mirror of
https://github.com/Relintai/pandemonium_engine.git
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1225 lines
38 KiB
C
1225 lines
38 KiB
C
/********************************************************************
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* *
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* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
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* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
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* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
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* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
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* *
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* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2015 *
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* by the Xiph.Org Foundation https://xiph.org/ *
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* *
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********************************************************************
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function: simple programmatic interface for encoder mode setup
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********************************************************************/
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include "vorbis/codec.h"
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#include "vorbis/vorbisenc.h"
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#include "codec_internal.h"
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#include "os.h"
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#include "misc.h"
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/* careful with this; it's using static array sizing to make managing
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all the modes a little less annoying. If we use a residue backend
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with > 12 partition types, or a different division of iteration,
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this needs to be updated. */
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typedef struct {
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const static_codebook *books[12][4];
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} static_bookblock;
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typedef struct {
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int res_type;
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int limit_type; /* 0 lowpass limited, 1 point stereo limited */
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int grouping;
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const vorbis_info_residue0 *res;
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const static_codebook *book_aux;
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const static_codebook *book_aux_managed;
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const static_bookblock *books_base;
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const static_bookblock *books_base_managed;
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} vorbis_residue_template;
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typedef struct {
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const vorbis_info_mapping0 *map;
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const vorbis_residue_template *res;
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} vorbis_mapping_template;
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typedef struct vp_adjblock{
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int block[P_BANDS];
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} vp_adjblock;
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typedef struct {
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int data[NOISE_COMPAND_LEVELS];
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} compandblock;
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/* high level configuration information for setting things up
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step-by-step with the detailed vorbis_encode_ctl interface.
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There's a fair amount of redundancy such that interactive setup
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does not directly deal with any vorbis_info or codec_setup_info
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initialization; it's all stored (until full init) in this highlevel
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setup, then flushed out to the real codec setup structs later. */
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typedef struct {
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int att[P_NOISECURVES];
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float boost;
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float decay;
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} att3;
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typedef struct { int data[P_NOISECURVES]; } adj3;
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typedef struct {
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int pre[PACKETBLOBS];
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int post[PACKETBLOBS];
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float kHz[PACKETBLOBS];
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float lowpasskHz[PACKETBLOBS];
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} adj_stereo;
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typedef struct {
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int lo;
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int hi;
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int fixed;
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} noiseguard;
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typedef struct {
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int data[P_NOISECURVES][17];
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} noise3;
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typedef struct {
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int mappings;
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const double *rate_mapping;
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const double *quality_mapping;
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int coupling_restriction;
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long samplerate_min_restriction;
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long samplerate_max_restriction;
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const int *blocksize_short;
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const int *blocksize_long;
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const att3 *psy_tone_masteratt;
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const int *psy_tone_0dB;
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const int *psy_tone_dBsuppress;
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const vp_adjblock *psy_tone_adj_impulse;
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const vp_adjblock *psy_tone_adj_long;
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const vp_adjblock *psy_tone_adj_other;
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const noiseguard *psy_noiseguards;
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const noise3 *psy_noise_bias_impulse;
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const noise3 *psy_noise_bias_padding;
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const noise3 *psy_noise_bias_trans;
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const noise3 *psy_noise_bias_long;
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const int *psy_noise_dBsuppress;
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const compandblock *psy_noise_compand;
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const double *psy_noise_compand_short_mapping;
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const double *psy_noise_compand_long_mapping;
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const int *psy_noise_normal_start[2];
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const int *psy_noise_normal_partition[2];
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const double *psy_noise_normal_thresh;
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const int *psy_ath_float;
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const int *psy_ath_abs;
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const double *psy_lowpass;
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const vorbis_info_psy_global *global_params;
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const double *global_mapping;
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const adj_stereo *stereo_modes;
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const static_codebook *const *const *const floor_books;
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const vorbis_info_floor1 *floor_params;
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const int floor_mappings;
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const int **floor_mapping_list;
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const vorbis_mapping_template *maps;
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} ve_setup_data_template;
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/* a few static coder conventions */
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static const vorbis_info_mode _mode_template[2]={
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{0,0,0,0},
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{1,0,0,1}
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};
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static const vorbis_info_mapping0 _map_nominal[2]={
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{1, {0,0}, {0}, {0}, 1,{0},{1}},
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{1, {0,0}, {1}, {1}, 1,{0},{1}}
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};
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#include "modes/setup_44.h"
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#include "modes/setup_44u.h"
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#include "modes/setup_44p51.h"
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#include "modes/setup_32.h"
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#include "modes/setup_8.h"
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#include "modes/setup_11.h"
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#include "modes/setup_16.h"
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#include "modes/setup_22.h"
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#include "modes/setup_X.h"
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static const ve_setup_data_template *const setup_list[]={
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&ve_setup_44_stereo,
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&ve_setup_44_51,
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&ve_setup_44_uncoupled,
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&ve_setup_32_stereo,
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&ve_setup_32_uncoupled,
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&ve_setup_22_stereo,
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&ve_setup_22_uncoupled,
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&ve_setup_16_stereo,
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&ve_setup_16_uncoupled,
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&ve_setup_11_stereo,
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&ve_setup_11_uncoupled,
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&ve_setup_8_stereo,
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&ve_setup_8_uncoupled,
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&ve_setup_X_stereo,
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&ve_setup_X_uncoupled,
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&ve_setup_XX_stereo,
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&ve_setup_XX_uncoupled,
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0
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};
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static void vorbis_encode_floor_setup(vorbis_info *vi,int s,
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const static_codebook *const *const *const books,
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const vorbis_info_floor1 *in,
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const int *x){
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int i,k,is=s;
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vorbis_info_floor1 *f=_ogg_calloc(1,sizeof(*f));
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codec_setup_info *ci=vi->codec_setup;
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memcpy(f,in+x[is],sizeof(*f));
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/* books */
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{
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int partitions=f->partitions;
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int maxclass=-1;
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int maxbook=-1;
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for(i=0;i<partitions;i++)
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if(f->partitionclass[i]>maxclass)maxclass=f->partitionclass[i];
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for(i=0;i<=maxclass;i++){
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if(f->class_book[i]>maxbook)maxbook=f->class_book[i];
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f->class_book[i]+=ci->books;
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for(k=0;k<(1<<f->class_subs[i]);k++){
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if(f->class_subbook[i][k]>maxbook)maxbook=f->class_subbook[i][k];
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if(f->class_subbook[i][k]>=0)f->class_subbook[i][k]+=ci->books;
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}
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}
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for(i=0;i<=maxbook;i++)
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ci->book_param[ci->books++]=(static_codebook *)books[x[is]][i];
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}
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/* for now, we're only using floor 1 */
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ci->floor_type[ci->floors]=1;
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ci->floor_param[ci->floors]=f;
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ci->floors++;
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return;
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}
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static void vorbis_encode_global_psych_setup(vorbis_info *vi,double s,
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const vorbis_info_psy_global *in,
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const double *x){
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int i,is=s;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy_global *g=&ci->psy_g_param;
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memcpy(g,in+(int)x[is],sizeof(*g));
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ds=x[is]*(1.-ds)+x[is+1]*ds;
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is=(int)ds;
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ds-=is;
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if(ds==0 && is>0){
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is--;
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ds=1.;
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}
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/* interpolate the trigger threshholds */
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for(i=0;i<4;i++){
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g->preecho_thresh[i]=in[is].preecho_thresh[i]*(1.-ds)+in[is+1].preecho_thresh[i]*ds;
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g->postecho_thresh[i]=in[is].postecho_thresh[i]*(1.-ds)+in[is+1].postecho_thresh[i]*ds;
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}
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g->ampmax_att_per_sec=ci->hi.amplitude_track_dBpersec;
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return;
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}
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static void vorbis_encode_global_stereo(vorbis_info *vi,
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const highlevel_encode_setup *const hi,
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const adj_stereo *p){
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float s=hi->stereo_point_setting;
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int i,is=s;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy_global *g=&ci->psy_g_param;
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if(p){
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memcpy(g->coupling_prepointamp,p[is].pre,sizeof(*p[is].pre)*PACKETBLOBS);
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memcpy(g->coupling_postpointamp,p[is].post,sizeof(*p[is].post)*PACKETBLOBS);
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if(hi->managed){
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/* interpolate the kHz threshholds */
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for(i=0;i<PACKETBLOBS;i++){
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float kHz=p[is].kHz[i]*(1.-ds)+p[is+1].kHz[i]*ds;
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g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
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g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
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g->coupling_pkHz[i]=kHz;
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kHz=p[is].lowpasskHz[i]*(1.-ds)+p[is+1].lowpasskHz[i]*ds;
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g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
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g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
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}
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}else{
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float kHz=p[is].kHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].kHz[PACKETBLOBS/2]*ds;
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for(i=0;i<PACKETBLOBS;i++){
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g->coupling_pointlimit[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
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g->coupling_pointlimit[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
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g->coupling_pkHz[i]=kHz;
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}
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kHz=p[is].lowpasskHz[PACKETBLOBS/2]*(1.-ds)+p[is+1].lowpasskHz[PACKETBLOBS/2]*ds;
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for(i=0;i<PACKETBLOBS;i++){
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g->sliding_lowpass[0][i]=kHz*1000./vi->rate*ci->blocksizes[0];
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g->sliding_lowpass[1][i]=kHz*1000./vi->rate*ci->blocksizes[1];
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}
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}
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}else{
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for(i=0;i<PACKETBLOBS;i++){
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g->sliding_lowpass[0][i]=ci->blocksizes[0];
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g->sliding_lowpass[1][i]=ci->blocksizes[1];
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}
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}
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return;
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}
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static void vorbis_encode_psyset_setup(vorbis_info *vi,double s,
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const int *nn_start,
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const int *nn_partition,
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const double *nn_thresh,
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int block){
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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highlevel_encode_setup *hi=&ci->hi;
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int is=s;
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if(block>=ci->psys)
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ci->psys=block+1;
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if(!p){
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p=_ogg_calloc(1,sizeof(*p));
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ci->psy_param[block]=p;
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}
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memcpy(p,&_psy_info_template,sizeof(*p));
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p->blockflag=block>>1;
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if(hi->noise_normalize_p){
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p->normal_p=1;
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p->normal_start=nn_start[is];
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p->normal_partition=nn_partition[is];
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p->normal_thresh=nn_thresh[is];
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}
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return;
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}
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static void vorbis_encode_tonemask_setup(vorbis_info *vi,double s,int block,
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const att3 *att,
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const int *max,
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const vp_adjblock *in){
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int i,is=s;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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/* 0 and 2 are only used by bitmanagement, but there's no harm to always
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filling the values in here */
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p->tone_masteratt[0]=att[is].att[0]*(1.-ds)+att[is+1].att[0]*ds;
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p->tone_masteratt[1]=att[is].att[1]*(1.-ds)+att[is+1].att[1]*ds;
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p->tone_masteratt[2]=att[is].att[2]*(1.-ds)+att[is+1].att[2]*ds;
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p->tone_centerboost=att[is].boost*(1.-ds)+att[is+1].boost*ds;
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p->tone_decay=att[is].decay*(1.-ds)+att[is+1].decay*ds;
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p->max_curve_dB=max[is]*(1.-ds)+max[is+1]*ds;
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for(i=0;i<P_BANDS;i++)
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p->toneatt[i]=in[is].block[i]*(1.-ds)+in[is+1].block[i]*ds;
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return;
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}
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static void vorbis_encode_compand_setup(vorbis_info *vi,double s,int block,
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const compandblock *in,
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const double *x){
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int i,is=s;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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ds=x[is]*(1.-ds)+x[is+1]*ds;
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is=(int)ds;
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ds-=is;
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if(ds==0 && is>0){
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is--;
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ds=1.;
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}
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/* interpolate the compander settings */
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for(i=0;i<NOISE_COMPAND_LEVELS;i++)
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p->noisecompand[i]=in[is].data[i]*(1.-ds)+in[is+1].data[i]*ds;
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return;
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}
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static void vorbis_encode_peak_setup(vorbis_info *vi,double s,int block,
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const int *suppress){
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int is=s;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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p->tone_abs_limit=suppress[is]*(1.-ds)+suppress[is+1]*ds;
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return;
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}
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static void vorbis_encode_noisebias_setup(vorbis_info *vi,double s,int block,
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const int *suppress,
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const noise3 *in,
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const noiseguard *guard,
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double userbias){
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int i,is=s,j;
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double ds=s-is;
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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p->noisemaxsupp=suppress[is]*(1.-ds)+suppress[is+1]*ds;
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p->noisewindowlomin=guard[block].lo;
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p->noisewindowhimin=guard[block].hi;
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p->noisewindowfixed=guard[block].fixed;
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for(j=0;j<P_NOISECURVES;j++)
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for(i=0;i<P_BANDS;i++)
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p->noiseoff[j][i]=in[is].data[j][i]*(1.-ds)+in[is+1].data[j][i]*ds;
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/* impulse blocks may take a user specified bias to boost the
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nominal/high noise encoding depth */
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for(j=0;j<P_NOISECURVES;j++){
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float min=p->noiseoff[j][0]+6; /* the lowest it can go */
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for(i=0;i<P_BANDS;i++){
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p->noiseoff[j][i]+=userbias;
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if(p->noiseoff[j][i]<min)p->noiseoff[j][i]=min;
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}
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}
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return;
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}
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static void vorbis_encode_ath_setup(vorbis_info *vi,int block){
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codec_setup_info *ci=vi->codec_setup;
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vorbis_info_psy *p=ci->psy_param[block];
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p->ath_adjatt=ci->hi.ath_floating_dB;
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p->ath_maxatt=ci->hi.ath_absolute_dB;
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return;
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}
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static int book_dup_or_new(codec_setup_info *ci,const static_codebook *book){
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int i;
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for(i=0;i<ci->books;i++)
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if(ci->book_param[i]==book)return(i);
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return(ci->books++);
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}
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static void vorbis_encode_blocksize_setup(vorbis_info *vi,double s,
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const int *shortb,const int *longb){
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codec_setup_info *ci=vi->codec_setup;
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int is=s;
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int blockshort=shortb[is];
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int blocklong=longb[is];
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ci->blocksizes[0]=blockshort;
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ci->blocksizes[1]=blocklong;
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}
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static void vorbis_encode_residue_setup(vorbis_info *vi,
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int number, int block,
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const vorbis_residue_template *res){
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codec_setup_info *ci=vi->codec_setup;
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int i;
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vorbis_info_residue0 *r=ci->residue_param[number]=
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_ogg_malloc(sizeof(*r));
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memcpy(r,res->res,sizeof(*r));
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if(ci->residues<=number)ci->residues=number+1;
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r->grouping=res->grouping;
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ci->residue_type[number]=res->res_type;
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/* fill in all the books */
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{
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int booklist=0,k;
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|
|
|
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);
|
|
}
|