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vorbis decoder plugin based on mp3

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git-svn-id: file:///srv/svn/repos/haiku/trunk/current@5664 a95241bf-73f2-0310-859d-f6bbb57e9c96
This commit is contained in:
shatty 2003-12-13 20:17:56 +00:00
parent 1c828415cf
commit 6ae16148a5
49 changed files with 17084 additions and 0 deletions
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14
src/add-ons/media/plugins/vorbis/Jamfile Normal file
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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 ;

29
src/add-ons/media/plugins/vorbis/libvorbis/Jamfile Normal file
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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
;

119
src/add-ons/media/plugins/vorbis/libvorbis/analysis.c Normal file
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/********************************************************************
* *
* 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 <stdio.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#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<n;j++){
if(bark){
float b=toBARK((4000.f*j/n)+.25);
fprintf(of,"%f ",b);
}else
if(off!=0)
fprintf(of,"%f ",(double)(j+off)/8000.);
else
fprintf(of,"%f ",(double)j);
if(dB){
float val;
if(v[j]==0.)
val=-140.;
else
val=todB(v+j);
fprintf(of,"%f\n",val);
}else{
fprintf(of,"%f\n",v[j]);
}
}
fclose(of);
/* } */
}
void _analysis_output(char *base,int i,float *v,int n,int bark,int dB,
ogg_int64_t off){
if(analysis_noisy)_analysis_output_always(base,i,v,n,bark,dB,off);
}

145
src/add-ons/media/plugins/vorbis/libvorbis/backends.h Normal file
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/********************************************************************
* *
* 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 backend and mapping structures; needed for
static mode headers
last mod: $Id: backends.h,v 1.1 2003/12/13 20:17:55 shatty Exp $
********************************************************************/
/* this is exposed up here because we need it for static modes.
Lookups for each backend aren't exposed because there's no reason
to do so */
#ifndef _vorbis_backend_h_
#define _vorbis_backend_h_
#include "codec_internal.h"
/* this would all be simpler/shorter with templates, but.... */
/* Floor backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info_floor *,oggpack_buffer *);
vorbis_info_floor *(*unpack)(vorbis_info *,oggpack_buffer *);
vorbis_look_floor *(*look) (vorbis_dsp_state *,vorbis_info_floor *);
void (*free_info) (vorbis_info_floor *);
void (*free_look) (vorbis_look_floor *);
void *(*inverse1) (struct vorbis_block *,vorbis_look_floor *);
int (*inverse2) (struct vorbis_block *,vorbis_look_floor *,
void *buffer,float *);
} vorbis_func_floor;
typedef struct{
int order;
long rate;
long barkmap;
int ampbits;
int ampdB;
int numbooks; /* <= 16 */
int books[16];
float lessthan; /* encode-only config setting hacks for libvorbis */
float greaterthan; /* encode-only config setting hacks for libvorbis */
} vorbis_info_floor0;
#define VIF_POSIT 63
#define VIF_CLASS 16
#define VIF_PARTS 31
typedef struct{
int partitions; /* 0 to 31 */
int partitionclass[VIF_PARTS]; /* 0 to 15 */
int class_dim[VIF_CLASS]; /* 1 to 8 */
int class_subs[VIF_CLASS]; /* 0,1,2,3 (bits: 1<<n poss) */
int class_book[VIF_CLASS]; /* subs ^ dim entries */
int class_subbook[VIF_CLASS][8]; /* [VIF_CLASS][subs] */
int mult; /* 1 2 3 or 4 */
int postlist[VIF_POSIT+2]; /* first two implicit */
/* encode side analysis parameters */
float maxover;
float maxunder;
float maxerr;
float twofitweight;
float twofitatten;
int n;
} vorbis_info_floor1;
/* Residue backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info_residue *,oggpack_buffer *);
vorbis_info_residue *(*unpack)(vorbis_info *,oggpack_buffer *);
vorbis_look_residue *(*look) (vorbis_dsp_state *,
vorbis_info_residue *);
void (*free_info) (vorbis_info_residue *);
void (*free_look) (vorbis_look_residue *);
long **(*class) (struct vorbis_block *,vorbis_look_residue *,
float **,int *,int);
int (*forward) (struct vorbis_block *,vorbis_look_residue *,
float **,float **,int *,int,long **);
int (*inverse) (struct vorbis_block *,vorbis_look_residue *,
float **,int *,int);
} vorbis_func_residue;
typedef struct vorbis_info_residue0{
/* block-partitioned VQ coded straight residue */
long begin;
long end;
/* first stage (lossless partitioning) */
int grouping; /* group n vectors per partition */
int partitions; /* possible codebooks for a partition */
int groupbook; /* huffbook for partitioning */
int secondstages[64]; /* expanded out to pointers in lookup */
int booklist[256]; /* list of second stage books */
float classmetric1[64];
float classmetric2[64];
} vorbis_info_residue0;
/* Mapping backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info *,vorbis_info_mapping *,
oggpack_buffer *);
vorbis_info_mapping *(*unpack)(vorbis_info *,oggpack_buffer *);
void (*free_info) (vorbis_info_mapping *);
int (*forward) (struct vorbis_block *vb);
int (*inverse) (struct vorbis_block *vb,vorbis_info_mapping *);
} vorbis_func_mapping;
typedef struct vorbis_info_mapping0{
int submaps; /* <= 16 */
int chmuxlist[256]; /* up to 256 channels in a Vorbis stream */
int floorsubmap[16]; /* [mux] submap to floors */
int residuesubmap[16]; /* [mux] submap to residue */
int coupling_steps;
int coupling_mag[256];
int coupling_ang[256];
} vorbis_info_mapping0;
#endif

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src/add-ons/media/plugins/vorbis/libvorbis/barkmel.c Normal file
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/********************************************************************
* *
* 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: bark scale utility
last mod: $Id: barkmel.c,v 1.1 2003/12/13 20:17:55 shatty Exp $
********************************************************************/
#include <stdio.h>
#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);
}

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src/add-ons/media/plugins/vorbis/libvorbis/bitrate.c Normal file
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/********************************************************************
* *
* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#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+1<bm->queue_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;i<maxpackets;i++)
oggpack_writeinit(bm->packetbuffers+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;i<bm->queue_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;i<PACKETBLOBS;i++){
head_ptr[i]=vbi->packetblob_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;i<bins;i++)
bm->avg_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(upper<new)new=upper;
if(lower>new)new=lower;
slew=(new-bm->avgfloat)/samples*vi->rate;
if(slew<bi->avgfloat_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;i<bm->queue_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,
actual<i?actual:i));
}
bm->minmax_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 && bitrate<bi->queue_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(bitrate<bi->queue_hardmin){
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_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]<limit)break;
newstack--;
}
/* update bit counter with new limit and replace any stack
limits that have been replaced by our new lower limit */
stackctr=bm->minmax_stackptr;
while(stackctr>newstack){
bm->minmax_acctotal-=
LIMITBYTES(stackctr,bm->minmax_limitstack[stackctr]);
bm->minmax_acctotal+=LIMITBYTES(stackctr,limit);
if(stackctr<bm->minmax_stackptr)
for(i=0;i<bins*2;i++)
bm->minmax_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;i<bins;i++){
bm->minmax_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,
actual<i?
actual:i));
}
if(bm->minmax_limitstack[0]>actual)
actual=bm->minmax_limitstack[0];
if(bins+bm->minmax_limitstack[0]<actual)
actual=bins+bm->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;i<bin;i++)
op->packet+=markers[i];
op->bytes=bytes;
}
bm->next_to_flush++;
if(bm->next_to_flush>=bm->queue_size)bm->next_to_flush=0;
}
return(1);
}

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <ogg/ogg.h>
#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;i<ci->books;i++)
vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
}
b->psy=_ogg_calloc(ci->psys,sizeof(*b->psy));
for(i=0;i<ci->psys;i++){
_vp_psy_init(b->psy+i,
ci->psy_param[i],
&ci->psy_g_param,
ci->blocksizes[ci->psy_param[i]->blockflag]/2,
vi->rate);
}
v->analysisp=1;
}else{
/* finish the codebooks */
if(!ci->fullbooks){
ci->fullbooks=_ogg_calloc(ci->books,sizeof(*ci->fullbooks));
for(i=0;i<ci->books;i++){
vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]);
/* decode codebooks are now standalone after init */
vorbis_staticbook_destroy(ci->book_param[i]);
ci->book_param[i]=NULL;
}
}
}
/* initialize the storage vectors. 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;i<vi->channels;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;i<ci->floors;i++)
b->flr[i]=_floor_P[ci->floor_type[i]]->
look(v,ci->floor_param[i]);
for(i=0;i<ci->residues;i++)
b->residue[i]=_residue_P[ci->residue_type[i]]->
look(v,ci->residue_param[i]);
return(0);
}
/* arbitrary settings and spec-mandated numbers get filled in here */
int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi){
backend_lookup_state *b=NULL;
_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;i<ci->floors;i++)
_floor_P[ci->floor_type[i]]->
free_look(b->flr[i]);
_ogg_free(b->flr);
}
if(b->residue){
for(i=0;i<ci->residues;i++)
_residue_P[ci->residue_type[i]]->
free_look(b->residue[i]);
_ogg_free(b->residue);
}
if(b->psy){
for(i=0;i<ci->psys;i++)
_vp_psy_clear(b->psy+i);
_ogg_free(b->psy);
}
if(b->psy_g_look)_vp_global_free(b->psy_g_look);
vorbis_bitrate_clear(&b->bms);
drft_clear(&b->fft_look[0]);
drft_clear(&b->fft_look[1]);
}
if(v->pcm){
for(i=0;i<vi->channels;i++)
if(v->pcm[i])_ogg_free(v->pcm[i]);
_ogg_free(v->pcm);
if(v->pcmret)_ogg_free(v->pcmret);
}
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;i<vi->channels;i++){
v->pcm[i]=_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
}
}
for(i=0;i<vi->channels;i++)
v->pcmret[i]=v->pcm[i]+v->pcm_current;
return(v->pcmret);
}
static void _preextrapolate_helper(vorbis_dsp_state *v){
int i;
int order=32;
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;i<v->vi->channels;i++){
/* need to run the extrapolation in reverse! */
for(j=0;j<v->pcm_current;j++)
work[j]=v->pcm[i][v->pcm_current-j-1];
/* 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;j<v->pcm_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;i<vi->channels;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_current<blockbound)return(0); /* not enough data yet;
although this check is
less strict that the
_ve_envelope_search,
the search is not run
if we only use one
block size */
}
/* fill in the block. Note that for a short window, lW and nW are *short*
regardless of actual settings in the stream */
_vorbis_block_ripcord(vb);
vb->lW=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;i<vi->channels;i++){
vbi->pcmdelay[i]=
_vorbis_block_alloc(vb,(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
memcpy(vbi->pcmdelay[i],v->pcm[i],(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
vb->pcm[i]=vbi->pcmdelay[i]+beginW;
/* before we added the delay
vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i]));
memcpy(vb->pcm[i],v->pcm[i]+beginW,ci->blocksizes[v->W]*sizeof(*vb->pcm[i]));
*/
}
/* handle eof detection: eof==0 means that we've not yet received EOF
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;i<vi->channels;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;j<vi->channels;j++){
/* the overlap/add section */
if(v->lW){
if(v->W){
/* large/large */
float *pcm=v->pcm[j]+prevCenter;
float *p=vb->pcm[j];
for(i=0;i<n1;i++)
pcm[i]+=p[i];
}else{
/* large/small */
float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
float *p=vb->pcm[j];
for(i=0;i<n0;i++)
pcm[i]+=p[i];
}
}else{
if(v->W){
/* small/large */
float *pcm=v->pcm[j]+prevCenter;
float *p=vb->pcm[j]+n1/2-n0/2;
for(i=0;i<n0;i++)
pcm[i]+=p[i];
for(;i<n1/2+n0/2;i++)
pcm[i]=p[i];
}else{
/* small/small */
float *pcm=v->pcm[j]+prevCenter;
float *p=vb->pcm[j];
for(i=0;i<n0;i++)
pcm[i]+=p[i];
}
}
/* the copy section */
{
float *pcm=v->pcm[j]+thisCenter;
float *p=vb->pcm[j]+n;
for(i=0;i<n;i++)
pcm[i]=p[i];
}
}
if(v->centerW)
v->centerW=0;
else
v->centerW=n1;
/* deal with initial packet state; we do this using the explicit
pcm_returned==-1 flag otherwise we're sensitive to first block
being short or long */
if(v->pcm_returned==-1){
v->pcm_returned=thisCenter;
v->pcm_current=thisCenter;
}else{
v->pcm_returned=prevCenter;
v->pcm_current=prevCenter+
ci->blocksizes[v->lW]/4+
ci->blocksizes[v->W]/4;
}
}
/* 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_returned<v->pcm_current){
if(pcm){
int i;
for(i=0;i<vi->channels;i++)
v->pcmret[i]=v->pcm[i]+v->pcm_returned;
*pcm=v->pcmret;
}
return(v->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);
}

615
src/add-ons/media/plugins/vorbis/libvorbis/codebook.c Normal file
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@ -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 <stdlib.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#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;i<c->entries;i++)
if(c->lengthlist[i-1]==0 || c->lengthlist[i]<c->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;i<c->entries;i++){
long this=c->lengthlist[i];
long last=c->lengthlist[i-1];
if(this>last){
for(j=last;j<this;j++){
oggpack_write(opb,i-count,_ilog(c->entries-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;i<c->entries;i++)
if(c->lengthlist[i]==0)break;
if(i==c->entries){
oggpack_write(opb,0,1); /* no unused entries */
for(i=0;i<c->entries;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;i<c->entries;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;i<quantvals;i++)
oggpack_write(opb,labs(c->quantlist[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;i<s->entries;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;i<s->entries;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;i<s->entries;){
long num=oggpack_read(opb,_ilog(s->entries-i));
if(num==-1)goto _eofout;
for(j=0;j<num && i<s->entries;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;i<quantvals;i++)
s->quantlist[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;k<dim;k++)
a[k]=(book->valuelist+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;k<dim;k++)
a[k]=(book->valuelist+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;i<book->dim;i++,o+=step)
for (j=0;j<step;j++)
a[o+j]+=t[j][i];
return(0);
}
long vorbis_book_decodev_add(codebook *book,float *a,oggpack_buffer *b,int n){
int i,j,entry;
float *t;
if(book->dim>8){
for(i=0;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+entry*book->dim;
for (j=0;j<book->dim;)
a[i++]+=t[j++];
}
}else{
for(i=0;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+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;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+entry*book->dim;
for (j=0;j<book->dim;)
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;j<book->dim;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 <stdio.h>
#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<TESTSIZE;i+=c.dim){
int best=_best(&c,qv+i,1);
vorbis_book_encodev(&c,best,qv+i,&write);
}
vorbis_book_clear(&c);
fprintf(stderr,"OK.\n");
fprintf(stderr,"\tunpacking/decoding %ld... ",ptr);
/* transfer the write data to a read buffer and unpack/read */
oggpack_readinit(&read,oggpack_get_buffer(&write),oggpack_bytes(&write));
if(vorbis_staticbook_unpack(&read,&s)){
fprintf(stderr,"Error unpacking codebook.\n");
exit(1);
}
if(vorbis_book_init_decode(&c,&s)){
fprintf(stderr,"Error initializing codebook.\n");
exit(1);
}
for(i=0;i<TESTSIZE;i+=c.dim)
if(vorbis_book_decodev_set(&c,iv+i,&read,c.dim)==-1){
fprintf(stderr,"Error reading codebook test data (EOP).\n");
exit(1);
}
for(i=0;i<TESTSIZE;i++)
if(fabs(qv[i]-iv[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

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/********************************************************************
* *
* 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 <ogg/ogg.h>
/* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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 <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <math.h>
#include <ogg/ogg.h>
#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;i<n;i++){
e->mdct_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;j<VE_BANDS;j++){
n=e->band[j].end;
e->band[j].window=_ogg_malloc(n*sizeof(*e->band[0].window));
for(i=0;i<n;i++){
e->band[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;i<VE_BANDS;i++)
_ogg_free(e->band[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;i<n;i++)
vec[i]=data[i]*ve->mdct_win[i];
mdct_forward(&ve->mdct,vec,vec);
/*_analysis_output_always("mdct",seq2,vec,n/2,0,1,0); */
/* near-DC spreading function; this has nothing to do with
psychoacoustics, just sidelobe leakage and window size */
{
float temp=vec[0]*vec[0]+.7*vec[1]*vec[1]+.2*vec[2]*vec[2];
int ptr=filters->nearptr;
/* the accumulation is regularly refreshed from scratch to avoid
floating point creep */
if(ptr==0){
decay=filters->nearDC_acc=filters->nearDC_partialacc+temp;
filters->nearDC_partialacc=temp;
}else{
decay=filters->nearDC_acc+=temp;
filters->nearDC_partialacc+=temp;
}
filters->nearDC_acc-=filters->nearDC[ptr];
filters->nearDC[ptr]=temp;
decay*=(1./(VE_NEARDC+1));
filters->nearptr++;
if(filters->nearptr>=VE_NEARDC)filters->nearptr=0;
decay=todB(&decay)*.5-15.f;
}
/* perform spreading and limiting, also smooth the spectrum. yes,
the MDCT results in all real coefficients, but it still *behaves*
like real/imaginary pairs */
for(i=0;i<n/2;i+=2){
float val=vec[i]*vec[i]+vec[i+1]*vec[i+1];
val=todB(&val)*.5f;
if(val<decay)val=decay;
if(val<minV)val=minV;
vec[i>>1]=val;
decay-=8.;
}
/*_analysis_output_always("spread",seq2++,vec,n/4,0,0,0);*/
/* perform preecho/postecho triggering by band */
for(j=0;j<VE_BANDS;j++){
float acc=0.;
float valmax,valmin;
/* accumulate amplitude */
for(i=0;i<bands[j].end;i++)
acc+=vec[i+bands[j].begin]*bands[j].window[i];
acc*=bands[j].total;
/* convert amplitude to delta */
{
int p,this=filters[j].ampptr;
float postmax,postmin,premax=-99999.f,premin=99999.f;
p=this;
p--;
if(p<0)p+=VE_AMP;
postmax=max(acc,filters[j].ampbuf[p]);
postmin=min(acc,filters[j].ampbuf[p]);
for(i=0;i<stretch;i++){
p--;
if(p<0)p+=VE_AMP;
premax=max(premax,filters[j].ampbuf[p]);
premin=min(premin,filters[j].ampbuf[p]);
}
valmin=postmin-premin;
valmax=postmax-premax;
/*filters[j].markers[pos]=valmax;*/
filters[j].ampbuf[this]=acc;
filters[j].ampptr++;
if(filters[j].ampptr>=VE_AMP)filters[j].ampptr=0;
}
/* look at min/max, decide trigger */
if(valmax>gi->preecho_thresh[j]+penalty){
ret|=1;
ret|=4;
}
if(valmin<gi->postecho_thresh[j]-penalty)ret|=2;
}
return(ret);
}
#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;j<last;j++){
int ret=0;
ve->stretch++;
if(ve->stretch>VE_MAXSTRETCH*2)
ve->stretch=VE_MAXSTRETCH*2;
for(i=0;i<ve->ch;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(j<ve->current-(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;m<VE_BANDS;m++){
char buf[80];
sprintf(buf,"delL%d",m);
for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m].markers[l]*.1;
_analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
}
for(m=0;m<VE_BANDS;m++){
char buf[80];
sprintf(buf,"delR%d",m);
for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m+VE_BANDS].markers[l]*.1;
_analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
}
for(l=0;l<last;l++)marker[l*ve->searchstep]=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->curmark<endW)return(1);
{
long first=beginW/ve->searchstep;
long last=endW/ve->searchstep;
long i;
for(i=first;i<last;i++)
if(ve->mark[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;i<VE_BANDS*e->ch;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;
}

81
src/add-ons/media/plugins/vorbis/libvorbis/envelope.h Normal file
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/********************************************************************
* *
* 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

235
src/add-ons/media/plugins/vorbis/libvorbis/floor0.c Normal file
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/********************************************************************
* *
* 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 <stdlib.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#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 <stdio.h>
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;j<info->numbooks;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;j<n;j++){
int val=floor( toBARK((info->rate/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;j<look->ln;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<<info->ampbits)-1;
float amp=(float)ampraw/maxval*info->ampdB;
int booknum=oggpack_read(&vb->opb,_ilog(info->numbooks));
if(booknum!=-1 && booknum<info->numbooks){ /* 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;j<look->m;j+=b->dim)
if(vorbis_book_decodev_set(b,lsp+j,&vb->opb,b->dim)==-1)goto eop;
for(j=0;j<look->m;){
for(k=0;k<b->dim;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
};

1087
src/add-ons/media/plugins/vorbis/libvorbis/floor1.c Normal file
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58
src/add-ons/media/plugins/vorbis/libvorbis/highlevel.h Normal file
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/********************************************************************
* *
* 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: highlevel encoder setup struct seperated out for vorbisenc clarity
last mod: $Id: highlevel.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
typedef struct highlevel_byblocktype {
double tone_mask_setting;
double tone_peaklimit_setting;
double noise_bias_setting;
double noise_compand_setting;
} highlevel_byblocktype;
typedef struct highlevel_encode_setup {
void *setup;
int set_in_stone;
double base_setting;
double long_setting;
double short_setting;
double impulse_noisetune;
int managed;
long bitrate_min;
long bitrate_av_lo;
long bitrate_av_hi;
long bitrate_max;
double bitrate_limit_window;
double bitrate_av_window;
double bitrate_av_window_center;
int impulse_block_p;
int noise_normalize_p;
double stereo_point_setting;
double lowpass_kHz;
double ath_floating_dB;
double ath_absolute_dB;
double amplitude_track_dBpersec;
double trigger_setting;
highlevel_byblocktype block[4]; /* padding, impulse, transition, long */
} highlevel_encode_setup;

598
src/add-ons/media/plugins/vorbis/libvorbis/info.c Normal file
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/********************************************************************
* *
* 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: maintain the info structure, info <-> 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 <stdlib.h>
#include <string.h>
#include <ctype.h>
#include <ogg/ogg.h>
#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;i<vc->comments;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;i<vc->comments;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;i<vc->comments;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;i<ci->modes;i++)
if(ci->mode_param[i])_ogg_free(ci->mode_param[i]);
for(i=0;i<ci->maps;i++) /* unpack does the range checking */
_mapping_P[ci->map_type[i]]->free_info(ci->map_param[i]);
for(i=0;i<ci->floors;i++) /* unpack does the range checking */
_floor_P[ci->floor_type[i]]->free_info(ci->floor_param[i]);
for(i=0;i<ci->residues;i++) /* unpack does the range checking */
_residue_P[ci->residue_type[i]]->free_info(ci->residue_param[i]);
for(i=0;i<ci->books;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;i<ci->psys;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<<oggpack_read(opb,4);
ci->blocksizes[1]=1<<oggpack_read(opb,4);
if(vi->rate<1)goto err_out;
if(vi->channels<1)goto err_out;
if(ci->blocksizes[0]<8)goto err_out;
if(ci->blocksizes[1]<ci->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;i<vc->comments;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;i<ci->books;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<times;i++){
int test=oggpack_read(opb,16);
if(test<0 || test>=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;i<ci->floors;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;i<ci->residues;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;i<ci->maps;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;i<ci->modes;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;i<vc->comments;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;i<ci->books;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;i<ci->floors;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;i<ci->residues;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;i<ci->maps;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;i<ci->modes;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);
}

94
src/add-ons/media/plugins/vorbis/libvorbis/lookup.c Normal file
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@ -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 <math.h>
#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 <stdio.h>
/* interpolated lookup based fromdB function, domain -140dB to 0dB only */
float vorbis_fromdBlook(float a){
int i=vorbis_ftoi(a*((float)(-(1<<FROMdB2_SHIFT)))-.5f);
return (i<0)?1.f:
((i>=(FROMdB_LOOKUP_SZ<<FROMdB_SHIFT))?0.f:
FROMdB_LOOKUP[i>>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))?0.f:
FROMdB_LOOKUP[i>>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

32
src/add-ons/media/plugins/vorbis/libvorbis/lookup.h Normal file
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/********************************************************************
* *
* 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

189
src/add-ons/media/plugins/vorbis/libvorbis/lookup_data.h Normal file
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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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 <stdlib.h>
#include <string.h>
#include <math.h>
#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;i<n;i++)d+=data[i]*data[i-j];
aut[j]=d;
}
/* Generate lpc coefficients from autocorr values */
error=aut[0];
for(i=0;i<m;i++){
float r= -aut[i+1];
if(error==0){
memset(lpc,0,m*sizeof(*lpc));
return 0;
}
/* Sum up this iteration's reflection coefficient; note that in
Vorbis we don't save it. If anyone wants to recycle this code
and needs reflection coefficients, save the results of 'r' from
each iteration. */
for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
r/=error;
/* Update LPC coefficients and total error */
lpc[i]=r;
for(j=0;j<i/2;j++){
float tmp=lpc[j];
lpc[j]+=r*lpc[i-1-j];
lpc[i-1-j]+=r*tmp;
}
if(i%2)lpc[j]+=lpc[j]*r;
error*=1.f-r*r;
}
/* we need the error value to know how big an impulse to hit the
filter with later */
return error;
}
/* Input : n element envelope spectral curve
Output: m lpc coefficients, excitation energy */
float vorbis_lpc_from_curve(float *curve,float *lpc,lpc_lookup *l){
int n=l->ln;
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;i<n;i++){
work[i*2]=curve[i]*fscale;
work[i*2+1]=0;
}
work[n*2-1]=curve[n-1]*fscale;
n*=2;
drft_backward(&l->fft,work);
/* The autocorrelation will not be circular. Shift, else we lose
most of the power in the edges. */
for(i=0,j=n/2;i<n/2;){
float temp=work[i];
work[i++]=work[j];
work[j++]=temp;
}
/* we *could* shave speed here by skimping on the edges (thus
speeding up the autocorrelation in vorbis_lpc_from_data) but we
don't right now. */
return(vorbis_lpc_from_data(work,lpc,n,m));
}
void lpc_init(lpc_lookup *l,long mapped, int m){
memset(l,0,sizeof(*l));
l->ln=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<m;i++)
work[i]=0.f;
else
for(i=0;i<m;i++)
work[i]=prime[i];
for(i=0;i<n;i++){
y=0;
o=i;
p=m;
for(j=0;j<m;j++)
y-=work[o++]*coeff[--p];
data[i]=work[o]=y;
}
}

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/********************************************************************
* *
* 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.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#ifndef _V_LPC_H_
#define _V_LPC_H_
#include "vorbis/codec.h"
#include "smallft.h"
typedef struct lpclook{
/* en/decode lookups */
drft_lookup fft;
int ln;
int m;
} lpc_lookup;
extern void lpc_init(lpc_lookup *l,long mapped, int m);
extern void lpc_clear(lpc_lookup *l);
/* simple linear scale LPC code */
extern float vorbis_lpc_from_data(float *data,float *lpc,int n,int m);
extern float vorbis_lpc_from_curve(float *curve,float *lpc,lpc_lookup *l);
extern void vorbis_lpc_predict(float *coeff,float *prime,int m,
float *data,long n);
#endif

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/********************************************************************
* *
* 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: LSP (also called LSF) conversion routines
last mod: $Id: lsp.c,v 1.1 2003/12/13 20:17:56 shatty Exp $
The LSP generation code is taken (with minimal modification and a
few bugfixes) from "On the Computation of the LSP Frequencies" by
Joseph Rothweiler <rothwlr@altavista.net>, 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 <math.h>
#include <string.h>
#include <stdlib.h>
#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<m;i++)lsp[i]=vorbis_coslook(lsp[i]);
i=0;
while(i<n){
int k=map[i];
int qexp;
float p=.7071067812f;
float q=.7071067812f;
float w=vorbis_coslook(wdel*k);
float *ftmp=lsp;
int c=m>>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<m;i++)ilsp[i]=vorbis_coslook_i(lsp[i]/M_PI*65536.f+.5f);
i=0;
while(i<n){
int j,k=map[i];
unsigned long pi=46341; /* 2**-.5 in 0.16 */
unsigned long qi=46341;
int qexp=0,shift;
long wi=vorbis_coslook_i(k*65536/ln);
qi*=labs(ilsp[0]-wi);
pi*=labs(ilsp[1]-wi);
for(j=3;j<m;j+=2){
if(!(shift=MLOOP_1[(pi|qi)>>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<m;i++)lsp[i]=2.f*cos(lsp[i]);
i=0;
while(i<n){
int j,k=map[i];
float p=.5f;
float q=.5f;
float w=2.f*cos(wdel*k);
for(j=1;j<m;j+=2){
q *= w-lsp[j-1];
p *= w-lsp[j];
}
if(j==m){
/* odd order filter; slightly assymetric */
/* the last coefficient */
q*=w-lsp[j-1];
p*=p*(4.f-w*w);
q*=q;
}else{
/* even order filter; still symmetric */
p*=p*(2.f-w);
q*=q*(2.f+w);
}
q=fromdB(amp/sqrt(p+q)-ampoffset);
curve[i]*=q;
while(map[++i]==k)curve[i]*=q;
}
}
#endif
#endif
static void cheby(float *g, int ord) {
int i, j;
g[0] *= .5f;
for(i=2; i<= ord; i++) {
for(j=ord; j >= 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;
}else{
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; i<ord;i++) root[i] = r[i];
while(error>1e-20){
error=0;
for(i=0; i<ord; i++) { /* Update each point. */
double pp=0.,delta;
double rooti=root[i];
double p=a[ord];
for(k=ord-1; k>= 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<ord;i++) r[i] = root[i];
return(0);
}
/* Convert lpc coefficients to lsp coefficients */
int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m){
int order2=(m+1)>>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<g1_order;i++)
lsp[i*2] = acos(g1r[i]);
for(i=0;i<g2_order;i++)
lsp[i*2+1] = acos(g2r[i]);
return(0);
}

28
src/add-ons/media/plugins/vorbis/libvorbis/lsp.h Normal file
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/********************************************************************
* *
* 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: LSP (also called LSF) conversion routines
last mod: $Id: lsp.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#ifndef _V_LSP_H_
#define _V_LSP_H_
extern int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m);
extern void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,
float *lsp,int m,
float amp,float ampoffset);
#endif

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/********************************************************************
* *
* 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: channel mapping 0 implementation
last mod: $Id: mapping0.c,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#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;i<info->coupling_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;i<vi->channels;i++)
oggpack_write(opb,info->chmuxlist[i],4);
}
for(i=0;i<info->submaps;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;i<info->coupling_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;i<vi->channels;i++){
info->chmuxlist[i]=oggpack_read(opb,4);
if(info->chmuxlist[i]>=info->submaps)goto err_out;
}
}
for(i=0;i<info->submaps;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;i<vi->channels;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<n-1;j+=2){
float temp=pcm[j]*pcm[j]+pcm[j+1]*pcm[j+1];
temp=logfft[(j+1)>>1]=scale_dB+.5f*todB(&temp);
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;i<vi->channels;i++){
/* the encoder setup assumes that all the modes used by any
specific bitrate tweaking use the same floor */
int submap=info->chmuxlist[i];
/* the following makes things clearer to *me* anyway */
float *mdct =gmdct[i];
float *logfft =vb->pcm[i];
float *logmdct =logfft+n/2;
float *logmask =logfft;
vb->mode=modenumber;
floor_posts[i]=_vorbis_block_alloc(vb,PACKETBLOBS*sizeof(**floor_posts));
memset(floor_posts[i],0,sizeof(**floor_posts)*PACKETBLOBS);
for(j=0;j<n/2;j++)
logmdct[j]=todB(mdct+j);
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("mdctL",seq,logmdct,n/2,1,0,0);
else
_analysis_output("mdctR",seq,logmdct,n/2,1,0,0);
}else{
_analysis_output("mdct",seq,logmdct,n/2,1,0,0);
}
#endif
/* first step; noise masking. Not only does 'noise masking'
give us curves from which we can decide how much resolution
to give noise parts of the spectrum, it also implicitly hands
us a tonality estimate (the larger the value in the
'noise_depth' vector, the more tonal that area is) */
_vp_noisemask(psy_look,
logmdct,
noise); /* noise does not have by-frequency offset
bias applied yet */
#if 0
if(vi->channels==2){
if(i==0)
_analysis_output("noiseL",seq,noise,n/2,1,0,0);
else
_analysis_output("noiseR",seq,noise,n/2,1,0,0);
}
#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;k<PACKETBLOBS/2;k++)
floor_posts[i][k]=
floor1_interpolate_fit(vb,b->flr[info->floorsubmap[submap]],
floor_posts[i][0],
floor_posts[i][PACKETBLOBS/2],
k*65536/(PACKETBLOBS/2));
for(k=PACKETBLOBS/2+1;k<PACKETBLOBS-1;k++)
floor_posts[i][k]=
floor1_interpolate_fit(vb,b->flr[info->floorsubmap[submap]],
floor_posts[i][PACKETBLOBS/2],
floor_posts[i][PACKETBLOBS-1],
(k-PACKETBLOBS/2)*65536/(PACKETBLOBS/2));
}
}
}
vbi->ampmax=global_ampmax;
/*
the next phases are performed once for vbr-only and PACKETBLOB
times for bitrate managed modes.
1) encode actual mode being used
2) encode the floor for each channel, compute coded mask curve/res
3) normalize and couple.
4) encode residue
5) save packet bytes to the packetblob vector
*/
/* iterate over the many masking curve fits we've created */
{
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;i<vi->channels;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;i<vi->channels;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;j<n/2;j++)
work[j]=FLOOR1_fromdB_LOOKUP[ilogmask[j]];
_analysis_output(buf,seq,work,n/2,1,1,0);
}
#endif
_vp_remove_floor(psy_look,
mdct,
ilogmask,
res,
ci->psy_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;j<n/2;j++)
work[j]=FLOOR1_fromdB_LOOKUP[ilogmask[j]]*(res+n/2)[j];
sprintf(buf,"resI%c%d",i?'R':'L',k);
_analysis_output(buf,seq,work,n/2,1,1,0);
}
#endif
}
/* our iteration is now based on masking curve, not prequant and
coupling. Only one prequant/coupling step */
/* quantize/couple */
/* incomplete implementation that assumes the tree is all depth
one, or no tree at all */
if(info->coupling_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;i<info->submaps;i++){
int ch_in_bundle=0;
long **classifications;
int resnum=info->residuesubmap[i];
for(j=0;j<vi->channels;j++){
if(info->chmuxlist[j]==i){
zerobundle[ch_in_bundle]=0;
if(nonzero[j])zerobundle[ch_in_bundle]=1;
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;i<vi->channels;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;i<info->coupling_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;i<info->submaps;i++){
int ch_in_bundle=0;
for(j=0;j<vi->channels;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;j<n/2;j++){
float mag=pcmM[j];
float ang=pcmA[j];
if(mag>0)
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag-ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag+ang;
}
else
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag+ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag-ang;
}
}
}
/* compute and apply spectral envelope */
for(i=0;i<vi->channels;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;i<vi->channels;i++){
float *pcm=vb->pcm[i];
mdct_backward(b->transform[vb->W][0],pcm,pcm);
}
/* window the data */
for(i=0;i<vi->channels;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<n;j++)
pcm[j]=0.f;
}
/* all done! */
return(0);
}
/* export hooks */
vorbis_func_mapping mapping0_exportbundle={
&mapping0_pack,
&mapping0_unpack,
&mapping0_free_info,
&mapping0_forward,
&mapping0_inverse
};

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src/add-ons/media/plugins/vorbis/libvorbis/masking.h Normal file
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@ -0,0 +1,785 @@
/********************************************************************
* *
* 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: masking curve data for psychoacoustics
last mod: $Id: masking.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#ifndef _V_MASKING_H_
#define _V_MASKING_H_
/* more detailed ATH; the bass if flat to save stressing the floor
overly for only a bin or two of savings. */
#define MAX_ATH 88
static float ATH[]={
/*15*/ -51, -52, -53, -54, -55, -56, -57, -58,
/*31*/ -59, -60, -61, -62, -63, -64, -65, -66,
/*63*/ -67, -68, -69, -70, -71, -72, -73, -74,
/*125*/ -75, -76, -77, -78, -80, -81, -82, -83,
/*250*/ -84, -85, -86, -87, -88, -88, -89, -89,
/*500*/ -90, -91, -91, -92, -93, -94, -95, -96,
/*1k*/ -96, -97, -98, -98, -99, -99,-100,-100,
/*2k*/ -101,-102,-103,-104,-106,-107,-107,-107,
/*4k*/ -107,-105,-103,-102,-101, -99, -98, -96,
/*8k*/ -95, -95, -96, -97, -96, -95, -93, -90,
/*16k*/ -80, -70, -50, -40, -30, -30, -30, -30
};
/* The tone masking curves from Ehmer's and Fielder's papers have been
replaced by an empirically collected data set. The previously
published values were, far too often, simply on crack. */
#define EHMER_OFFSET 16
#define EHMER_MAX 56
/* masking tones from -50 to 0dB, 62.5 through 16kHz at half octaves
test tones from -2 octaves to +5 octaves sampled at eighth octaves */
/* (Vorbis 0dB, the loudest possible tone, is assumed to be ~100dB SPL
for collection of these curves) */
static float tonemasks[P_BANDS][6][EHMER_MAX]={
/* 62.5 Hz */
{{ -60, -60, -60, -60, -60, -60, -60, -60,
-60, -60, -60, -60, -62, -62, -65, -73,
-69, -68, -68, -67, -70, -70, -72, -74,
-75, -79, -79, -80, -83, -88, -93, -100,
-110, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -48, -48, -48, -48, -48, -48, -48, -48,
-48, -48, -48, -48, -48, -53, -61, -66,
-66, -68, -67, -70, -76, -76, -72, -73,
-75, -76, -78, -79, -83, -88, -93, -100,
-110, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -37, -37, -37, -37, -37, -37, -37, -37,
-38, -40, -42, -46, -48, -53, -55, -62,
-65, -58, -56, -56, -61, -60, -65, -67,
-69, -71, -77, -77, -78, -80, -82, -84,
-88, -93, -98, -106, -112, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -25, -25, -25, -25, -25, -25, -25, -25,
-25, -26, -27, -29, -32, -38, -48, -52,
-52, -50, -48, -48, -51, -52, -54, -60,
-67, -67, -66, -68, -69, -73, -73, -76,
-80, -81, -81, -85, -85, -86, -88, -93,
-100, -110, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -16, -16, -16, -16, -16, -16, -16, -16,
-17, -19, -20, -22, -26, -28, -31, -40,
-47, -39, -39, -40, -42, -43, -47, -51,
-57, -52, -55, -55, -60, -58, -62, -63,
-70, -67, -69, -72, -73, -77, -80, -82,
-83, -87, -90, -94, -98, -104, -115, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -8, -8, -8, -8, -8, -8, -8, -8,
-8, -8, -10, -11, -15, -19, -25, -30,
-34, -31, -30, -31, -29, -32, -35, -42,
-48, -42, -44, -46, -50, -50, -51, -52,
-59, -54, -55, -55, -58, -62, -63, -66,
-72, -73, -76, -75, -78, -80, -80, -81,
-84, -88, -90, -94, -98, -101, -106, -110}},
/* 88Hz */
{{ -66, -66, -66, -66, -66, -66, -66, -66,
-66, -66, -66, -66, -66, -67, -67, -67,
-76, -72, -71, -74, -76, -76, -75, -78,
-79, -79, -81, -83, -86, -89, -93, -97,
-100, -105, -110, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -47, -47, -47, -47, -47, -47, -47, -47,
-47, -47, -47, -48, -51, -55, -59, -66,
-66, -66, -67, -66, -68, -69, -70, -74,
-79, -77, -77, -78, -80, -81, -82, -84,
-86, -88, -91, -95, -100, -108, -116, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -36, -36, -36, -36, -36, -36, -36, -36,
-36, -37, -37, -41, -44, -48, -51, -58,
-62, -60, -57, -59, -59, -60, -63, -65,
-72, -71, -70, -72, -74, -77, -76, -78,
-81, -81, -80, -83, -86, -91, -96, -100,
-105, -110, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -28, -28, -28, -28, -28, -28, -28, -28,
-28, -30, -32, -32, -33, -35, -41, -49,
-50, -49, -47, -48, -48, -52, -51, -57,
-65, -61, -59, -61, -64, -69, -70, -74,
-77, -77, -78, -81, -84, -85, -87, -90,
-92, -96, -100, -107, -112, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -19, -19, -19, -19, -19, -19, -19, -19,
-20, -21, -23, -27, -30, -35, -36, -41,
-46, -44, -42, -40, -41, -41, -43, -48,
-55, -53, -52, -53, -56, -59, -58, -60,
-67, -66, -69, -71, -72, -75, -79, -81,
-84, -87, -90, -93, -97, -101, -107, -114,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -9, -9, -9, -9, -9, -9, -9, -9,
-11, -12, -12, -15, -16, -20, -23, -30,
-37, -34, -33, -34, -31, -32, -32, -38,
-47, -44, -41, -40, -47, -49, -46, -46,
-58, -50, -50, -54, -58, -62, -64, -67,
-67, -70, -72, -76, -79, -83, -87, -91,
-96, -100, -104, -110, -999, -999, -999, -999}},
/* 125 Hz */
{{ -62, -62, -62, -62, -62, -62, -62, -62,
-62, -62, -63, -64, -66, -67, -66, -68,
-75, -72, -76, -75, -76, -78, -79, -82,
-84, -85, -90, -94, -101, -110, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -59, -59, -59, -59, -59, -59, -59, -59,
-59, -59, -59, -60, -60, -61, -63, -66,
-71, -68, -70, -70, -71, -72, -72, -75,
-81, -78, -79, -82, -83, -86, -90, -97,
-103, -113, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -53, -53, -53, -53, -53, -53, -53, -53,
-53, -54, -55, -57, -56, -57, -55, -61,
-65, -60, -60, -62, -63, -63, -66, -68,
-74, -73, -75, -75, -78, -80, -80, -82,
-85, -90, -96, -101, -108, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -46, -46, -46, -46, -46, -46, -46, -46,
-46, -46, -47, -47, -47, -47, -48, -51,
-57, -51, -49, -50, -51, -53, -54, -59,
-66, -60, -62, -67, -67, -70, -72, -75,
-76, -78, -81, -85, -88, -94, -97, -104,
-112, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -36, -36, -36, -36, -36, -36, -36, -36,
-39, -41, -42, -42, -39, -38, -41, -43,
-52, -44, -40, -39, -37, -37, -40, -47,
-54, -50, -48, -50, -55, -61, -59, -62,
-66, -66, -66, -69, -69, -73, -74, -74,
-75, -77, -79, -82, -87, -91, -95, -100,
-108, -115, -999, -999, -999, -999, -999, -999},
{ -28, -26, -24, -22, -20, -20, -23, -29,
-30, -31, -28, -27, -28, -28, -28, -35,
-40, -33, -32, -29, -30, -30, -30, -37,
-45, -41, -37, -38, -45, -47, -47, -48,
-53, -49, -48, -50, -49, -49, -51, -52,
-58, -56, -57, -56, -60, -61, -62, -70,
-72, -74, -78, -83, -88, -93, -100, -106}},
/* 177 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -110, -105, -100, -95, -91, -87, -83,
-80, -78, -76, -78, -78, -81, -83, -85,
-86, -85, -86, -87, -90, -97, -107, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -110, -105, -100, -95, -90,
-85, -81, -77, -73, -70, -67, -67, -68,
-75, -73, -70, -69, -70, -72, -75, -79,
-84, -83, -84, -86, -88, -89, -89, -93,
-98, -105, -112, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-105, -100, -95, -90, -85, -80, -76, -71,
-68, -68, -65, -63, -63, -62, -62, -64,
-65, -64, -61, -62, -63, -64, -66, -68,
-73, -73, -74, -75, -76, -81, -83, -85,
-88, -89, -92, -95, -100, -108, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -80, -75, -71, -68, -65, -63, -62, -61,
-61, -61, -61, -59, -56, -57, -53, -50,
-58, -52, -50, -50, -52, -53, -54, -58,
-67, -63, -67, -68, -72, -75, -78, -80,
-81, -81, -82, -85, -89, -90, -93, -97,
-101, -107, -114, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{ -65, -61, -59, -57, -56, -55, -55, -56,
-56, -57, -55, -53, -52, -47, -44, -44,
-50, -44, -41, -39, -39, -42, -40, -46,
-51, -49, -50, -53, -54, -63, -60, -61,
-62, -66, -66, -66, -70, -73, -74, -75,
-76, -75, -79, -85, -89, -91, -96, -102,
-110, -999, -999, -999, -999, -999, -999, -999},
{ -52, -50, -49, -49, -48, -48, -48, -49,
-50, -50, -49, -46, -43, -39, -35, -33,
-38, -36, -32, -29, -32, -32, -32, -35,
-44, -39, -38, -38, -46, -50, -45, -46,
-53, -50, -50, -50, -54, -54, -53, -53,
-56, -57, -59, -66, -70, -72, -74, -79,
-83, -85, -90, -97, -114, -999, -999, -999}},
/* 250 Hz */
{{-999, -999, -999, -999, -999, -999, -110, -105,
-100, -95, -90, -86, -80, -75, -75, -79,
-80, -79, -80, -81, -82, -88, -95, -103,
-110, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -108, -103, -98, -93,
-88, -83, -79, -78, -75, -71, -67, -68,
-73, -73, -72, -73, -75, -77, -80, -82,
-88, -93, -100, -107, -114, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -110, -105, -101, -96, -90,
-86, -81, -77, -73, -69, -66, -61, -62,
-66, -64, -62, -65, -66, -70, -72, -76,
-81, -80, -84, -90, -95, -102, -110, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -107, -103, -97, -92, -88,
-83, -79, -74, -70, -66, -59, -53, -58,
-62, -55, -54, -54, -54, -58, -61, -62,
-72, -70, -72, -75, -78, -80, -81, -80,
-83, -83, -88, -93, -100, -107, -115, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -105, -100, -95, -90, -85,
-80, -75, -70, -66, -62, -56, -48, -44,
-48, -46, -46, -43, -46, -48, -48, -51,
-58, -58, -59, -60, -62, -62, -61, -61,
-65, -64, -65, -68, -70, -74, -75, -78,
-81, -86, -95, -110, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -105, -100, -95, -90, -85, -80,
-75, -70, -65, -61, -55, -49, -39, -33,
-40, -35, -32, -38, -40, -33, -35, -37,
-46, -41, -45, -44, -46, -42, -45, -46,
-52, -50, -50, -50, -54, -54, -55, -57,
-62, -64, -66, -68, -70, -76, -81, -90,
-100, -110, -999, -999, -999, -999, -999, -999}},
/* 354 hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-105, -98, -90, -85, -82, -83, -80, -78,
-84, -79, -80, -83, -87, -89, -91, -93,
-99, -106, -117, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-105, -98, -90, -85, -80, -75, -70, -68,
-74, -72, -74, -77, -80, -82, -85, -87,
-92, -89, -91, -95, -100, -106, -112, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-105, -98, -90, -83, -75, -71, -63, -64,
-67, -62, -64, -67, -70, -73, -77, -81,
-84, -83, -85, -89, -90, -93, -98, -104,
-109, -114, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-103, -96, -88, -81, -75, -68, -58, -54,
-56, -54, -56, -56, -58, -60, -63, -66,
-74, -69, -72, -72, -75, -74, -77, -81,
-81, -82, -84, -87, -93, -96, -99, -104,
-110, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -108, -102, -96,
-91, -85, -80, -74, -68, -60, -51, -46,
-48, -46, -43, -45, -47, -47, -49, -48,
-56, -53, -55, -58, -57, -63, -58, -60,
-66, -64, -67, -70, -70, -74, -77, -84,
-86, -89, -91, -93, -94, -101, -109, -118,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -108, -103, -98, -93, -88,
-83, -78, -73, -68, -60, -53, -44, -35,
-38, -38, -34, -34, -36, -40, -41, -44,
-51, -45, -46, -47, -46, -54, -50, -49,
-50, -50, -50, -51, -54, -57, -58, -60,
-66, -66, -66, -64, -65, -68, -77, -82,
-87, -95, -110, -999, -999, -999, -999, -999}},
/* 500 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-107, -102, -97, -92, -87, -83, -78, -75,
-82, -79, -83, -85, -89, -92, -95, -98,
-101, -105, -109, -113, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -106,
-100, -95, -90, -86, -81, -78, -74, -69,
-74, -74, -76, -79, -83, -84, -86, -89,
-92, -97, -93, -100, -103, -107, -110, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -106, -100,
-95, -90, -87, -83, -80, -75, -69, -60,
-66, -66, -68, -70, -74, -78, -79, -81,
-81, -83, -84, -87, -93, -96, -99, -103,
-107, -110, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -108, -103, -98,
-93, -89, -85, -82, -78, -71, -62, -55,
-58, -58, -54, -54, -55, -59, -61, -62,
-70, -66, -66, -67, -70, -72, -75, -78,
-84, -84, -84, -88, -91, -90, -95, -98,
-102, -103, -106, -110, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -108, -103, -98, -94,
-90, -87, -82, -79, -73, -67, -58, -47,
-50, -45, -41, -45, -48, -44, -44, -49,
-54, -51, -48, -47, -49, -50, -51, -57,
-58, -60, -63, -69, -70, -69, -71, -74,
-78, -82, -90, -95, -101, -105, -110, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -105, -101, -97, -93, -90,
-85, -80, -77, -72, -65, -56, -48, -37,
-40, -36, -34, -40, -50, -47, -38, -41,
-47, -38, -35, -39, -38, -43, -40, -45,
-50, -45, -44, -47, -50, -55, -48, -48,
-52, -66, -70, -76, -82, -90, -97, -105,
-110, -999, -999, -999, -999, -999, -999, -999}},
/* 707 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -108, -103, -98, -93, -86, -79, -76,
-83, -81, -85, -87, -89, -93, -98, -102,
-107, -112, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -108, -103, -98, -93, -86, -79, -71,
-77, -74, -77, -79, -81, -84, -85, -90,
-92, -93, -92, -98, -101, -108, -112, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-108, -103, -98, -93, -87, -78, -68, -65,
-66, -62, -65, -67, -70, -73, -75, -78,
-82, -82, -83, -84, -91, -93, -98, -102,
-106, -110, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-105, -100, -95, -90, -82, -74, -62, -57,
-58, -56, -51, -52, -52, -54, -54, -58,
-66, -59, -60, -63, -66, -69, -73, -79,
-83, -84, -80, -81, -81, -82, -88, -92,
-98, -105, -113, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -107,
-102, -97, -92, -84, -79, -69, -57, -47,
-52, -47, -44, -45, -50, -52, -42, -42,
-53, -43, -43, -48, -51, -56, -55, -52,
-57, -59, -61, -62, -67, -71, -78, -83,
-86, -94, -98, -103, -110, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -105, -100,
-95, -90, -84, -78, -70, -61, -51, -41,
-40, -38, -40, -46, -52, -51, -41, -40,
-46, -40, -38, -38, -41, -46, -41, -46,
-47, -43, -43, -45, -41, -45, -56, -67,
-68, -83, -87, -90, -95, -102, -107, -113,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 1000 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -109, -105, -101, -96, -91, -84, -77,
-82, -82, -85, -89, -94, -100, -106, -110,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -106, -103, -98, -92, -85, -80, -71,
-75, -72, -76, -80, -84, -86, -89, -93,
-100, -107, -113, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -107,
-104, -101, -97, -92, -88, -84, -80, -64,
-66, -63, -64, -66, -69, -73, -77, -83,
-83, -86, -91, -98, -104, -111, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -107,
-104, -101, -97, -92, -90, -84, -74, -57,
-58, -52, -55, -54, -50, -52, -50, -52,
-63, -62, -69, -76, -77, -78, -78, -79,
-82, -88, -94, -100, -106, -111, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -106, -102,
-98, -95, -90, -85, -83, -78, -70, -50,
-50, -41, -44, -49, -47, -50, -50, -44,
-55, -46, -47, -48, -48, -54, -49, -49,
-58, -62, -71, -81, -87, -92, -97, -102,
-108, -114, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -106, -102,
-98, -95, -90, -85, -83, -78, -70, -45,
-43, -41, -47, -50, -51, -50, -49, -45,
-47, -41, -44, -41, -39, -43, -38, -37,
-40, -41, -44, -50, -58, -65, -73, -79,
-85, -92, -97, -101, -105, -109, -113, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 1414 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -107, -100, -95, -87, -81,
-85, -83, -88, -93, -100, -107, -114, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -107, -101, -95, -88, -83, -76,
-73, -72, -79, -84, -90, -95, -100, -105,
-110, -115, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -104, -98, -92, -87, -81, -70,
-65, -62, -67, -71, -74, -80, -85, -91,
-95, -99, -103, -108, -111, -114, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -103, -97, -90, -85, -76, -60,
-56, -54, -60, -62, -61, -56, -63, -65,
-73, -74, -77, -75, -78, -81, -86, -87,
-88, -91, -94, -98, -103, -110, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -105,
-100, -97, -92, -86, -81, -79, -70, -57,
-51, -47, -51, -58, -60, -56, -53, -50,
-58, -52, -50, -50, -53, -55, -64, -69,
-71, -85, -82, -78, -81, -85, -95, -102,
-112, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -105,
-100, -97, -92, -85, -83, -79, -72, -49,
-40, -43, -43, -54, -56, -51, -50, -40,
-43, -38, -36, -35, -37, -38, -37, -44,
-54, -60, -57, -60, -70, -75, -84, -92,
-103, -112, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 2000 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -102, -95, -89, -82,
-83, -84, -90, -92, -99, -107, -113, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -107, -101, -95, -89, -83, -72,
-74, -78, -85, -88, -88, -90, -92, -98,
-105, -111, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -109, -103, -97, -93, -87, -81, -70,
-70, -67, -75, -73, -76, -79, -81, -83,
-88, -89, -97, -103, -110, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -107, -100, -94, -88, -83, -75, -63,
-59, -59, -63, -66, -60, -62, -67, -67,
-77, -76, -81, -88, -86, -92, -96, -102,
-109, -116, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -105, -98, -92, -86, -81, -73, -56,
-52, -47, -55, -60, -58, -52, -51, -45,
-49, -50, -53, -54, -61, -71, -70, -69,
-78, -79, -87, -90, -96, -104, -112, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -103, -96, -90, -86, -78, -70, -51,
-42, -47, -48, -55, -54, -54, -53, -42,
-35, -28, -33, -38, -37, -44, -47, -49,
-54, -63, -68, -78, -82, -89, -94, -99,
-104, -109, -114, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 2828 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -110, -100, -90, -79,
-85, -81, -82, -82, -89, -94, -99, -103,
-109, -115, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -105, -97, -85, -72,
-74, -70, -70, -70, -76, -85, -91, -93,
-97, -103, -109, -115, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -112, -93, -81, -68,
-62, -60, -60, -57, -63, -70, -77, -82,
-90, -93, -98, -104, -109, -113, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -113, -100, -93, -84, -63,
-58, -48, -53, -54, -52, -52, -57, -64,
-66, -76, -83, -81, -85, -85, -90, -95,
-98, -101, -103, -106, -108, -111, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -105, -95, -86, -74, -53,
-50, -38, -43, -49, -43, -42, -39, -39,
-46, -52, -57, -56, -72, -69, -74, -81,
-87, -92, -94, -97, -99, -102, -105, -108,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -108, -99, -90, -76, -66, -45,
-43, -41, -44, -47, -43, -47, -40, -30,
-31, -31, -39, -33, -40, -41, -43, -53,
-59, -70, -73, -77, -79, -82, -84, -87,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 4000 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -91, -76,
-75, -85, -93, -98, -104, -110, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -91, -70,
-70, -75, -86, -89, -94, -98, -101, -106,
-110, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -110, -95, -80, -60,
-65, -64, -74, -83, -88, -91, -95, -99,
-103, -107, -110, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -110, -95, -80, -58,
-55, -49, -66, -68, -71, -78, -78, -80,
-88, -85, -89, -97, -100, -105, -110, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -110, -95, -80, -53,
-52, -41, -59, -59, -49, -58, -56, -63,
-86, -79, -90, -93, -98, -103, -107, -112,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -97, -91, -73, -45,
-40, -33, -53, -61, -49, -54, -50, -50,
-60, -52, -67, -74, -81, -92, -96, -100,
-105, -110, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 5657 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -113, -106, -99, -92, -77,
-80, -88, -97, -106, -115, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -116, -109, -102, -95, -89, -74,
-72, -88, -87, -95, -102, -109, -116, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -116, -109, -102, -95, -89, -75,
-66, -74, -77, -78, -86, -87, -90, -96,
-105, -115, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -115, -108, -101, -94, -88, -66,
-56, -61, -70, -65, -78, -72, -83, -84,
-93, -98, -105, -110, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -110, -105, -95, -89, -82, -57,
-52, -52, -59, -56, -59, -58, -69, -67,
-88, -82, -82, -89, -94, -100, -108, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -110, -101, -96, -90, -83, -77, -54,
-43, -38, -50, -48, -52, -48, -42, -42,
-51, -52, -53, -59, -65, -71, -78, -85,
-95, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 8000 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -120, -105, -86, -68,
-78, -79, -90, -100, -110, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -120, -105, -86, -66,
-73, -77, -88, -96, -105, -115, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -120, -105, -92, -80, -61,
-64, -68, -80, -87, -92, -100, -110, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -120, -104, -91, -79, -52,
-60, -54, -64, -69, -77, -80, -82, -84,
-85, -87, -88, -90, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -118, -100, -87, -77, -49,
-50, -44, -58, -61, -61, -67, -65, -62,
-62, -62, -65, -68, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -115, -98, -84, -62, -49,
-44, -38, -46, -49, -49, -46, -39, -37,
-39, -40, -42, -43, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 11314 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -88, -74,
-77, -82, -82, -85, -90, -94, -99, -104,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -88, -66,
-70, -81, -80, -81, -84, -88, -91, -93,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -88, -61,
-63, -70, -71, -74, -77, -80, -83, -85,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -110, -86, -62,
-63, -62, -62, -58, -52, -50, -50, -52,
-54, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -118, -108, -84, -53,
-50, -50, -50, -55, -47, -45, -40, -40,
-40, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -118, -100, -73, -43,
-37, -42, -43, -53, -38, -37, -35, -35,
-38, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}},
/* 16000 Hz */
{{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -91, -84, -74,
-80, -80, -80, -80, -80, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -91, -84, -74,
-68, -68, -68, -68, -68, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -86, -78, -70,
-60, -45, -30, -21, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -87, -78, -67,
-48, -38, -29, -21, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -86, -69, -56,
-45, -35, -33, -29, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999},
{-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -110, -100, -83, -71, -48,
-27, -38, -37, -34, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999,
-999, -999, -999, -999, -999, -999, -999, -999}}
};
#endif

564
src/add-ons/media/plugins/vorbis/libvorbis/mdct.c Normal file
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@ -0,0 +1,564 @@
/********************************************************************
* *
* 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: normalized modified discrete cosine transform
power of two length transform only [64 <= n ]
last mod: $Id: mdct.c,v 1.1 2003/12/13 20:17:56 shatty Exp $
Original algorithm adapted long ago from _The use of multirate filter
banks for coding of high quality digital audio_, by T. Sporer,
K. Brandenburg and B. Edler, collection of the European Signal
Processing Conference (EUSIPCO), Amsterdam, June 1992, Vol.1, pp
211-214
The below code implements an algorithm that no longer looks much like
that presented in the paper, but the basic structure remains if you
dig deep enough to see it.
This module DOES NOT INCLUDE code to generate/apply the window
function. Everybody has their own weird favorite including me... I
happen to like the properties of y=sin(2PI*sin^2(x)), but others may
vehemently disagree.
********************************************************************/
/* this can also be run as an integer transform by uncommenting a
define in mdct.h; the integerization is a first pass and although
it's likely stable for Vorbis, the dynamic range is constrained and
roundoff isn't done (so it's noisy). Consider it functional, but
only a starting point. There's no point on a machine with an FPU */
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#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<n/4;i++){
T[i*2]=FLOAT_CONV(cos((M_PI/n)*(4*i)));
T[i*2+1]=FLOAT_CONV(-sin((M_PI/n)*(4*i)));
T[n2+i*2]=FLOAT_CONV(cos((M_PI/(2*n))*(2*i+1)));
T[n2+i*2+1]=FLOAT_CONV(sin((M_PI/(2*n))*(2*i+1)));
}
for(i=0;i<n/8;i++){
T[n+i*2]=FLOAT_CONV(cos((M_PI/n)*(4*i+2))*.5);
T[n+i*2+1]=FLOAT_CONV(-sin((M_PI/n)*(4*i+2))*.5);
}
/* bitreverse lookup... */
{
int mask=(1<<(log2n-1))-1,i,j;
int msb=1<<(log2n-2);
for(i=0;i<n/8;i++){
int acc=0;
for(j=0;msb>>j;j++)
if((msb>>j)&i)acc|=1<<j;
bitrev[i*2]=((~acc)&mask)-1;
bitrev[i*2+1]=acc;
}
}
lookup->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++)
mdct_butterfly_generic(T,x+(points>>i)*j,points>>i,4<<i);
}
for(j=0;j<points;j+=32)
mdct_butterfly_32(x+j);
}
void mdct_clear(mdct_lookup *l){
if(l){
if(l->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(w0<w1);
}
void mdct_backward(mdct_lookup *init, DATA_TYPE *in, DATA_TYPE *out){
int n=init->n;
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(iX<oX1);
iX=out+n2+n4;
oX1=out+n4;
oX2=oX1;
do{
oX1-=4;
iX-=4;
oX2[0] = -(oX1[3] = iX[3]);
oX2[1] = -(oX1[2] = iX[2]);
oX2[2] = -(oX1[1] = iX[1]);
oX2[3] = -(oX1[0] = iX[0]);
oX2+=4;
}while(oX2<iX);
iX=out+n2+n4;
oX1=out+n2+n4;
oX2=out+n2;
do{
oX1-=4;
oX1[0]= iX[3];
oX1[1]= iX[2];
oX1[2]= iX[1];
oX1[3]= iX[0];
iX+=4;
}while(oX1>oX2);
}
}
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;i<n8;i+=2){
x0 -=4;
T-=2;
r0= x0[2] + x1[0];
r1= x0[0] + x1[2];
w2[i]= MULT_NORM(r1*T[1] + r0*T[0]);
w2[i+1]= MULT_NORM(r1*T[0] - r0*T[1]);
x1 +=4;
}
x1=in+1;
for(;i<n2-n8;i+=2){
T-=2;
x0 -=4;
r0= x0[2] - x1[0];
r1= x0[0] - x1[2];
w2[i]= MULT_NORM(r1*T[1] + r0*T[0]);
w2[i+1]= MULT_NORM(r1*T[0] - r0*T[1]);
x1 +=4;
}
x0=in+n;
for(;i<n2;i+=2){
T-=2;
x0 -=4;
r0= -x0[2] - x1[0];
r1= -x0[0] - x1[2];
w2[i]= MULT_NORM(r1*T[1] + r0*T[0]);
w2[i+1]= MULT_NORM(r1*T[0] - r0*T[1]);
x1 +=4;
}
mdct_butterflies(init,w+n2,n2);
mdct_bitreverse(init,w);
/* roatate + window */
T=init->trig+n2;
x0=out+n2;
for(i=0;i<n4;i++){
x0--;
out[i] =MULT_NORM((w[0]*T[0]+w[1]*T[1])*init->scale);
x0[0] =MULT_NORM((w[0]*T[1]-w[1]*T[0])*init->scale);
w+=2;
T+=2;
}
}

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/********************************************************************
* *
* 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)+.5))
#define MULT_NORM(x) ((x)>>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

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/********************************************************************
* *
* 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

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#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 <math.h>
#include <ogg/os_types.h>
#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 <malloc.h>
# 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 <alloca.h>
#endif
#ifdef USE_MEMORY_H
# include <memory.h>
#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 */

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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 <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <math.h>
#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;j<n;j++){
if(dB && v[j]==0)
fprintf(of,"\n\n");
else{
if(bark)
fprintf(of,"%g ",toBARK(22050.f*j/n));
else
fprintf(of,"%g ",(float)j);
if(dB){
fprintf(of,"%g\n",todB(v+j));
}else{
fprintf(of,"%g\n",v[j]);
}
}
}
fclose(of);
}
}
long frameno=0;
/****************************************************************/
int main(int argc,char *argv[]){
int eos=0;
float nonz=0.f;
float acc=0.f;
float tot=0.f;
float ampmax=-9999,newmax;
float local_ampmax[2];
int framesize=2048;
float ampmax_att_per_sec=-6.;
float *pcm[2],*out[2],*window,*flr[2],*mask[2],*work[2];
signed char *buffer,*buffer2;
mdct_lookup m_look;
drft_lookup f_look;
vorbis_look_psy p_look;
vorbis_look_psy_global *pg_look;
vorbis_look_floor *floor_look;
vorbis_info vi;
long i,j,k;
int ath=0;
int decayp=0;
argv++;
while(*argv){
if(*argv[0]=='-'){
/* option */
if(argv[0][1]=='v'){
noisy=0;
}
}else
if(*argv[0]=='+'){
/* option */
if(argv[0][1]=='v'){
noisy=1;
}
}else
framesize=atoi(argv[0]);
argv++;
}
vi.channels=2;
vi.codec_setup=&codec_setup0;
pcm[0]=_ogg_malloc(framesize*sizeof(float));
pcm[1]=_ogg_malloc(framesize*sizeof(float));
out[0]=_ogg_calloc(framesize/2,sizeof(float));
out[1]=_ogg_calloc(framesize/2,sizeof(float));
work[0]=_ogg_calloc(framesize,sizeof(float));
work[1]=_ogg_calloc(framesize,sizeof(float));
flr[0]=_ogg_calloc(framesize/2,sizeof(float));
flr[1]=_ogg_calloc(framesize/2,sizeof(float));
buffer=_ogg_malloc(framesize*4);
buffer2=buffer+framesize*2;
window=_vorbis_window(0,framesize,framesize/2,framesize/2);
mdct_init(&m_look,framesize);
drft_init(&f_look,framesize);
_vp_psy_init(&p_look,&_psy_set0,&_psy_set0G,framesize/2,44100);
pg_look=_vp_global_look(&vi);
floor_look=_floor_P[1]->look(NULL,NULL,&_floor_set0);
/* we cheat on the WAV header; we just bypass 44 bytes and never
verify that it matches 16bit/stereo/44.1kHz. */
fread(buffer,1,44,stdin);
fwrite(buffer,1,44,stdout);
memset(buffer,0,framesize*2);
analysis("window",0,window,framesize,0,0);
fprintf(stderr,"Processing for frame size %d...\n",framesize);
while(!eos){
long bytes=fread(buffer2,1,framesize*2,stdin);
if(bytes<framesize*2)
memset(buffer2+bytes,0,framesize*2-bytes);
if(bytes!=0){
int nonzero[2];
/* uninterleave samples */
for(i=0;i<framesize;i++){
pcm[0][i]=((buffer[i*4+1]<<8)|
(0x00ff&(int)buffer[i*4]))/32768.f;
pcm[1][i]=((buffer[i*4+3]<<8)|
(0x00ff&(int)buffer[i*4+2]))/32768.f;
}
{
float secs=framesize/44100.;
ampmax+=secs*ampmax_att_per_sec;
if(ampmax<-9999)ampmax=-9999;
}
for(i=0;i<2;i++){
float scale=4.f/framesize;
float *fft=work[i];
float *mdct=pcm[i];
float *logmdct=mdct+framesize/2;
analysis("pre",frameno+i,pcm[i],framesize,0,0);
/* fft and mdct transforms */
for(j=0;j<framesize;j++)
fft[j]=pcm[i][j]*=window[j];
drft_forward(&f_look,fft);
local_ampmax[i]=-9999.f;
fft[0]*=scale;
fft[0]=todB(fft);
for(j=1;j<framesize-1;j+=2){
float temp=scale*FAST_HYPOT(fft[j],fft[j+1]);
temp=fft[(j+1)>>1]=todB(&temp);
if(temp>local_ampmax[i])local_ampmax[i]=temp;
}
if(local_ampmax[i]>ampmax)ampmax=local_ampmax[i];
mdct_forward(&m_look,pcm[i],mdct);
for(j=0;j<framesize/2;j++)
logmdct[j]=todB(mdct+j);
analysis("mdct",frameno+i,logmdct,framesize/2,1,0);
analysis("fft",frameno+i,fft,framesize/2,1,0);
}
for(i=0;i<2;i++){
float amp;
float *fft=work[i];
float *logmax=fft;
float *mdct=pcm[i];
float *logmdct=mdct+framesize/2;
float *mask=fft+framesize/2;
/* floor psychoacoustics */
_vp_compute_mask(&p_look,
pg_look,
i,
fft,
logmdct,
mask,
ampmax,
local_ampmax[i],
framesize/2);
analysis("mask",frameno+i,mask,framesize/2,1,0);
{
vorbis_block vb;
vorbis_dsp_state vd;
memset(&vd,0,sizeof(vd));
vd.vi=&vi;
vb.vd=&vd;
vb.pcmend=framesize;
/* floor quantization/application */
nonzero[i]=_floor_P[1]->forward(&vb,floor_look,
mdct,
logmdct,
mask,
logmax,
flr[i]);
}
_vp_remove_floor(&p_look,
pg_look,
logmdct,
mdct,
flr[i],
pcm[i],
local_ampmax[i]);
for(j=0;j<framesize/2;j++)
if(fabs(pcm[i][j])>1500)
fprintf(stderr,"%ld ",frameno+i);
analysis("res",frameno+i,pcm[i],framesize/2,1,0);
analysis("codedflr",frameno+i,flr[i],framesize/2,1,1);
}
/* residue prequantization */
_vp_partition_prequant(&p_look,
&vi,
pcm,
nonzero);
for(i=0;i<2;i++)
analysis("quant",frameno+i,pcm[i],framesize/2,1,0);
/* channel coupling / stereo quantization */
_vp_couple(&p_look,
&mapping_info,
pcm,
nonzero);
for(i=0;i<2;i++)
analysis("coupled",frameno+i,pcm[i],framesize/2,1,0);
/* decoupling */
for(i=mapping_info.coupling_steps-1;i>=0;i--){
float *pcmM=pcm[mapping_info.coupling_mag[i]];
float *pcmA=pcm[mapping_info.coupling_ang[i]];
for(j=0;j<framesize/2;j++){
float mag=pcmM[j];
float ang=pcmA[j];
if(mag>0)
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag-ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag+ang;
}
else
if(ang>0){
pcmM[j]=mag;
pcmA[j]=mag+ang;
}else{
pcmA[j]=mag;
pcmM[j]=mag-ang;
}
}
}
for(i=0;i<2;i++)
analysis("decoupled",frameno+i,pcm[i],framesize/2,1,0);
for(i=0;i<2;i++){
float amp;
for(j=0;j<framesize/2;j++)
pcm[i][j]*=flr[i][j];
analysis("final",frameno+i,pcm[i],framesize/2,1,1);
/* take it back to time */
mdct_backward(&m_look,pcm[i],pcm[i]);
for(j=0;j<framesize/2;j++)
out[i][j]+=pcm[i][j]*window[j];
analysis("out",frameno+i,out[i],framesize/2,0,0);
}
/* write data. Use the part of buffer we're about to shift out */
for(i=0;i<2;i++){
char *ptr=buffer+i*2;
float *mono=out[i];
int flag=0;
for(j=0;j<framesize/2;j++){
int val=mono[j]*32767.;
/* might as well guard against clipping */
if(val>32767){
if(!flag)fprintf(stderr,"clipping in frame %ld ",frameno+i);
flag=1;
val=32767;
}
if(val<-32768){
if(!flag)fprintf(stderr,"clipping in frame %ld ",frameno+i);
flag=1;
val=-32768;
}
ptr[0]=val&0xff;
ptr[1]=(val>>8)&0xff;
ptr+=4;
}
}
fprintf(stderr,"*");
fwrite(buffer,1,framesize*2,stdout);
memmove(buffer,buffer2,framesize*2);
for(i=0;i<2;i++){
for(j=0,k=framesize/2;j<framesize/2;j++,k++)
out[i][j]=pcm[i][k]*window[k];
}
frameno+=2;
}else
eos=1;
}
fprintf(stderr,"average raw bits of entropy: %.03g/sample\n",acc/tot);
fprintf(stderr,"average nonzero samples: %.03g/%d\n",nonz/tot*framesize/2,
framesize/2);
fprintf(stderr,"Done\n\n");
return 0;
}

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/********************************************************************
* *
* 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: registry for time, floor, res backends and channel mappings
last mod: $Id: registry.c,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#include "vorbis/codec.h"
#include "codec_internal.h"
#include "registry.h"
#include "misc.h"
/* seems like major overkill now; the backend numbers will grow into
the infrastructure soon enough */
extern vorbis_func_floor floor0_exportbundle;
extern vorbis_func_floor floor1_exportbundle;
extern vorbis_func_residue residue0_exportbundle;
extern vorbis_func_residue residue1_exportbundle;
extern vorbis_func_residue residue2_exportbundle;
extern vorbis_func_mapping mapping0_exportbundle;
vorbis_func_floor *_floor_P[]={
&floor0_exportbundle,
&floor1_exportbundle,
};
vorbis_func_residue *_residue_P[]={
&residue0_exportbundle,
&residue1_exportbundle,
&residue2_exportbundle,
};
vorbis_func_mapping *_mapping_P[]={
&mapping0_exportbundle,
};

32
src/add-ons/media/plugins/vorbis/libvorbis/registry.h Normal file
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@ -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: registry for time, floor, res backends and channel mappings
last mod: $Id: registry.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#ifndef _V_REG_H_
#define _V_REG_H_
#define VI_TRANSFORMB 1
#define VI_WINDOWB 1
#define VI_TIMEB 1
#define VI_FLOORB 2
#define VI_RESB 3
#define VI_MAPB 1
extern vorbis_func_floor *_floor_P[];
extern vorbis_func_residue *_residue_P[];
extern vorbis_func_mapping *_mapping_P[];
#endif

906
src/add-ons/media/plugins/vorbis/libvorbis/res0.c Normal file
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/********************************************************************
* *
* 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: residue backend 0, 1 and 2 implementation
last mod: $Id: res0.c,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
/* Slow, slow, slow, simpleminded and did I mention it was slow? The
encode/decode loops are coded for clarity and performance is not
yet even a nagging little idea lurking in the shadows. Oh and BTW,
it's slow. */
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include <ogg/ogg.h>
#include "vorbis/codec.h"
#include "codec_internal.h"
#include "registry.h"
#include "codebook.h"
#include "misc.h"
#include "os.h"
#ifdef TRAIN_RES
#include <stdio.h>
#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;j<look->parts;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;l<statebook->entries;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;j<look->parts;j++){
long acc=0;
fprintf(stderr,"\t[%d] == ",j);
for(k=0;k<look->stages;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;j<look->parts;j++)
if(look->partbooks[j])_ogg_free(look->partbooks[j]);
_ogg_free(look->partbooks);
for(j=0;j<look->partvals;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;j<info->partitions;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;j<acc;j++)
oggpack_write(opb,info->booklist[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;j<info->partitions;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;j<acc;j++)
info->booklist[j]=oggpack_read(opb,8);
if(info->groupbook>=ci->books)goto errout;
for(j=0;j<acc;j++)
if(info->booklist[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;j<look->parts;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;k<stages;k++)
if(info->secondstages[j]&(1<<k)){
look->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;j<look->partvals;j++){
long val=j;
long mult=look->partvals/look->parts;
look->decodemap[j]=_ogg_malloc(dim*sizeof(*look->decodemap[j]));
for(k=0;k<dim;k++){
long deco=val/mult;
val-=deco*mult;
mult/=look->parts;
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;k<dim;++k){
float val=a[--o];
i=tt->threshvals>>1;
if(val<tt->quantthresh[i]){
if(val<tt->quantthresh[i-1]){
for(--i;i>0;--i)
if(val>=tt->quantthresh[i-1])
break;
}
}else{
for(++i;i<tt->threshvals-1;++i)
if(val<tt->quantthresh[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;i<book->entries;i++){
if(c->lengthlist[i]>0){
float this=0.f;
for(j=0;j<dim;j++){
float val=(e[j]-a[j]);
this+=val*val;
}
if(best==-1 || this<bestf){
bestf=this;
best=i;
}
}
e++;
}
}
{
float *ptr=book->valuelist+best*dim;
for(i=0;i<dim;i++)
*a++ -= *ptr++;
}
return(best);
}
static int _encodepart(oggpack_buffer *opb,float *vec, int n,
codebook *book,long *acc){
int i,bits=0;
int dim=book->dim;
int step=n/dim;
for(i=0;i<step;i++){
int entry=local_book_besterror(book,vec+i*dim);
#ifdef TRAIN_RES
acc[entry]++;
#endif
bits+=vorbis_book_encode(book,entry,opb);
}
return(bits);
}
static long **_01class(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int ch){
long i,j,k;
vorbis_look_residue0 *look=(vorbis_look_residue0 *)vl;
vorbis_info_residue0 *info=look->info;
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;i<ch;i++){
partword[i]=_vorbis_block_alloc(vb,n/samples_per_partition*sizeof(*partword[i]));
memset(partword[i],0,n/samples_per_partition*sizeof(*partword[i]));
}
for(i=0;i<partvals;i++){
int offset=i*samples_per_partition+info->begin;
for(j=0;j<ch;j++){
float max=0.;
float ent=0.;
for(k=0;k<samples_per_partition;k++){
if(fabs(in[j][offset+k])>max)max=fabs(in[j][offset+k]);
ent+=fabs(rint(in[j][offset+k]));
}
ent*=scale;
for(k=0;k<possible_partitions-1;k++)
if(max<=info->classmetric1[k] &&
(info->classmetric2[k]<0 || (int)ent<info->classmetric2[k]))
break;
partword[j][i]=k;
}
}
#ifdef TRAIN_RESAUX
{
FILE *of;
char buffer[80];
for(i=0;i<ch;i++){
sprintf(buffer,"resaux_%d.vqd",look->train_seq);
of=fopen(buffer,"a");
for(j=0;j<partvals;j++)
fprintf(of,"%ld, ",partword[i][j]);
fprintf(of,"\n");
fclose(of);
}
}
#endif
look->frames++;
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;i<partvals;i++){
float magmax=0.f;
float angmax=0.f;
for(j=0;j<samples_per_partition;j+=ch){
if(fabs(in[0][l])>magmax)magmax=fabs(in[0][l]);
for(k=1;k<ch;k++)
if(fabs(in[k][l])>angmax)angmax=fabs(in[k][l]);
l++;
}
for(j=0;j<possible_partitions-1;j++)
if(magmax<=info->classmetric1[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;i<partvals;i++)
fprintf(of,"%ld, ",partword[0][i]);
fprintf(of,"\n");
fclose(of);
#endif
look->frames++;
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;i<ch;i++)
for(j=info->begin;j<info->end;j++){
if(in[i][j]>look->tmax)look->tmax=in[i][j];
if(in[i][j]<look->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;s<look->stages;s++){
for(i=0;i<partvals;){
/* first we encode a partition codeword for each channel */
if(s==0){
for(j=0;j<ch;j++){
long val=partword[j][i];
for(k=1;k<partitions_per_word;k++){
val*=possible_partitions;
if(i+k<partvals)
val+=partword[j][i+k];
}
/* training hack */
if(val<look->phrasebook->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;k<partitions_per_word && i<partvals;k++,i++){
long offset=i*samples_per_partition+info->begin;
for(j=0;j<ch;j++){
if(s==0)resvals[partword[j][i]]+=samples_per_partition;
if(info->secondstages[partword[j][i]]&(1<<s)){
codebook *statebook=look->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;l<samples_per_partition;l++){
if(samples[l]<look->training_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;k<possible_partitions;k++){
fprintf(stderr,"%ld/%1.2g, ",resvals[k],(float)resbits[k]/resvals[k]);
total+=resvals[k];
totalbits+=resbits[k];
}
fprintf(stderr,":: %ld:%1.2g\n",total,(double)totalbits/total);
}*/
return(0);
}
/* a truncated packet here just means 'stop working'; it's not an error */
static int _01inverse(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int ch,
long (*decodepart)(codebook *, float *,
oggpack_buffer *,int)){
long i,j,k,l,s;
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 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;j<ch;j++)
partword[j]=_vorbis_block_alloc(vb,partwords*sizeof(*partword[j]));
for(s=0;s<look->stages;s++){
/* each loop decodes on partition codeword containing
partitions_pre_word partitions */
for(i=0,l=0;i<partvals;l++){
if(s==0){
/* fetch the partition word for each channel */
for(j=0;j<ch;j++){
int temp=vorbis_book_decode(look->phrasebook,&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;k<partitions_per_word && i<partvals;k++,i++)
for(j=0;j<ch;j++){
long offset=info->begin+i*samples_per_partition;
if(info->secondstages[partword[j][l][k]]&(1<<s)){
codebook *stagebook=look->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;i<ch;i++)
if(nonzero[i])
in[used++]=in[i];
if(used)
/*return(_01class(vb,vl,in,used,_interleaved_testhack));*/
return(_01class(vb,vl,in,used));
else
return(0);
}
int res0_forward(vorbis_block *vb,vorbis_look_residue *vl,
float **in,float **out,int *nonzero,int ch,
long **partword){
/* we encode only the nonzero parts of a bundle */
int i,j,used=0,n=vb->pcmend/2;
for(i=0;i<ch;i++)
if(nonzero[i]){
if(out)
for(j=0;j<n;j++)
out[i][j]+=in[i][j];
in[used++]=in[i];
}
if(used){
int ret=_01forward(vb,vl,in,used,partword,
_interleaved_encodepart);
if(out){
used=0;
for(i=0;i<ch;i++)
if(nonzero[i]){
for(j=0;j<n;j++)
out[i][j]-=in[used][j];
used++;
}
}
return(ret);
}else{
return(0);
}
}
#endif
int res0_inverse(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int *nonzero,int ch){
int i,used=0;
for(i=0;i<ch;i++)
if(nonzero[i])
in[used++]=in[i];
if(used)
return(_01inverse(vb,vl,in,used,vorbis_book_decodevs_add));
else
return(0);
}
int res1_forward(vorbis_block *vb,vorbis_look_residue *vl,
float **in,float **out,int *nonzero,int ch,
long **partword){
int i,j,used=0,n=vb->pcmend/2;
for(i=0;i<ch;i++)
if(nonzero[i]){
if(out)
for(j=0;j<n;j++)
out[i][j]+=in[i][j];
in[used++]=in[i];
}
if(used){
int ret=_01forward(vb,vl,in,used,partword,_encodepart);
if(out){
used=0;
for(i=0;i<ch;i++)
if(nonzero[i]){
for(j=0;j<n;j++)
out[i][j]-=in[used][j];
used++;
}
}
return(ret);
}else{
return(0);
}
}
long **res1_class(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int *nonzero,int ch){
int i,used=0;
for(i=0;i<ch;i++)
if(nonzero[i])
in[used++]=in[i];
if(used)
return(_01class(vb,vl,in,used));
else
return(0);
}
int res1_inverse(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int *nonzero,int ch){
int i,used=0;
for(i=0;i<ch;i++)
if(nonzero[i])
in[used++]=in[i];
if(used)
return(_01inverse(vb,vl,in,used,vorbis_book_decodev_add));
else
return(0);
}
long **res2_class(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int *nonzero,int ch){
int i,used=0;
for(i=0;i<ch;i++)
if(nonzero[i])used++;
if(used)
return(_2class(vb,vl,in,ch));
else
return(0);
}
/* res2 is slightly more different; all the channels are interleaved
into a single vector and encoded. */
int res2_forward(vorbis_block *vb,vorbis_look_residue *vl,
float **in,float **out,int *nonzero,int ch,
long **partword){
long i,j,k,n=vb->pcmend/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;i<ch;i++){
float *pcm=in[i];
if(nonzero[i])used++;
for(j=0,k=i;j<n;j++,k+=ch)
work[k]=pcm[j];
}
if(used){
int ret=_01forward(vb,vl,&work,1,partword,_encodepart);
/* update the sofar vector */
if(out){
for(i=0;i<ch;i++){
float *pcm=in[i];
float *sofar=out[i];
for(j=0,k=i;j<n;j++,k+=ch)
sofar[j]+=pcm[j]-work[k];
}
}
return(ret);
}else{
return(0);
}
}
/* duplicate code here as speed is somewhat more important */
int res2_inverse(vorbis_block *vb,vorbis_look_residue *vl,
float **in,int *nonzero,int ch){
long i,k,l,s;
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 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;i<ch;i++)if(nonzero[i])break;
if(i==ch)return(0); /* no nonzero vectors */
for(s=0;s<look->stages;s++){
for(i=0,l=0;i<partvals;l++){
if(s==0){
/* fetch the partition word */
int temp=vorbis_book_decode(look->phrasebook,&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;k<partitions_per_word && i<partvals;k++,i++)
if(info->secondstages[partword[l][k]]&(1<<s)){
codebook *stagebook=look->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
};

88
src/add-ons/media/plugins/vorbis/libvorbis/scales.h Normal file
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@ -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 <math.h>
#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

733
src/add-ons/media/plugins/vorbis/libvorbis/sharedbook.c Normal file
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/********************************************************************
* *
* 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 <stdlib.h>
#include <math.h>
#include <string.h>
#include <ogg/ogg.h>
#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;
return(sign|exp|mant);
}
float _float32_unpack(long val){
double mant=val&0x1fffff;
int sign=val&0x80000000;
long exp =(val&0x7fe00000L)>>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;i<n;i++){
long length=l[i];
if(length>0){
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<n;i++){
ogg_uint32_t temp=0;
for(j=0;j<l[i];j++){
temp<<=1;
temp|=(r[count]>>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;i<b->dim;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*dim<entries), which means that
we'll have 'left over' entries; left over entries use zeroed
values (and are wasted). So don't generate codebooks like
that */
quantvals=_book_maptype1_quantvals(b);
for(j=0;j<b->entries;j++){
if((sparsemap && b->lengthlist[j]) || !sparsemap){
float last=0.f;
int indexdiv=1;
for(k=0;k<b->dim;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;j<b->entries;j++){
if((sparsemap && b->lengthlist[j]) || !sparsemap){
float last=0.f;
for(k=0;k<b->dim;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;i<s->entries;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;i<n;i++){
codes[i]=bitreverse(codes[i]);
codep[i]=codes+i;
}
qsort(codep,n,sizeof(*codep),sort32a);
sortindex=alloca(n*sizeof(*sortindex));
c->codelist=_ogg_malloc(n*sizeof(*c->codelist));
/* the index is a reverse index */
for(i=0;i<n;i++){
int position=codep[i]-codes;
sortindex[position]=i;
}
for(i=0;i<n;i++)
c->codelist[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;i<s->entries;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;i<s->entries;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<<c->dec_firsttablen;
c->dec_firsttable=_ogg_calloc(tabn,sizeof(*c->dec_firsttable));
c->dec_maxlength=0;
for(i=0;i<n;i++){
if(c->dec_maxlength<c->dec_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<<c->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;i<tabn;i++){
ogg_uint32_t word=i<<(32-c->dec_firsttablen);
if(c->dec_firsttable[bitreverse(word)]==0){
while((lo+1)<n && c->codelist[lo+1]<=word)lo++;
while( hi<n && word>=(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;i<el;i++){
float val=(ref[i]-b[i*step]);
acc+=val*val;
}
return(acc);
}
int _best(codebook *book, float *a, int step){
encode_aux_threshmatch *tt=book->c->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);k<dim;k++,o-=step){
i=tt->threshvals>>1;
if(a[o]<tt->quantthresh[i]){
for(;i>0;i--)
if(a[o]>=tt->quantthresh[i-1])
break;
}else{
for(i++;i<tt->threshvals-1;i++)
if(a[o]<tt->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;k<dim;k++,o+=step){
pv=(int)((a[o]-qlast-pt->min)/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);k<dim;k++,o-=step){
int pv=(int)((a[o]-pt->min)/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;i<pt->fitlength[entry];i++){
float this=_dist(dim,book->valuelist+list[i]*dim,a,step);
if(besti==-1 || this<best){
best=this;
besti=list[i];
}
}
return(besti);
}
}
if(nt){
/* optimized using the decision tree */
while(1){
float c=0.f;
float *p=book->valuelist+nt->p[ptr];
float *q=book->valuelist+nt->q[ptr];
for(k=0,o=0;k<dim;k++,o+=step)
c+=(p[k]-q[k])*(a[o]-(p[k]+q[k])*.5);
if(c>0.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;i<book->entries;i++){
if(c->lengthlist[i]>0){
float this=_dist(dim,e,a,step);
if(besti==-1 || this<best){
best=this;
besti=i;
}
}
e+=dim;
}
/*if(savebest!=-1 && savebest!=besti){
fprintf(stderr,"brute force/pigeonhole disagreement:\n"
"original:");
for(i=0;i<dim*step;i+=step)fprintf(stderr,"%g,",a[i]);
fprintf(stderr,"\n"
"pigeonhole (entry %d, err %g):",savebest,saverr);
for(i=0;i<dim;i++)fprintf(stderr,"%g,",
(book->valuelist+savebest*dim)[i]);
fprintf(stderr,"\n"
"bruteforce (entry %d, err %g):",besti,best);
for(i=0;i<dim;i++)fprintf(stderr,"%g,",
(book->valuelist+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 <stdio.h>
/* 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;i<b->entries*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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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 <stdio.h>
#include <ogg/ogg.h>
#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;i<vi->channels;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]);
}

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#include <stdlib.h>
#include <stdio.h>
#include <math.h>
#include <string.h>
void usage(){
fprintf(stderr,"tone <frequency_Hz>,[<amplitude>] [<frequency_Hz>,[<amplitude>]...]\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;j<argc-1;j++)
val+=amp[j]*sin(i/44100.f*f[j]*2*M_PI);
ival=rint(val);
if(ival>32767.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);
}

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/********************************************************************
* *
* 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 <ogg/ogg.h>
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

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/********************************************************************
* *
* 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 <stdlib.h>
#include <math.h>
#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<left;i++){
float x=(i+.5f)/left*M_PI/2.;
x=sin(x);
x*=x;
x*=M_PI/2.f;
x=sin(x);
ret[i]=x;
}
}
break;
default:
_ogg_free(ret);
return(NULL);
}
return(ret);
}
void _vorbis_apply_window(float *d,float *window[2],long *blocksizes,
int lW,int W,int nW){
lW=(W?lW:0);
nW=(W?nW:0);
{
long n=blocksizes[W];
long ln=blocksizes[lW];
long rn=blocksizes[nW];
long leftbegin=n/4-ln/4;
long leftend=leftbegin+ln/2;
long rightbegin=n/2+n/4-rn/4;
long rightend=rightbegin+rn/2;
int i,p;
for(i=0;i<leftbegin;i++)
d[i]=0.f;
for(p=0;i<leftend;i++,p++)
d[i]*=window[lW][p];
for(i=rightbegin,p=rn/2-1;i<rightend;i++,p--)
d[i]*=window[nW][p];
for(;i<n;i++)
d[i]=0.f;
}
}

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src/add-ons/media/plugins/vorbis/libvorbis/window.h Normal file
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/********************************************************************
* *
* 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.h,v 1.1 2003/12/13 20:17:56 shatty Exp $
********************************************************************/
#ifndef _V_WINDOW_
#define _V_WINDOW_
extern float *_vorbis_window(int type,int left);
extern void _vorbis_apply_window(float *d,float *window[2],long *blocksizes,
int lW,int W,int nW);
#endif

207
src/add-ons/media/plugins/vorbis/vorbisCodecPlugin.cpp Normal file
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#include <stdio.h>
#include <DataIO.h>
#include <Locker.h>
#include <MediaFormats.h>
#include <MediaRoster.h>
#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<uint8 *>(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;
}

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src/add-ons/media/plugins/vorbis/vorbisCodecPlugin.h Normal file
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#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();
};