Source code
Revision control
Copy as Markdown
Other Tools
/********************************************************************
* *
* 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 1994-2015 *
* *
********************************************************************
function: simple programmatic interface for encoder mode setup
********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "vorbis/codec.h"
#include "vorbis/vorbisenc.h"
#include "codec_internal.h"
#include "os.h"
#include "misc.h"
/* careful with this; it's using static array sizing to make managing
all the modes a little less annoying. If we use a residue backend
with > 12 partition types, or a different division of iteration,
this needs to be updated. */
typedef struct {
const static_codebook *books[12][4];
} static_bookblock;
typedef struct {
int res_type;
int limit_type; /* 0 lowpass limited, 1 point stereo limited */
int grouping;
const vorbis_info_residue0 *res;
const static_codebook *book_aux;
const static_codebook *book_aux_managed;
const static_bookblock *books_base;
const static_bookblock *books_base_managed;
} vorbis_residue_template;
typedef struct {
const vorbis_info_mapping0 *map;
const vorbis_residue_template *res;
} vorbis_mapping_template;
typedef struct vp_adjblock{
int block[P_BANDS];
} vp_adjblock;
typedef struct {
int data[NOISE_COMPAND_LEVELS];
} compandblock;
/* high level configuration information for setting things up
step-by-step with the detailed vorbis_encode_ctl interface.
There's a fair amount of redundancy such that interactive setup
does not directly deal with any vorbis_info or codec_setup_info
initialization; it's all stored (until full init) in this highlevel
setup, then flushed out to the real codec setup structs later. */
typedef struct {
int att[P_NOISECURVES];
float boost;
float decay;
} att3;
typedef struct { int data[P_NOISECURVES]; } adj3;
typedef struct {
int pre[PACKETBLOBS];
int post[PACKETBLOBS];
float kHz[PACKETBLOBS];
float lowpasskHz[PACKETBLOBS];
} adj_stereo;
typedef struct {
int lo;
int hi;
int fixed;
} noiseguard;
typedef struct {
int data[P_NOISECURVES][17];
} noise3;
typedef struct {
int mappings;
const double *rate_mapping;
const double *quality_mapping;
int coupling_restriction;
long samplerate_min_restriction;
long samplerate_max_restriction;
const int *blocksize_short;
const int *blocksize_long;
const att3 *psy_tone_masteratt;
const int *psy_tone_0dB;
const int *psy_tone_dBsuppress;
const vp_adjblock *psy_tone_adj_impulse;
const vp_adjblock *psy_tone_adj_long;
const vp_adjblock *psy_tone_adj_other;
const noiseguard *psy_noiseguards;
const noise3 *psy_noise_bias_impulse;
const noise3 *psy_noise_bias_padding;
const noise3 *psy_noise_bias_trans;
const noise3 *psy_noise_bias_long;
const int *psy_noise_dBsuppress;
const compandblock *psy_noise_compand;
const double *psy_noise_compand_short_mapping;
const double *psy_noise_compand_long_mapping;
const int *psy_noise_normal_start[2];
const int *psy_noise_normal_partition[2];
const double *psy_noise_normal_thresh;
const int *psy_ath_float;
const int *psy_ath_abs;
const double *psy_lowpass;
const vorbis_info_psy_global *global_params;
const double *global_mapping;
const adj_stereo *stereo_modes;
const static_codebook *const *const *const floor_books;
const vorbis_info_floor1 *floor_params;
const int floor_mappings;
const int **floor_mapping_list;
const vorbis_mapping_template *maps;
} ve_setup_data_template;
/* a few static coder conventions */
static const vorbis_info_mode _mode_template[2]={
{0,0,0,0},
{1,0,0,1}
};
static const vorbis_info_mapping0 _map_nominal[2]={
{1, {0,0}, {0}, {0}, 1,{0},{1}},
{1, {0,0}, {1}, {1}, 1,{0},{1}}
};
#include "modes/setup_44.h"
#include "modes/setup_44u.h"
#include "modes/setup_44p51.h"
#include "modes/setup_32.h"
#include "modes/setup_8.h"
#include "modes/setup_11.h"
#include "modes/setup_16.h"
#include "modes/setup_22.h"
#include "modes/setup_X.h"
static const ve_setup_data_template *const setup_list[]={
&ve_setup_44_stereo,
&ve_setup_44_51,
&ve_setup_44_uncoupled,
&ve_setup_32_stereo,
&ve_setup_32_uncoupled,
&ve_setup_22_stereo,
&ve_setup_22_uncoupled,
&ve_setup_16_stereo,
&ve_setup_16_uncoupled,
&ve_setup_11_stereo,
&ve_setup_11_uncoupled,
&ve_setup_8_stereo,
&ve_setup_8_uncoupled,
&ve_setup_X_stereo,
&ve_setup_X_uncoupled,
&ve_setup_XX_stereo,
&ve_setup_XX_uncoupled,
0
};
static void vorbis_encode_floor_setup(vorbis_info *vi,int s,
const static_codebook *const *const *const books,
const vorbis_info_floor1 *in,
const int *x){
int i,k,is=s;
vorbis_info_floor1 *f=_ogg_calloc(1,sizeof(*f));
codec_setup_info *ci=vi->codec_setup;
memcpy(f,in+x[is],sizeof(*f));
/* books */
{
int partitions=f->partitions;
int maxclass=-1;
int maxbook=-1;
for(i=0;i<partitions;i++)
if(f->partitionclass[i]>maxclass)maxclass=f->partitionclass[i];
for(i=0;i<=maxclass;i++){
if(f->class_book[i]>maxbook)maxbook=f->class_book[i];
f->class_book[i]+=ci->books;
for(k=0;k<(1<<f->class_subs[i]);k++){
if(f->class_subbook[i][k]>maxbook)maxbook=f->class_subbook[i][k];
if(f->class_subbook[i][k]>=0)f->class_subbook[i][k]+=ci->books;
}
}
for(i=0;i<=maxbook;i++)
ci->book_param[ci->books++]=(static_codebook *)books[x[is]][i];
}
/* for now, we're only using floor 1 */
ci->floor_type[ci->floors]=1;
ci->floor_param[ci->floors]=f;
ci->floors++;
return;
}
static void vorbis_encode_global_psych_setup(vorbis_info *vi,double s,
const vorbis_info_psy_global *in,
const double *x){
int i,is=s;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy_global *g=&ci->psy_g_param;
memcpy(g,in+(int)x[is],sizeof(*g));
ds=x[is]*(1.-ds)+x[is+1]*ds;
is=(int)ds;
ds-=is;
if(ds==0 && is>0){
is--;
ds=1.;
}
/* interpolate the trigger threshholds */
for(i=0;i<4;i++){
g->preecho_thresh[i]=in[is].preecho_thresh[i]*(1.-ds)+in[is+1].preecho_thresh[i]*ds;
g->postecho_thresh[i]=in[is].postecho_thresh[i]*(1.-ds)+in[is+1].postecho_thresh[i]*ds;
}
g->ampmax_att_per_sec=ci->hi.amplitude_track_dBpersec;
return;
}
static void vorbis_encode_global_stereo(vorbis_info *vi,
const highlevel_encode_setup *const hi,
const adj_stereo *p){
float s=hi->stereo_point_setting;
int i,is=s;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy_global *g=&ci->psy_g_param;
if(p){
memcpy(g->coupling_prepointamp,p[is].pre,sizeof(*p[is].pre)*PACKETBLOBS);
memcpy(g->coupling_postpointamp,p[is].post,sizeof(*p[is].post)*PACKETBLOBS);
if(hi->managed){
/* interpolate the kHz threshholds */
for(i=0;i<PACKETBLOBS;i++){
float kHz=p[is].kHz[i]*(1.-ds)+p[is+1].kHz[i]*ds;
g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
g->coupling_pkHz[i]=kHz;
kHz=p[is].lowpasskHz[i]*(1.-ds)+p[is+1].lowpasskHz[i]*ds;
g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
}
}else{
float kHz=p[is].kHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].kHz[PACKETBLOBS/2]*ds;
for(i=0;i<PACKETBLOBS;i++){
g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
g->coupling_pkHz[i]=kHz;
}
kHz=p[is].lowpasskHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].lowpasskHz[PACKETBLOBS/2]*ds;
for(i=0;i<PACKETBLOBS;i++){
g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
}
}
}else{
for(i=0;i<PACKETBLOBS;i++){
g->sliding_lowpass[0][i]=ci->blocksizes[0];
g->sliding_lowpass[1][i]=ci->blocksizes[1];
}
}
return;
}
static void vorbis_encode_psyset_setup(vorbis_info *vi,double s,
const int *nn_start,
const int *nn_partition,
const double *nn_thresh,
int block){
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
highlevel_encode_setup *hi=&ci->hi;
int is=s;
if(block>=ci->psys)
ci->psys=block+1;
if(!p){
p=_ogg_calloc(1,sizeof(*p));
ci->psy_param[block]=p;
}
memcpy(p,&_psy_info_template,sizeof(*p));
p->blockflag=block>>1;
if(hi->noise_normalize_p){
p->normal_p=1;
p->normal_start=nn_start[is];
p->normal_partition=nn_partition[is];
p->normal_thresh=nn_thresh[is];
}
return;
}
static void vorbis_encode_tonemask_setup(vorbis_info *vi,double s,int block,
const att3 *att,
const int *max,
const vp_adjblock *in){
int i,is=s;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
/* 0 and 2 are only used by bitmanagement, but there's no harm to always
filling the values in here */
p->tone_masteratt[0]=att[is].att[0]*(1.-ds)+att[is+1].att[0]*ds;
p->tone_masteratt[1]=att[is].att[1]*(1.-ds)+att[is+1].att[1]*ds;
p->tone_masteratt[2]=att[is].att[2]*(1.-ds)+att[is+1].att[2]*ds;
p->tone_centerboost=att[is].boost*(1.-ds)+att[is+1].boost*ds;
p->tone_decay=att[is].decay*(1.-ds)+att[is+1].decay*ds;
p->max_curve_dB=max[is]*(1.-ds)+max[is+1]*ds;
for(i=0;i<P_BANDS;i++)
p->toneatt[i]=in[is].block[i]*(1.-ds)+in[is+1].block[i]*ds;
return;
}
static void vorbis_encode_compand_setup(vorbis_info *vi,double s,int block,
const compandblock *in,
const double *x){
int i,is=s;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
ds=x[is]*(1.-ds)+x[is+1]*ds;
is=(int)ds;
ds-=is;
if(ds==0 && is>0){
is--;
ds=1.;
}
/* interpolate the compander settings */
for(i=0;i<NOISE_COMPAND_LEVELS;i++)
p->noisecompand[i]=in[is].data[i]*(1.-ds)+in[is+1].data[i]*ds;
return;
}
static void vorbis_encode_peak_setup(vorbis_info *vi,double s,int block,
const int *suppress){
int is=s;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
p->tone_abs_limit=suppress[is]*(1.-ds)+suppress[is+1]*ds;
return;
}
static void vorbis_encode_noisebias_setup(vorbis_info *vi,double s,int block,
const int *suppress,
const noise3 *in,
const noiseguard *guard,
double userbias){
int i,is=s,j;
double ds=s-is;
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
p->noisemaxsupp=suppress[is]*(1.-ds)+suppress[is+1]*ds;
p->noisewindowlomin=guard[block].lo;
p->noisewindowhimin=guard[block].hi;
p->noisewindowfixed=guard[block].fixed;
for(j=0;j<P_NOISECURVES;j++)
for(i=0;i<P_BANDS;i++)
p->noiseoff[j][i]=in[is].data[j][i]*(1.-ds)+in[is+1].data[j][i]*ds;
/* impulse blocks may take a user specified bias to boost the
nominal/high noise encoding depth */
for(j=0;j<P_NOISECURVES;j++){
float min=p->noiseoff[j][0]+6; /* the lowest it can go */
for(i=0;i<P_BANDS;i++){
p->noiseoff[j][i]+=userbias;
if(p->noiseoff[j][i]<min)p->noiseoff[j][i]=min;
}
}
return;
}
static void vorbis_encode_ath_setup(vorbis_info *vi,int block){
codec_setup_info *ci=vi->codec_setup;
vorbis_info_psy *p=ci->psy_param[block];
p->ath_adjatt=ci->hi.ath_floating_dB;
p->ath_maxatt=ci->hi.ath_absolute_dB;
return;
}
static int book_dup_or_new(codec_setup_info *ci,const static_codebook *book){
int i;
for(i=0;i<ci->books;i++)
if(ci->book_param[i]==book)return(i);
return(ci->books++);
}
static void vorbis_encode_blocksize_setup(vorbis_info *vi,double s,
const int *shortb,const int *longb){
codec_setup_info *ci=vi->codec_setup;
int is=s;
int blockshort=shortb[is];
int blocklong=longb[is];
ci->blocksizes[0]=blockshort;
ci->blocksizes[1]=blocklong;
}
static void vorbis_encode_residue_setup(vorbis_info *vi,
int number, int block,
const vorbis_residue_template *res){
codec_setup_info *ci=vi->codec_setup;
int i;
vorbis_info_residue0 *r=ci->residue_param[number]=
_ogg_malloc(sizeof(*r));
memcpy(r,res->res,sizeof(*r));
if(ci->residues<=number)ci->residues=number+1;
r->grouping=res->grouping;
ci->residue_type[number]=res->res_type;
/* fill in all the books */
{
int booklist=0,k;
if(ci->hi.managed){
for(i=0;i<r->partitions;i++)
for(k=0;k<4;k++)
if(res->books_base_managed->books[i][k])
r->secondstages[i]|=(1<<k);
r->groupbook=book_dup_or_new(ci,res->book_aux_managed);
ci->book_param[r->groupbook]=(static_codebook *)res->book_aux_managed;
for(i=0;i<r->partitions;i++){
for(k=0;k<4;k++){
if(res->books_base_managed->books[i][k]){
int bookid=book_dup_or_new(ci,res->books_base_managed->books[i][k]);
r->booklist[booklist++]=bookid;
ci->book_param[bookid]=(static_codebook *)res->books_base_managed->books[i][k];
}
}
}
}else{
for(i=0;i<r->partitions;i++)
for(k=0;k<4;k++)
if(res->books_base->books[i][k])
r->secondstages[i]|=(1<<k);
r->groupbook=book_dup_or_new(ci,res->book_aux);
ci->book_param[r->groupbook]=(static_codebook *)res->book_aux;
for(i=0;i<r->partitions;i++){
for(k=0;k<4;k++){
if(res->books_base->books[i][k]){
int bookid=book_dup_or_new(ci,res->books_base->books[i][k]);
r->booklist[booklist++]=bookid;
ci->book_param[bookid]=(static_codebook *)res->books_base->books[i][k];
}
}
}
}
}
/* lowpass setup/pointlimit */
{
double freq=ci->hi.lowpass_kHz*1000.;
vorbis_info_floor1 *f=ci->floor_param[block]; /* by convention */
double nyq=vi->rate/2.;
long blocksize=ci->blocksizes[block]>>1;
/* lowpass needs to be set in the floor and the residue. */
if(freq>nyq)freq=nyq;
/* in the floor, the granularity can be very fine; it doesn't alter
the encoding structure, only the samples used to fit the floor
approximation */
f->n=freq/nyq*blocksize;
/* this res may by limited by the maximum pointlimit of the mode,
not the lowpass. the floor is always lowpass limited. */
switch(res->limit_type){
case 1: /* point stereo limited */
if(ci->hi.managed)
freq=ci->psy_g_param.coupling_pkHz[PACKETBLOBS-1]*1000.;
else
freq=ci->psy_g_param.coupling_pkHz[PACKETBLOBS/2]*1000.;
if(freq>nyq)freq=nyq;
break;
case 2: /* LFE channel; lowpass at ~ 250Hz */
freq=250;
break;
default:
/* already set */
break;
}
/* in the residue, we're constrained, physically, by partition
boundaries. We still lowpass 'wherever', but we have to round up
here to next boundary, or the vorbis spec will round it *down* to
previous boundary in encode/decode */
if(ci->residue_type[number]==2){
/* residue 2 bundles together multiple channels; used by stereo
and surround. Count the channels in use */
/* Multiple maps/submaps can point to the same residue. In the case
of residue 2, they all better have the same number of
channels/samples. */
int j,k,ch=0;
for(i=0;i<ci->maps&&ch==0;i++){
vorbis_info_mapping0 *mi=(vorbis_info_mapping0 *)ci->map_param[i];
for(j=0;j<mi->submaps && ch==0;j++)
if(mi->residuesubmap[j]==number) /* we found a submap referencing theis residue backend */
for(k=0;k<vi->channels;k++)
if(mi->chmuxlist[k]==j) /* this channel belongs to the submap */
ch++;
}
r->end=(int)((freq/nyq*blocksize*ch)/r->grouping+.9)* /* round up only if we're well past */
r->grouping;
/* the blocksize and grouping may disagree at the end */
if(r->end>blocksize*ch)r->end=blocksize*ch/r->grouping*r->grouping;
}else{
r->end=(int)((freq/nyq*blocksize)/r->grouping+.9)* /* round up only if we're well past */
r->grouping;
/* the blocksize and grouping may disagree at the end */
if(r->end>blocksize)r->end=blocksize/r->grouping*r->grouping;
}
if(r->end==0)r->end=r->grouping; /* LFE channel */
}
}
/* we assume two maps in this encoder */
static void vorbis_encode_map_n_res_setup(vorbis_info *vi,double s,
const vorbis_mapping_template *maps){
codec_setup_info *ci=vi->codec_setup;
int i,j,is=s,modes=2;
const vorbis_info_mapping0 *map=maps[is].map;
const vorbis_info_mode *mode=_mode_template;
const vorbis_residue_template *res=maps[is].res;
if(ci->blocksizes[0]==ci->blocksizes[1])modes=1;
for(i=0;i<modes;i++){
ci->map_param[i]=_ogg_calloc(1,sizeof(*map));
ci->mode_param[i]=_ogg_calloc(1,sizeof(*mode));
memcpy(ci->mode_param[i],mode+i,sizeof(*_mode_template));
if(i>=ci->modes)ci->modes=i+1;
ci->map_type[i]=0;
memcpy(ci->map_param[i],map+i,sizeof(*map));
if(i>=ci->maps)ci->maps=i+1;
for(j=0;j<map[i].submaps;j++)
vorbis_encode_residue_setup(vi,map[i].residuesubmap[j],i
,res+map[i].residuesubmap[j]);
}
}
static double setting_to_approx_bitrate(vorbis_info *vi){
codec_setup_info *ci=vi->codec_setup;
highlevel_encode_setup *hi=&ci->hi;
ve_setup_data_template *setup=(ve_setup_data_template *)hi->setup;
int is=hi->base_setting;
double ds=hi->base_setting-is;
int ch=vi->channels;
const double *r=setup->rate_mapping;
if(r==NULL)
return(-1);
return((r[is]*(1.-ds)+r[is+1]*ds)*ch);
}
static const void *get_setup_template(long ch,long srate,
double req,int q_or_bitrate,
double *base_setting){
int i=0,j;
if(q_or_bitrate)req/=ch;
while(setup_list[i]){
if(setup_list[i]->coupling_restriction==-1 ||
setup_list[i]->coupling_restriction==ch){
if(srate>=setup_list[i]->samplerate_min_restriction &&
srate<=setup_list[i]->samplerate_max_restriction){
int mappings=setup_list[i]->mappings;
const double *map=(q_or_bitrate?
setup_list[i]->rate_mapping:
setup_list[i]->quality_mapping);
/* the template matches. Does the requested quality mode
fall within this template's modes? */
if(req<map[0]){++i;continue;}
if(req>map[setup_list[i]->mappings]){++i;continue;}
for(j=0;j<mappings;j++)
if(req>=map[j] && req<map[j+1])break;
/* an all-points match */
if(j==mappings)
*base_setting=j-.001;
else{
float low=map[j];
float high=map[j+1];
float del=(req-low)/(high-low);
*base_setting=j+del;
}
return(setup_list[i]);
}
}
i++;
}
return NULL;
}
/* encoders will need to use vorbis_info_init beforehand and call
vorbis_info clear when all done */
/* two interfaces; this, more detailed one, and later a convenience
layer on top */
/* the final setup call */
int vorbis_encode_setup_init(vorbis_info *vi){
int i,i0=0,singleblock=0;
codec_setup_info *ci=vi->codec_setup;
ve_setup_data_template *setup=NULL;
highlevel_encode_setup *hi=&ci->hi;
if(ci==NULL)return(OV_EINVAL);
if(vi->channels<1||vi->channels>255)return(OV_EINVAL);
if(!hi->impulse_block_p)i0=1;
/* too low/high an ATH floater is nonsensical, but doesn't break anything */
if(hi->ath_floating_dB>-80)hi->ath_floating_dB=-80;
if(hi->ath_floating_dB<-200)hi->ath_floating_dB=-200;
/* again, bound this to avoid the app shooting itself int he foot
too badly */
if(hi->amplitude_track_dBpersec>0.)hi->amplitude_track_dBpersec=0.;
if(hi->amplitude_track_dBpersec<-99999.)hi->amplitude_track_dBpersec=-99999.;
/* get the appropriate setup template; matches the fetch in previous
stages */
setup=(ve_setup_data_template *)hi->setup;
if(setup==NULL)return(OV_EINVAL);
hi->set_in_stone=1;
/* choose block sizes from configured sizes as well as paying
attention to long_block_p and short_block_p. If the configured
short and long blocks are the same length, we set long_block_p
and unset short_block_p */
vorbis_encode_blocksize_setup(vi,hi->base_setting,
setup->blocksize_short,
setup->blocksize_long);
if(ci->blocksizes[0]==ci->blocksizes[1])singleblock=1;
/* floor setup; choose proper floor params. Allocated on the floor
stack in order; if we alloc only a single long floor, it's 0 */
for(i=0;i<setup->floor_mappings;i++)
vorbis_encode_floor_setup(vi,hi->base_setting,
setup->floor_books,
setup->floor_params,
setup->floor_mapping_list[i]);
/* setup of [mostly] short block detection and stereo*/
vorbis_encode_global_psych_setup(vi,hi->trigger_setting,
setup->global_params,
setup->global_mapping);
vorbis_encode_global_stereo(vi,hi,setup->stereo_modes);
/* basic psych setup and noise normalization */
vorbis_encode_psyset_setup(vi,hi->base_setting,
setup->psy_noise_normal_start[0],
setup->psy_noise_normal_partition[0],
setup->psy_noise_normal_thresh,
0);
vorbis_encode_psyset_setup(vi,hi->base_setting,
setup->psy_noise_normal_start[0],
setup->psy_noise_normal_partition[0],
setup->psy_noise_normal_thresh,
1);
if(!singleblock){
vorbis_encode_psyset_setup(vi,hi->base_setting,
setup->psy_noise_normal_start[1],
setup->psy_noise_normal_partition[1],
setup->psy_noise_normal_thresh,
2);
vorbis_encode_psyset_setup(vi,hi->base_setting,
setup->psy_noise_normal_start[1],
setup->psy_noise_normal_partition[1],
setup->psy_noise_normal_thresh,
3);
}
/* tone masking setup */
vorbis_encode_tonemask_setup(vi,hi->block[i0].tone_mask_setting,0,
setup->psy_tone_masteratt,
setup->psy_tone_0dB,
setup->psy_tone_adj_impulse);
vorbis_encode_tonemask_setup(vi,hi->block[1].tone_mask_setting,1,
setup->psy_tone_masteratt,
setup->psy_tone_0dB,
setup->psy_tone_adj_other);
if(!singleblock){
vorbis_encode_tonemask_setup(vi,hi->block[2].tone_mask_setting,2,
setup->psy_tone_masteratt,
setup->psy_tone_0dB,
setup->psy_tone_adj_other);
vorbis_encode_tonemask_setup(vi,hi->block[3].tone_mask_setting,3,
setup->psy_tone_masteratt,
setup->psy_tone_0dB,
setup->psy_tone_adj_long);
}
/* noise companding setup */
vorbis_encode_compand_setup(vi,hi->block[i0].noise_compand_setting,0,
setup->psy_noise_compand,
setup->psy_noise_compand_short_mapping);
vorbis_encode_compand_setup(vi,hi->block[1].noise_compand_setting,1,
setup->psy_noise_compand,
setup->psy_noise_compand_short_mapping);
if(!singleblock){
vorbis_encode_compand_setup(vi,hi->block[2].noise_compand_setting,2,
setup->psy_noise_compand,
setup->psy_noise_compand_long_mapping);
vorbis_encode_compand_setup(vi,hi->block[3].noise_compand_setting,3,
setup->psy_noise_compand,
setup->psy_noise_compand_long_mapping);
}
/* peak guarding setup */
vorbis_encode_peak_setup(vi,hi->block[i0].tone_peaklimit_setting,0,
setup->psy_tone_dBsuppress);
vorbis_encode_peak_setup(vi,hi->block[1].tone_peaklimit_setting,1,
setup->psy_tone_dBsuppress);
if(!singleblock){
vorbis_encode_peak_setup(vi,hi->block[2].tone_peaklimit_setting,2,
setup->psy_tone_dBsuppress);
vorbis_encode_peak_setup(vi,hi->block[3].tone_peaklimit_setting,3,
setup->psy_tone_dBsuppress);
}
/* noise bias setup */
vorbis_encode_noisebias_setup(vi,hi->block[i0].noise_bias_setting,0,
setup->psy_noise_dBsuppress,
setup->psy_noise_bias_impulse,
setup->psy_noiseguards,
(i0==0?hi->impulse_noisetune:0.));
vorbis_encode_noisebias_setup(vi,hi->block[1].noise_bias_setting,1,
setup->psy_noise_dBsuppress,
setup->psy_noise_bias_padding,
setup->psy_noiseguards,0.);
if(!singleblock){
vorbis_encode_noisebias_setup(vi,hi->block[2].noise_bias_setting,2,
setup->psy_noise_dBsuppress,
setup->psy_noise_bias_trans,
setup->psy_noiseguards,0.);
vorbis_encode_noisebias_setup(vi,hi->block[3].noise_bias_setting,3,
setup->psy_noise_dBsuppress,
setup->psy_noise_bias_long,
setup->psy_noiseguards,0.);
}
vorbis_encode_ath_setup(vi,0);
vorbis_encode_ath_setup(vi,1);
if(!singleblock){
vorbis_encode_ath_setup(vi,2);
vorbis_encode_ath_setup(vi,3);
}
vorbis_encode_map_n_res_setup(vi,hi->base_setting,setup->maps);
/* set bitrate readonlies and management */
if(hi->bitrate_av>0)
vi->bitrate_nominal=hi->bitrate_av;
else{
vi->bitrate_nominal=setting_to_approx_bitrate(vi);
}
vi->bitrate_lower=hi->bitrate_min;
vi->bitrate_upper=hi->bitrate_max;
if(hi->bitrate_av)
vi->bitrate_window=(double)hi->bitrate_reservoir/hi->bitrate_av;
else
vi->bitrate_window=0.;
if(hi->managed){
ci->bi.avg_rate=hi->bitrate_av;
ci->bi.min_rate=hi->bitrate_min;
ci->bi.max_rate=hi->bitrate_max;
ci->bi.reservoir_bits=hi->bitrate_reservoir;
ci->bi.reservoir_bias=
hi->bitrate_reservoir_bias;
ci->bi.slew_damp=hi->bitrate_av_damp;
}
return(0);
}
static void vorbis_encode_setup_setting(vorbis_info *vi,
long channels,
long rate){
int i,is;
codec_setup_info *ci=vi->codec_setup;
highlevel_encode_setup *hi=&ci->hi;
const ve_setup_data_template *setup=hi->setup;
double ds;
vi->version=0;
vi->channels=channels;
vi->rate=rate;
hi->impulse_block_p=1;
hi->noise_normalize_p=1;
is=hi->base_setting;
ds=hi->base_setting-is;
hi->stereo_point_setting=hi->base_setting;
if(!hi->lowpass_altered)
hi->lowpass_kHz=
setup->psy_lowpass[is]*(1.-ds)+setup->psy_lowpass[is+1]*ds;
hi->ath_floating_dB=setup->psy_ath_float[is]*(1.-ds)+
setup->psy_ath_float[is+1]*ds;
hi->ath_absolute_dB=setup->psy_ath_abs[is]*(1.-ds)+
setup->psy_ath_abs[is+1]*ds;
hi->amplitude_track_dBpersec=-6.;
hi->trigger_setting=hi->base_setting;
for(i=0;i<4;i++){
hi->block[i].tone_mask_setting=hi->base_setting;
hi->block[i].tone_peaklimit_setting=hi->base_setting;
hi->block[i].noise_bias_setting=hi->base_setting;
hi->block[i].noise_compand_setting=hi->base_setting;
}
}
int vorbis_encode_setup_vbr(vorbis_info *vi,
long channels,
long rate,
float quality){
codec_setup_info *ci;
highlevel_encode_setup *hi;
if(rate<=0) return OV_EINVAL;
ci=vi->codec_setup;
hi=&ci->hi;
quality+=.0000001;
if(quality>=1.)quality=.9999;
hi->req=quality;
hi->setup=get_setup_template(channels,rate,quality,0,&hi->base_setting);
if(!hi->setup)return OV_EIMPL;
vorbis_encode_setup_setting(vi,channels,rate);
hi->managed=0;
hi->coupling_p=1;
return 0;
}
int vorbis_encode_init_vbr(vorbis_info *vi,
long channels,
long rate,
float base_quality /* 0. to 1. */
){
int ret=0;
ret=vorbis_encode_setup_vbr(vi,channels,rate,base_quality);
if(ret){
vorbis_info_clear(vi);
return ret;
}
ret=vorbis_encode_setup_init(vi);
if(ret)
vorbis_info_clear(vi);
return(ret);
}
int vorbis_encode_setup_managed(vorbis_info *vi,
long channels,
long rate,
long max_bitrate,
long nominal_bitrate,
long min_bitrate){
codec_setup_info *ci;
highlevel_encode_setup *hi;
double tnominal;
if(rate<=0) return OV_EINVAL;
ci=vi->codec_setup;
hi=&ci->hi;
tnominal=nominal_bitrate;
if(nominal_bitrate<=0.){
if(max_bitrate>0.){
if(min_bitrate>0.)
nominal_bitrate=(max_bitrate+min_bitrate)*.5;
else
nominal_bitrate=max_bitrate*.875;
}else{
if(min_bitrate>0.){
nominal_bitrate=min_bitrate;
}else{
return(OV_EINVAL);
}
}
}
hi->req=nominal_bitrate;
hi->setup=get_setup_template(channels,rate,nominal_bitrate,1,&hi->base_setting);
if(!hi->setup)return OV_EIMPL;
vorbis_encode_setup_setting(vi,channels,rate);
/* initialize management with sane defaults */
hi->coupling_p=1;
hi->managed=1;
hi->bitrate_min=min_bitrate;
hi->bitrate_max=max_bitrate;
hi->bitrate_av=tnominal;
hi->bitrate_av_damp=1.5f; /* full range in no less than 1.5 second */
hi->bitrate_reservoir=nominal_bitrate*2;
hi->bitrate_reservoir_bias=.1; /* bias toward hoarding bits */
return(0);
}
int vorbis_encode_init(vorbis_info *vi,
long channels,
long rate,
long max_bitrate,
long nominal_bitrate,
long min_bitrate){
int ret=vorbis_encode_setup_managed(vi,channels,rate,
max_bitrate,
nominal_bitrate,
min_bitrate);
if(ret){
vorbis_info_clear(vi);
return(ret);
}
ret=vorbis_encode_setup_init(vi);
if(ret)
vorbis_info_clear(vi);
return(ret);
}
int vorbis_encode_ctl(vorbis_info *vi,int number,void *arg){
if(vi){
codec_setup_info *ci=vi->codec_setup;
highlevel_encode_setup *hi=&ci->hi;
int setp=(number&0xf); /* a read request has a low nibble of 0 */
if(setp && hi->set_in_stone)return(OV_EINVAL);
switch(number){
/* now deprecated *****************/
case OV_ECTL_RATEMANAGE_GET:
{
struct ovectl_ratemanage_arg *ai=
(struct ovectl_ratemanage_arg *)arg;
ai->management_active=hi->managed;
ai->bitrate_hard_window=ai->bitrate_av_window=
(double)hi->bitrate_reservoir/vi->rate;
ai->bitrate_av_window_center=1.;
ai->bitrate_hard_min=hi->bitrate_min;
ai->bitrate_hard_max=hi->bitrate_max;
ai->bitrate_av_lo=hi->bitrate_av;
ai->bitrate_av_hi=hi->bitrate_av;
}
return(0);
/* now deprecated *****************/
case OV_ECTL_RATEMANAGE_SET:
{
struct ovectl_ratemanage_arg *ai=
(struct ovectl_ratemanage_arg *)arg;
if(ai==NULL){
hi->managed=0;
}else{
hi->managed=ai->management_active;
vorbis_encode_ctl(vi,OV_ECTL_RATEMANAGE_AVG,arg);
vorbis_encode_ctl(vi,OV_ECTL_RATEMANAGE_HARD,arg);
}
}
return 0;
/* now deprecated *****************/
case OV_ECTL_RATEMANAGE_AVG:
{
struct ovectl_ratemanage_arg *ai=
(struct ovectl_ratemanage_arg *)arg;
if(ai==NULL){
hi->bitrate_av=0;
}else{
hi->bitrate_av=(ai->bitrate_av_lo+ai->bitrate_av_hi)*.5;
}
}
return(0);
/* now deprecated *****************/
case OV_ECTL_RATEMANAGE_HARD:
{
struct ovectl_ratemanage_arg *ai=
(struct ovectl_ratemanage_arg *)arg;
if(ai==NULL){
hi->bitrate_min=0;
hi->bitrate_max=0;
}else{
hi->bitrate_min=ai->bitrate_hard_min;
hi->bitrate_max=ai->bitrate_hard_max;
hi->bitrate_reservoir=ai->bitrate_hard_window*
(hi->bitrate_max+hi->bitrate_min)*.5;
}
if(hi->bitrate_reservoir<128.)
hi->bitrate_reservoir=128.;
}
return(0);
/* replacement ratemanage interface */
case OV_ECTL_RATEMANAGE2_GET:
{
struct ovectl_ratemanage2_arg *ai=
(struct ovectl_ratemanage2_arg *)arg;
if(ai==NULL)return OV_EINVAL;
ai->management_active=hi->managed;
ai->bitrate_limit_min_kbps=hi->bitrate_min/1000;
ai->bitrate_limit_max_kbps=hi->bitrate_max/1000;
ai->bitrate_average_kbps=hi->bitrate_av/1000;
ai->bitrate_average_damping=hi->bitrate_av_damp;
ai->bitrate_limit_reservoir_bits=hi->bitrate_reservoir;
ai->bitrate_limit_reservoir_bias=hi->bitrate_reservoir_bias;
}
return (0);
case OV_ECTL_RATEMANAGE2_SET:
{
struct ovectl_ratemanage2_arg *ai=
(struct ovectl_ratemanage2_arg *)arg;
if(ai==NULL){
hi->managed=0;
}else{
/* sanity check; only catch invariant violations */
if(ai->bitrate_limit_min_kbps>0 &&
ai->bitrate_average_kbps>0 &&
ai->bitrate_limit_min_kbps>ai->bitrate_average_kbps)
return OV_EINVAL;
if(ai->bitrate_limit_max_kbps>0 &&
ai->bitrate_average_kbps>0 &&
ai->bitrate_limit_max_kbps<ai->bitrate_average_kbps)
return OV_EINVAL;
if(ai->bitrate_limit_min_kbps>0 &&
ai->bitrate_limit_max_kbps>0 &&
ai->bitrate_limit_min_kbps>ai->bitrate_limit_max_kbps)
return OV_EINVAL;
if(ai->bitrate_average_damping <= 0.)
return OV_EINVAL;
if(ai->bitrate_limit_reservoir_bits < 0)
return OV_EINVAL;
if(ai->bitrate_limit_reservoir_bias < 0.)
return OV_EINVAL;
if(ai->bitrate_limit_reservoir_bias > 1.)
return OV_EINVAL;
hi->managed=ai->management_active;
hi->bitrate_min=ai->bitrate_limit_min_kbps * 1000;
hi->bitrate_max=ai->bitrate_limit_max_kbps * 1000;
hi->bitrate_av=ai->bitrate_average_kbps * 1000;
hi->bitrate_av_damp=ai->bitrate_average_damping;
hi->bitrate_reservoir=ai->bitrate_limit_reservoir_bits;
hi->bitrate_reservoir_bias=ai->bitrate_limit_reservoir_bias;
}
}
return 0;
case OV_ECTL_LOWPASS_GET:
{
double *farg=(double *)arg;
*farg=hi->lowpass_kHz;
}
return(0);
case OV_ECTL_LOWPASS_SET:
{
double *farg=(double *)arg;
hi->lowpass_kHz=*farg;
if(hi->lowpass_kHz<2.)hi->lowpass_kHz=2.;
if(hi->lowpass_kHz>99.)hi->lowpass_kHz=99.;
hi->lowpass_altered=1;
}
return(0);
case OV_ECTL_IBLOCK_GET:
{
double *farg=(double *)arg;
*farg=hi->impulse_noisetune;
}
return(0);
case OV_ECTL_IBLOCK_SET:
{
double *farg=(double *)arg;
hi->impulse_noisetune=*farg;
if(hi->impulse_noisetune>0.)hi->impulse_noisetune=0.;
if(hi->impulse_noisetune<-15.)hi->impulse_noisetune=-15.;
}
return(0);
case OV_ECTL_COUPLING_GET:
{
int *iarg=(int *)arg;
*iarg=hi->coupling_p;
}
return(0);
case OV_ECTL_COUPLING_SET:
{
const void *new_template;
double new_base=0.;
int *iarg=(int *)arg;
hi->coupling_p=((*iarg)!=0);
/* Fetching a new template can alter the base_setting, which
many other parameters are based on. Right now, the only
parameter drawn from the base_setting that can be altered
by an encctl is the lowpass, so that is explictly flagged
to not be overwritten when we fetch a new template and
recompute the dependant settings */
new_template = get_setup_template(hi->coupling_p?vi->channels:-1,
vi->rate,
hi->req,
hi->managed,
&new_base);
if(!new_template)return OV_EIMPL;
hi->setup=new_template;
hi->base_setting=new_base;
vorbis_encode_setup_setting(vi,vi->channels,vi->rate);
}
return(0);
}
return(OV_EIMPL);
}
return(OV_EINVAL);
}