File *vf,char *initial,long ibytes){ + ov_callbacks callbacks = { + (size_t (*)(void *, size_t, size_t, void *)) fread, + (int (*)(void *, ogg_int64_t, int)) _fseek64_wrap, + (int (*)(void *)) fclose, + (long (*)(void *)) ftell + }; + + return ov_test_callbacks((void *)f, vf, initial, ibytes, callbacks); +} + +int ov_test_open(OggVorbis_File *vf){ + if(vf->ready_state!=PARTOPEN)return(OV_EINVAL); + return _ov_open2(vf); +} + +/* How many logical bitstreams in this physical bitstream? */ +long ov_streams(OggVorbis_File *vf){ + return vf->links; +} + +/* Is the FILE * associated with vf seekable? */ +long ov_seekable(OggVorbis_File *vf){ + return vf->seekable; +} + +/* returns the bitrate for a given logical bitstream or the entire + physical bitstream. If the file is open for random access, it will + find the *actual* average bitrate. If the file is streaming, it + returns the nominal bitrate (if set) else the average of the + upper/lower bounds (if set) else -1 (unset). + + If you want the actual bitrate field settings, get them from the + vorbis_info structs */ + +long ov_bitrate(OggVorbis_File *vf,int i){ + if(vf->ready_state=vf->links)return(OV_EINVAL); + if(!vf->seekable && i!=0)return(ov_bitrate(vf,0)); + if(i<0){ + ogg_int64_t bits=0; + int i; + for(i=0;ilinks;i++) + bits+=(vf->offsets[i+1]-vf->dataoffsets[i])*8; + return(rint(bits/ov_time_total(vf,-1))); + }else{ + if(vf->seekable){ + /* return the actual bitrate */ + return(rint((vf->offsets[i+1]-vf->dataoffsets[i])*8/ov_time_total(vf,i))); + }else{ + /* return nominal if set */ + if(vf->vi[i].bitrate_nominal>0){ + return vf->vi[i].bitrate_nominal; + }else{ + if(vf->vi[i].bitrate_upper>0){ + if(vf->vi[i].bitrate_lower>0){ + return (vf->vi[i].bitrate_upper+vf->vi[i].bitrate_lower)/2; + }else{ + return vf->vi[i].bitrate_upper; + } + } + return(OV_FALSE); + } + } + } +} + +/* returns the actual bitrate since last call. returns -1 if no + additional data to offer since last call (or at beginning of stream), + EINVAL if stream is only partially open +*/ +long ov_bitrate_instant(OggVorbis_File *vf){ + int link=(vf->seekable?vf->current_link:0); + long ret; + if(vf->ready_statesamptrack==0)return(OV_FALSE); + ret=vf->bittrack/vf->samptrack*vf->vi[link].rate+.5; + vf->bittrack=0.f; + vf->samptrack=0.f; + return(ret); +} + +/* Guess */ +long ov_serialnumber(OggVorbis_File *vf,int i){ + if(i>=vf->links)return(ov_serialnumber(vf,vf->links-1)); + if(!vf->seekable && i>=0)return(ov_serialnumber(vf,-1)); + if(i<0){ + return(vf->current_serialno); + }else{ + return(vf->serialnos[i]); + } +} + +/* returns: total raw (compressed) length of content if i==-1 + raw (compressed) length of that logical bitstream for i==0 to n + OV_EINVAL if the stream is not seekable (we can't know the length) + or if stream is only partially open +*/ +ogg_int64_t ov_raw_total(OggVorbis_File *vf,int i){ + if(vf->ready_stateseekable || i>=vf->links)return(OV_EINVAL); + if(i<0){ + ogg_int64_t acc=0; + int i; + for(i=0;ilinks;i++) + acc+=ov_raw_total(vf,i); + return(acc); + }else{ + return(vf->offsets[i+1]-vf->offsets[i]); + } +} + +/* returns: total PCM length (samples) of content if i==-1 PCM length + (samples) of that logical bitstream for i==0 to n + OV_EINVAL if the stream is not seekable (we can't know the + length) or only partially open +*/ +ogg_int64_t ov_pcm_total(OggVorbis_File *vf,int i){ + if(vf->ready_stateseekable || i>=vf->links)return(OV_EINVAL); + if(i<0){ + ogg_int64_t acc=0; + int i; + for(i=0;ilinks;i++) + acc+=ov_pcm_total(vf,i); + return(acc); + }else{ + return(vf->pcmlengths[i*2+1]); + } +} + +/* returns: total seconds of content if i==-1 + seconds in that logical bitstream for i==0 to n + OV_EINVAL if the stream is not seekable (we can't know the + length) or only partially open +*/ +double ov_time_total(OggVorbis_File *vf,int i){ + if(vf->ready_stateseekable || i>=vf->links)return(OV_EINVAL); + if(i<0){ + double acc=0; + int i; + for(i=0;ilinks;i++) + acc+=ov_time_total(vf,i); + return(acc); + }else{ + return((double)(vf->pcmlengths[i*2+1])/vf->vi[i].rate); + } +} + +/* seek to an offset relative to the *compressed* data. This also + scans packets to update the PCM cursor. It will cross a logical + bitstream boundary, but only if it can't get any packets out of the + tail of the bitstream we seek to (so no surprises). + + returns zero on success, nonzero on failure */ + +int ov_raw_seek(OggVorbis_File *vf,ogg_int64_t pos){ + ogg_stream_state work_os; + + if(vf->ready_stateseekable) + return(OV_ENOSEEK); /* don't dump machine if we can't seek */ + + if(pos<0 || pos>vf->end)return(OV_EINVAL); + + /* clear out decoding machine state */ + vf->pcm_offset=-1; + _decode_clear(vf); + + _seek_helper(vf,pos); + + /* we need to make sure the pcm_offset is set, but we don't want to + advance the raw cursor past good packets just to get to the first + with a granulepos. That's not equivalent behavior to beginning + decoding as immediately after the seek position as possible. + + So, a hack. We use two stream states; a local scratch state and + a the shared vf->os stream state. We use the local state to + scan, and the shared state as a buffer for later decode. + + Unfortuantely, on the last page we still advance to last packet + because the granulepos on the last page is not necessarily on a + packet boundary, and we need to make sure the granpos is + correct. + */ + + { + ogg_page og; + ogg_packet op; + int lastblock=0; + int accblock=0; + int thisblock; + int eosflag; + + ogg_stream_init(&work_os,-1); /* get the memory ready */ + + while(1){ + if(vf->ready_state==STREAMSET){ + /* snarf/scan a packet if we can */ + int result=ogg_stream_packetout(&work_os,&op); + + if(result>0){ + + if(vf->vi[vf->current_link].codec_setup) + thisblock=vorbis_packet_blocksize(vf->vi+vf->current_link,&op); + if(eosflag) + ogg_stream_packetout(&vf->os,NULL); + else + if(lastblock)accblock+=(lastblock+thisblock)>>2; + + if(op.granulepos!=-1){ + int i,link=vf->current_link; + ogg_int64_t granulepos=op.granulepos-vf->pcmlengths[link*2]; + if(granulepos<0)granulepos=0; + + for(i=0;ipcmlengths[i*2+1]; + vf->pcm_offset=granulepos-accblock; + break; + } + lastblock=thisblock; + continue; + } + } + + if(!lastblock){ + if(_get_next_page(vf,&og,-1)<0){ + vf->pcm_offset=ov_pcm_total(vf,-1); + break; + } + }else{ + /* huh? Bogus stream with packets but no granulepos */ + vf->pcm_offset=-1; + break; + } + + /* has our decoding just traversed a bitstream boundary? */ + if(vf->ready_state==STREAMSET) + if(vf->current_serialno!=ogg_page_serialno(&og)){ + _decode_clear(vf); /* clear out stream state */ + ogg_stream_clear(&work_os); + } + + if(vf->ready_statecurrent_serialno=ogg_page_serialno(&og); + for(link=0;linklinks;link++) + if(vf->serialnos[link]==vf->current_serialno)break; + if(link==vf->links)goto seek_error; /* sign of a bogus stream. + error out, leave + machine uninitialized */ + vf->current_link=link; + + ogg_stream_reset_serialno(&vf->os,vf->current_serialno); + ogg_stream_reset_serialno(&work_os,vf->current_serialno); + vf->ready_state=STREAMSET; + + } + + ogg_stream_pagein(&vf->os,&og); + ogg_stream_pagein(&work_os,&og); + eosflag=ogg_page_eos(&og); + } + } + + ogg_stream_clear(&work_os); + return(0); + + seek_error: + /* dump the machine so we're in a known state */ + vf->pcm_offset=-1; + ogg_stream_clear(&work_os); + _decode_clear(vf); + return OV_EBADLINK; +} + +/* Page granularity seek (faster than sample granularity because we + don't do the last bit of decode to find a specific sample). + + Seek to the last [granule marked] page preceeding the specified pos + location, such that decoding past the returned point will quickly + arrive at the requested position. */ +int ov_pcm_seek_page(OggVorbis_File *vf,ogg_int64_t pos){ + int link=-1; + ogg_int64_t result=0; + ogg_int64_t total=ov_pcm_total(vf,-1); + + if(vf->ready_stateseekable)return(OV_ENOSEEK); + + if(pos<0 || pos>total)return(OV_EINVAL); + + /* which bitstream section does this pcm offset occur in? */ + for(link=vf->links-1;link>=0;link--){ + total-=vf->pcmlengths[link*2+1]; + if(pos>=total)break; + } + + /* search within the logical bitstream for the page with the highest + pcm_pos preceeding (or equal to) pos. There is a danger here; + missing pages or incorrect frame number information in the + bitstream could make our task impossible. Account for that (it + would be an error condition) */ + + /* new search algorithm by HB (Nicholas Vinen) */ + { + ogg_int64_t end=vf->offsets[link+1]; + ogg_int64_t begin=vf->offsets[link]; + ogg_int64_t begintime = vf->pcmlengths[link*2]; + ogg_int64_t endtime = vf->pcmlengths[link*2+1]+begintime; + ogg_int64_t target=pos-total+begintime; + ogg_int64_t best=begin; + + ogg_page og; + while(beginoffset); + if(result==OV_EREAD) goto seek_error; + if(result<0){ + if(bisect<=begin+1) + end=begin; /* found it */ + else{ + if(bisect==0) goto seek_error; + bisect-=CHUNKSIZE; + if(bisect<=begin)bisect=begin+1; + _seek_helper(vf,bisect); + } + }else{ + ogg_int64_t granulepos=ogg_page_granulepos(&og); + if(granulepos==-1)continue; + if(granuleposoffset; /* raw offset of next page */ + begintime=granulepos; + + if(target-begintime>44100)break; + bisect=begin; /* *not* begin + 1 */ + }else{ + if(bisect<=begin+1) + end=begin; /* found it */ + else{ + if(end==vf->offset){ /* we're pretty close - we'd be stuck in */ + end=result; + bisect-=CHUNKSIZE; /* an endless loop otherwise. */ + if(bisect<=begin)bisect=begin+1; + _seek_helper(vf,bisect); + }else{ + end=result; + endtime=granulepos; + break; + } + } + } + } + } + } + + /* found our page. seek to it, update pcm offset. Easier case than + raw_seek, don't keep packets preceeding granulepos. */ + { + ogg_page og; + ogg_packet op; + /* clear out decoding machine state */ + _decode_clear(vf); + /* seek */ + _seek_helper(vf,best); + + if(_get_next_page(vf,&og,-1)<0)return(OV_EOF); /* shouldn't happen */ + vf->current_serialno=ogg_page_serialno(&og); + vf->current_link=link; + + ogg_stream_reset_serialno(&vf->os,vf->current_serialno); + vf->ready_state=STREAMSET; + ogg_stream_pagein(&vf->os,&og); + + /* pull out all but last packet; the one with granulepos */ + while(1){ + result=ogg_stream_packetpeek(&vf->os,&op); + if(result==0){ + /* !!! the packet finishing this page originated on a + preceeding page. Keep fetching previous pages until we + get one with a granulepos or without the 'continued' flag + set. Then just use raw_seek for simplicity. */ + + _decode_clear(vf); + _seek_helper(vf,best); + + while(1){ + result=_get_prev_page(vf,&og); + if(result<0) goto seek_error; + if(ogg_page_granulepos(&og)>-1 || + !ogg_page_continued(&og)){ + return ov_raw_seek(vf,result); + } + vf->offset=result; + } + } + if(result<0){ + result = OV_EBADPACKET; + goto seek_error; + } + if(op.granulepos!=-1){ + vf->pcm_offset=op.granulepos-vf->pcmlengths[vf->current_link*2]; + if(vf->pcm_offset<0)vf->pcm_offset=0; + vf->pcm_offset+=total; + break; + }else + result=ogg_stream_packetout(&vf->os,NULL); + } + } + } + + /* verify result */ + if(vf->pcm_offset>pos || pos>ov_pcm_total(vf,-1)){ + result=OV_EFAULT; + goto seek_error; + } + return(0); + + seek_error: + /* dump machine so we're in a known state */ + vf->pcm_offset=-1; + _decode_clear(vf); + return (int)result; +} + +/* seek to a sample offset relative to the decompressed pcm stream + returns zero on success, nonzero on failure */ + +int ov_pcm_seek(OggVorbis_File *vf,ogg_int64_t pos){ + int thisblock,lastblock=0; + int ret=ov_pcm_seek_page(vf,pos); + if(ret<0)return(ret); + _make_decode_ready(vf); + + /* discard leading packets we don't need for the lapping of the + position we want; don't decode them */ + + while(1){ + ogg_packet op; + ogg_page og; + + int ret=ogg_stream_packetpeek(&vf->os,&op); + if(ret>0){ + thisblock=vorbis_packet_blocksize(vf->vi+vf->current_link,&op); + if(thisblock<0)thisblock=0; /* non audio packet */ + if(lastblock)vf->pcm_offset+=(lastblock+thisblock)>>2; + + if(vf->pcm_offset+((thisblock+ + vorbis_info_blocksize(vf->vi,1))>>2)>=pos)break; + + /* remove the packet from packet queue and track its granulepos */ + ogg_stream_packetout(&vf->os,NULL); + vorbis_synthesis_trackonly(&vf->vb,&op); /* set up a vb with + only tracking, no + pcm_decode */ + vorbis_synthesis_blockin(&vf->vd,&vf->vb); + + /* end of logical stream case is hard, especially with exact + length positioning. */ + + if(op.granulepos>-1){ + int i; + /* always believe the stream markers */ + vf->pcm_offset=op.granulepos-vf->pcmlengths[vf->current_link*2]; + if(vf->pcm_offset<0)vf->pcm_offset=0; + for(i=0;icurrent_link;i++) + vf->pcm_offset+=vf->pcmlengths[i*2+1]; + } + + lastblock=thisblock; + + }else{ + if(ret<0 && ret!=OV_HOLE)break; + + /* suck in a new page */ + if(_get_next_page(vf,&og,-1)<0)break; + if(vf->current_serialno!=ogg_page_serialno(&og))_decode_clear(vf); + + if(vf->ready_statecurrent_serialno=ogg_page_serialno(&og); + for(link=0;linklinks;link++) + if(vf->serialnos[link]==vf->current_serialno)break; + if(link==vf->links)return(OV_EBADLINK); + vf->current_link=link; + + ogg_stream_reset_serialno(&vf->os,vf->current_serialno); + vf->ready_state=STREAMSET; + _make_decode_ready(vf); + lastblock=0; + } + + ogg_stream_pagein(&vf->os,&og); + } + } + + /* discard samples until we reach the desired position. Crossing a + logical bitstream boundary with abandon is OK. */ + while(vf->pcm_offsetpcm_offset; + long samples=vorbis_synthesis_pcmout(&vf->vd,&pcm); + + if(samples>target)samples=target; + vorbis_synthesis_read(&vf->vd,samples); + vf->pcm_offset+=samples; + + if(samplespcm_offset=ov_pcm_total(vf,-1); /* eof */ + } + return 0; +} + +/* seek to a playback time relative to the decompressed pcm stream + returns zero on success, nonzero on failure */ +int ov_time_seek(OggVorbis_File *vf,double seconds){ + /* translate time to PCM position and call ov_pcm_seek */ + + int link=-1; + ogg_int64_t pcm_total=ov_pcm_total(vf,-1); + double time_total=ov_time_total(vf,-1); + + if(vf->ready_stateseekable)return(OV_ENOSEEK); + if(seconds<0 || seconds>time_total)return(OV_EINVAL); + + /* which bitstream section does this time offset occur in? */ + for(link=vf->links-1;link>=0;link--){ + pcm_total-=vf->pcmlengths[link*2+1]; + time_total-=ov_time_total(vf,link); + if(seconds>=time_total)break; + } + + /* enough information to convert time offset to pcm offset */ + { + ogg_int64_t target=pcm_total+(seconds-time_total)*vf->vi[link].rate; + return(ov_pcm_seek(vf,target)); + } +} + +/* page-granularity version of ov_time_seek + returns zero on success, nonzero on failure */ +int ov_time_seek_page(OggVorbis_File *vf,double seconds){ + /* translate time to PCM position and call ov_pcm_seek */ + + int link=-1; + ogg_int64_t pcm_total=ov_pcm_total(vf,-1); + double time_total=ov_time_total(vf,-1); + + if(vf->ready_stateseekable)return(OV_ENOSEEK); + if(seconds<0 || seconds>time_total)return(OV_EINVAL); + + /* which bitstream section does this time offset occur in? */ + for(link=vf->links-1;link>=0;link--){ + pcm_total-=vf->pcmlengths[link*2+1]; + time_total-=ov_time_total(vf,link); + if(seconds>=time_total)break; + } + + /* enough information to convert time offset to pcm offset */ + { + ogg_int64_t target=pcm_total+(seconds-time_total)*vf->vi[link].rate; + return(ov_pcm_seek_page(vf,target)); + } +} + +/* tell the current stream offset cursor. Note that seek followed by + tell will likely not give the set offset due to caching */ +ogg_int64_t ov_raw_tell(OggVorbis_File *vf){ + if(vf->ready_stateoffset); +} + +/* return PCM offset (sample) of next PCM sample to be read */ +ogg_int64_t ov_pcm_tell(OggVorbis_File *vf){ + if(vf->ready_statepcm_offset); +} + +/* return time offset (seconds) of next PCM sample to be read */ +double ov_time_tell(OggVorbis_File *vf){ + /* translate time to PCM position and call ov_pcm_seek */ + + int link=-1; + ogg_int64_t pcm_total=0; + double time_total=0.f; + + if(vf->ready_stateseekable){ + pcm_total=ov_pcm_total(vf,-1); + time_total=ov_time_total(vf,-1); + + /* which bitstream section does this time offset occur in? */ + for(link=vf->links-1;link>=0;link--){ + pcm_total-=vf->pcmlengths[link*2+1]; + time_total-=ov_time_total(vf,link); + if(vf->pcm_offset>=pcm_total)break; + } + } + + return((double)time_total+(double)(vf->pcm_offset-pcm_total)/vf->vi[link].rate); +} + +/* link: -1) return the vorbis_info struct for the bitstream section + currently being decoded + 0-n) to request information for a specific bitstream section + + In the case of a non-seekable bitstream, any call returns the + current bitstream. NULL in the case that the machine is not + initialized */ + +vorbis_info *ov_info(OggVorbis_File *vf,int link){ + if(vf->seekable){ + if(link<0) + if(vf->ready_state>=STREAMSET) + return vf->vi+vf->current_link; + else + return vf->vi; + else + if(link>=vf->links) + return NULL; + else + return vf->vi+link; + }else{ + return vf->vi; + } +} + +/* grr, strong typing, grr, no templates/inheritence, grr */ +vorbis_comment *ov_comment(OggVorbis_File *vf,int link){ + if(vf->seekable){ + if(link<0) + if(vf->ready_state>=STREAMSET) + return vf->vc+vf->current_link; + else + return vf->vc; + else + if(link>=vf->links) + return NULL; + else + return vf->vc+link; + }else{ + return vf->vc; + } +} + +static int host_is_big_endian() { + ogg_int32_t pattern = 0xfeedface; /* deadbeef */ + unsigned char *bytewise = (unsigned char *)&pattern; + if (bytewise[0] == 0xfe) return 1; + return 0; +} + +/* up to this point, everything could more or less hide the multiple + logical bitstream nature of chaining from the toplevel application + if the toplevel application didn't particularly care. However, at + the point that we actually read audio back, the multiple-section + nature must surface: Multiple bitstream sections do not necessarily + have to have the same number of channels or sampling rate. + + ov_read returns the sequential logical bitstream number currently + being decoded along with the PCM data in order that the toplevel + application can take action on channel/sample rate changes. This + number will be incremented even for streamed (non-seekable) streams + (for seekable streams, it represents the actual logical bitstream + index within the physical bitstream. Note that the accessor + functions above are aware of this dichotomy). + + input values: buffer) a buffer to hold packed PCM data for return + length) the byte length requested to be placed into buffer + bigendianp) should the data be packed LSB first (0) or + MSB first (1) + word) word size for output. currently 1 (byte) or + 2 (16 bit short) + + return values: <0) error/hole in data (OV_HOLE), partial open (OV_EINVAL) + 0) EOF + n) number of bytes of PCM actually returned. The + below works on a packet-by-packet basis, so the + return length is not related to the 'length' passed + in, just guaranteed to fit. + + *section) set to the logical bitstream number */ + +long ov_read(OggVorbis_File *vf,char *buffer,int length, + int bigendianp,int word,int sgned,int *bitstream){ + int i,j; + int host_endian = host_is_big_endian(); + + float **pcm; + long samples; + + if(vf->ready_stateready_state>=STREAMSET){ + samples=vorbis_synthesis_pcmout(&vf->vd,&pcm); + if(samples)break; + } + + /* suck in another packet */ + { + int ret=_fetch_and_process_packet(vf,NULL,1); + if(ret==OV_EOF)return(0); + if(ret<=0)return(ret); + } + + } + + if(samples>0){ + + /* yay! proceed to pack data into the byte buffer */ + + long channels=ov_info(vf,-1)->channels; + long bytespersample=word * channels; + vorbis_fpu_control fpu; + if(samples>length/bytespersample)samples=length/bytespersample; + + if(samples <= 0) + return OV_EINVAL; + + /* a tight loop to pack each size */ + { + int val; + if(word==1){ + int off=(sgned?0:128); + vorbis_fpu_setround(&fpu); + for(j=0;j127)val=127; + else if(val<-128)val=-128; + *buffer++=val+off; + } + vorbis_fpu_restore(fpu); + }else{ + int off=(sgned?0:32768); + + if(host_endian==bigendianp){ + if(sgned){ + + vorbis_fpu_setround(&fpu); + for(i=0;i32767)val=32767; + else if(val<-32768)val=-32768; + *dest=val; + dest+=channels; + } + } + vorbis_fpu_restore(fpu); + + }else{ + + vorbis_fpu_setround(&fpu); + for(i=0;i32767)val=32767; + else if(val<-32768)val=-32768; + *dest=val+off; + dest+=channels; + } + } + vorbis_fpu_restore(fpu); + + } + }else if(bigendianp){ + + vorbis_fpu_setround(&fpu); + for(j=0;j32767)val=32767; + else if(val<-32768)val=-32768; + val+=off; + *buffer++=(val>>8); + *buffer++=(val&0xff); + } + vorbis_fpu_restore(fpu); + + }else{ + int val; + vorbis_fpu_setround(&fpu); + for(j=0;j32767)val=32767; + else if(val<-32768)val=-32768; + val+=off; + *buffer++=(val&0xff); + *buffer++=(val>>8); + } + vorbis_fpu_restore(fpu); + + } + } + } + + vorbis_synthesis_read(&vf->vd,samples); + vf->pcm_offset+=samples; + if(bitstream)*bitstream=vf->current_link; + return(samples*bytespersample); + }else{ + return(samples); + } +} + +/* input values: pcm_channels) a float vector per channel of output + length) the sample length being read by the app + + return values: <0) error/hole in data (OV_HOLE), partial open (OV_EINVAL) + 0) EOF + n) number of samples of PCM actually returned. The + below works on a packet-by-packet basis, so the + return length is not related to the 'length' passed + in, just guaranteed to fit. + + *section) set to the logical bitstream number */ + + + +long ov_read_float(OggVorbis_File *vf,float ***pcm_channels,int length, + int *bitstream){ + + if(vf->ready_stateready_state>=STREAMSET){ + float **pcm; + long samples=vorbis_synthesis_pcmout(&vf->vd,&pcm); + if(samples){ + if(pcm_channels)*pcm_channels=pcm; + if(samples>length)samples=length; + vorbis_synthesis_read(&vf->vd,samples); + vf->pcm_offset+=samples; + if(bitstream)*bitstream=vf->current_link; + return samples; + + } + } + + /* suck in another packet */ + { + int ret=_fetch_and_process_packet(vf,NULL,1); + if(ret==OV_EOF)return(0); + if(ret<=0)return(ret); + } + + } +} + diff --git a/src/add-ons/media/plugins/vorbis/libvorbis/window.c b/src/add-ons/media/plugins/vorbis/libvorbis/window.c new file mode 100644 index 0000000000..bbe11ea8cf --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/libvorbis/window.c @@ -0,0 +1,79 @@ +/******************************************************************** + * * + * THIS FILE IS PART OF THE OggVorbis SOFTWARE 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 SOURCE CODE IS (C) COPYRIGHT 1994-2002 * + * by the XIPHOPHORUS Company http://www.xiph.org/ * + * * + ******************************************************************** + + function: window functions + last mod: $Id: window.c,v 1.1 2003/12/13 20:17:56 shatty Exp $ + + ********************************************************************/ + +#include +#include +#include "os.h" +#include "misc.h" + +float *_vorbis_window(int type, int left){ + float *ret=_ogg_calloc(left,sizeof(*ret)); + int i; + + switch(type){ + case 0: + /* The 'vorbis window' (window 0) is sin(sin(x)*sin(x)*2pi) */ + { + + for(i=0;i +#include +#include +#include +#include +#include "vorbisCodecPlugin.h" + +#define TRACE_THIS 1 +#if TRACE_THIS + #define TRACE printf +#else + #define TRACE(a...) +#endif + +#define DECODE_BUFFER_SIZE (32 * 1024) + +vorbisDecoder::vorbisDecoder() +{ + InitMP3(&fMpgLibPrivate); + fResidualBytes = 0; + fResidualBuffer = 0; + fDecodeBuffer = new uint8 [DECODE_BUFFER_SIZE]; + fStartTime = 0; + fFrameSize = 0; + fFrameRate = 0; + fBitRate = 0; + fChannelCount = 0; + fOutputBufferSize = 0; +} + + +vorbisDecoder::~vorbisDecoder() +{ + ExitMP3(&fMpgLibPrivate); + delete [] fDecodeBuffer; +} + + +status_t +vorbisDecoder::Setup(media_format *ioEncodedFormat, + const void *infoBuffer, int32 infoSize) +{ + // decode first chunk to initialize mpeg library + if (B_OK != DecodeNextChunk()) { + printf("vorbisDecoder::Setup failed, can't decode first chunk\n"); + return B_ERROR; + } + + // initialize fBitRate, fFrameRate and fChannelCount from mpg decode library values of first header + extern int tabsel_123[2][3][16]; + extern long freqs[9]; + fBitRate = tabsel_123[fMpgLibPrivate.fr.lsf][fMpgLibPrivate.fr.lay-1][fMpgLibPrivate.fr.bitrate_index] * 1000; + fFrameRate = freqs[fMpgLibPrivate.fr.sampling_frequency]; + fChannelCount = fMpgLibPrivate.fr.stereo; + + printf("vorbisDecoder::Setup: channels %d, bitrate %d, framerate %d\n", fChannelCount, fBitRate, fFrameRate); + + // put some more useful info into the media_format describing our input format + ioEncodedFormat->u.encoded_audio.bit_rate = fBitRate; + + return B_OK; +} + +status_t +vorbisDecoder::NegotiateOutputFormat(media_format *ioDecodedFormat) +{ + // fFrameRate and fChannelCount are already valid here + + // BeBook says: The codec will find and return in ioFormat its best matching format + // => This means, we never return an error, and always change the format values + // that we don't support to something more applicable + + ioDecodedFormat->type = B_MEDIA_RAW_AUDIO; + ioDecodedFormat->u.raw_audio.frame_rate = fFrameRate; + ioDecodedFormat->u.raw_audio.channel_count = fChannelCount; + ioDecodedFormat->u.raw_audio.format = media_raw_audio_format::B_AUDIO_SHORT; // XXX should support other formats, too + ioDecodedFormat->u.raw_audio.byte_order = B_MEDIA_HOST_ENDIAN; // XXX should support other endain, too + if (ioDecodedFormat->u.raw_audio.buffer_size < 512 || ioDecodedFormat->u.raw_audio.buffer_size > 65536) + ioDecodedFormat->u.raw_audio.buffer_size = BMediaRoster::Roster()->AudioBufferSizeFor( + fChannelCount, + ioDecodedFormat->u.raw_audio.format, + fFrameRate); + if (ioDecodedFormat->u.raw_audio.channel_mask == 0) + ioDecodedFormat->u.raw_audio.channel_mask = (fChannelCount == 1) ? B_CHANNEL_LEFT : B_CHANNEL_LEFT | B_CHANNEL_RIGHT; + + // setup rest of the needed variables + fFrameSize = (ioDecodedFormat->u.raw_audio.format & 0xf) * fChannelCount; + fOutputBufferSize = ioDecodedFormat->u.raw_audio.buffer_size; + + return B_OK; +} + + +status_t +vorbisDecoder::Seek(uint32 seekTo, + int64 seekFrame, int64 *frame, + bigtime_t seekTime, bigtime_t *time) +{ + ExitMP3(&fMpgLibPrivate); + InitMP3(&fMpgLibPrivate); + fResidualBytes = 0; + return B_OK; +} + + +status_t +vorbisDecoder::Decode(void *buffer, int64 *frameCount, + media_header *mediaHeader, media_decode_info *info /* = 0 */) +{ + uint8 * out_buffer = static_cast(buffer); + int32 out_bytes_needed = fOutputBufferSize; + + mediaHeader->start_time = fStartTime; + //TRACE("vorbisDecoder: Decoding start time %.6f\n", fStartTime / 1000000.0); + + while (out_bytes_needed > 0) { + if (fResidualBytes) { + int32 bytes = min_c(fResidualBytes, out_bytes_needed); + memcpy(out_buffer, fResidualBuffer, bytes); + fResidualBuffer += bytes; + fResidualBytes -= bytes; + out_buffer += bytes; + out_bytes_needed -= bytes; + + fStartTime += (1000000LL * (bytes / fFrameSize)) / fFrameRate; + + //TRACE("vorbisDecoder: fStartTime inc'd to %.6f\n", fStartTime / 1000000.0); + continue; + } + + if (B_OK != DecodeNextChunk()) + break; + } + + *frameCount = (fOutputBufferSize - out_bytes_needed) / fFrameSize; + + // XXX this doesn't guarantee that we always return B_LAST_BUFFER_ERROR bofore returning B_ERROR + return (out_bytes_needed == 0) ? B_OK : (out_bytes_needed == fOutputBufferSize) ? B_ERROR : B_LAST_BUFFER_ERROR; +} + +status_t +vorbisDecoder::DecodeNextChunk() +{ + void *chunkBuffer; + int32 chunkSize; + media_header mh; + if (B_OK != GetNextChunk(&chunkBuffer, &chunkSize, &mh)) { + TRACE("vorbisDecoder::Decode: GetNextChunk failed\n"); + return B_ERROR; + } + + fStartTime = mh.start_time; + + //TRACE("vorbisDecoder: fStartTime reset to %.6f\n", fStartTime / 1000000.0); + + int outsize; + int result; + result = decodeMP3(&fMpgLibPrivate, (char *)chunkBuffer, chunkSize, (char *)fDecodeBuffer, DECODE_BUFFER_SIZE, &outsize); + if (result == MP3_ERR) { + TRACE("vorbisDecoder::Decode: decodeMP3 returned MP3_ERR\n"); + return B_ERROR; + } + + //printf("vorbisDecoder::Decode: decoded %d bytes into %d bytes\n",chunkSize, outsize); + + fResidualBuffer = fDecodeBuffer; + fResidualBytes = outsize; + + return B_OK; +} + + +Decoder * +vorbisDecoderPlugin::NewDecoder() +{ + static BLocker locker; + static bool initdone = false; + locker.Lock(); + if (!initdone) { + InitMpgLib(); + initdone = true; + } + locker.Unlock(); + return new vorbisDecoder; +} + + +status_t +vorbisDecoderPlugin::RegisterPlugin() +{ + PublishDecoder("audiocodec/mpeg1layer1", "mp3", "MPEG 1 audio layer 1 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg1layer2", "mp3", "MPEG 1 audio layer 2 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg1layer3", "mp3", "MPEG 1 audio layer 3 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2layer1", "mp3", "MPEG 2 audio layer 1 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2layer2", "mp3", "MPEG 2 audio layer 2 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2layer3", "mp3", "MPEG 2 audio layer 3 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2.5layer1", "mp3", "MPEG 2.5 audio layer 1 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2.5layer2", "mp3", "MPEG 2.5 audio layer 2 decoder, based on mpeg123 mpglib"); + PublishDecoder("audiocodec/mpeg2.5layer3", "mp3", "MPEG 2.5 audio layer 3 decoder, based on mpeg123 mpglib"); + return B_OK; +} + + +MediaPlugin *instantiate_plugin() +{ + return new vorbisDecoderPlugin; +} diff --git a/src/add-ons/media/plugins/vorbis/vorbisCodecPlugin.h b/src/add-ons/media/plugins/vorbis/vorbisCodecPlugin.h new file mode 100644 index 0000000000..42045d51e7 --- /dev/null +++ b/src/add-ons/media/plugins/vorbis/vorbisCodecPlugin.h @@ -0,0 +1,45 @@ +#include "DecoderPlugin.h" + +#include "libvorbis/vorbis/codec.h" + +class vorbisDecoder : public Decoder +{ +public: + vorbisDecoder(); + ~vorbisDecoder(); + + status_t Setup(media_format *ioEncodedFormat, + const void *infoBuffer, int32 infoSize); + + status_t NegotiateOutputFormat(media_format *ioDecodedFormat); + + status_t Seek(uint32 seekTo, + int64 seekFrame, int64 *frame, + bigtime_t seekTime, bigtime_t *time); + + + status_t Decode(void *buffer, int64 *frameCount, + media_header *mediaHeader, media_decode_info *info); + + status_t DecodeNextChunk(); + +private: + struct mpstr fMpgLibPrivate; + int32 fResidualBytes; + uint8 * fResidualBuffer; + uint8 * fDecodeBuffer; + bigtime_t fStartTime; + int fFrameSize; + int fFrameRate; + int fBitRate; + int fChannelCount; + int fOutputBufferSize; +}; + + +class vorbisDecoderPlugin : public DecoderPlugin +{ +public: + Decoder * NewDecoder(); + status_t RegisterPlugin(); +};