diff options
author | Aki <please@ignore.pl> | 2021-10-01 23:46:47 +0200 |
---|---|---|
committer | Aki <please@ignore.pl> | 2021-10-01 23:46:47 +0200 |
commit | ddd76aa4e1571c8d5965709db5c3cd47a89c9a38 (patch) | |
tree | 4e07bfaebede6531a70984fdebc855bca52f394a /vorbis/lib/vorbisenc.c | |
parent | 3a507e08b1d4e5970b27401a7e6517570d529400 (diff) | |
parent | c2d2c3551501110fddd78674d5435bfaa70382a3 (diff) | |
download | starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.zip starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.tar.gz starshatter-ddd76aa4e1571c8d5965709db5c3cd47a89c9a38.tar.bz2 |
Project is now built with CMake
Diffstat (limited to 'vorbis/lib/vorbisenc.c')
-rw-r--r-- | vorbis/lib/vorbisenc.c | 1223 |
1 files changed, 1223 insertions, 0 deletions
diff --git a/vorbis/lib/vorbisenc.c b/vorbis/lib/vorbisenc.c new file mode 100644 index 0000000..4a4607c --- /dev/null +++ b/vorbis/lib/vorbisenc.c @@ -0,0 +1,1223 @@ +/******************************************************************** + * * + * 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 * + * by the Xiph.Org Foundation http://www.xiph.org/ * + * * + ******************************************************************** + + 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(!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(!hi->setup)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); +} |