/* * LAME MP3 encoding engine * * Copyright (c) 1999 Mark Taylor * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /* $Id: encoder.c,v 1.43 2001/03/12 04:38:35 markt Exp $ */ #ifdef HAVE_CONFIG_H#include <config.h>#endif #include <assert.h> #include "lame.h"#include "util.h"#include "newmdct.h"#include "psymodel.h"#include "quantize.h"#include "quantize_pvt.h"#include "bitstream.h"#include "VbrTag.h" #ifdef WITH_DMALLOC#include <dmalloc.h>#endif /* * auto-adjust of ATH, useful for low volume * Gabriel Bouvigne 3 feb 2001 * * modifies some values in * gfp->internal_flags->ATH * (gfc->ATH) */voidadjust_ATH( lame_global_flags* const gfp, FLOAT8 tot_ener[2][4] ){ lame_internal_flags* const gfc = gfp->internal_flags; int gr, channel; if (gfc->ATH->use_adjust) { FLOAT8 max_val = 0; for ( gr = 0; gr < gfc->mode_gr; ++gr ) for ( channel = 0; channel < gfc->channels_out; ++channel ) max_val = Max( max_val, tot_ener[gr][channel] ); /* scale to 0..1, and then rescale to 0..32767 */ max_val *= 32767/1e13; /* adjust ATH depending on range of maximum value */ if (vbr_mtrh == gfp->VBR) { /* this code reduces slowly the ATH (speed of 12 dB per second) * with some supporting stages to limit the reduction * 640 -> ~17 dB * : * 32640 -> ~0.01 dB */ FLOAT8 x = Max (640, 320*(int)(max_val/320)); x = x/32768; gfc->ATH->adjust *= gfc->ATH->decay; if (gfc->ATH->adjust < x) /* but not more than f(x) dB */ gfc->ATH->adjust = x; } else {#ifdef OLD_ATH_AUTO_ADJUST if (0.5 < max_val / 32768) { /* value above 50 % */ gfc->ATH->adjust = 1.0; /* do not reduce ATH */ } else if (0.3 < max_val / 32768) { /* value above 30 % */ gfc->ATH->adjust *= 0.955; /* reduce by ~0.2 dB */ if (gfc->ATH->adjust < 0.3) /* but ~5 dB in maximum */ gfc->ATH->adjust = 0.3; } else { /* value below 30 % */ gfc->ATH->adjust *= 0.93; /* reduce by ~0.3 dB */ if (gfc->ATH->adjust < 0.01) /* but 20 dB in maximum */ gfc->ATH->adjust = 0.01; }#else /* jd - 27 feb 2001 */ /* continuous curves based on approximation */ /* to GB's original values */ FLOAT8 max_val_n = max_val / 32768; FLOAT8 adj_lim_new; /* For an increase in approximate loudness, */ /* set ATH adjust to adjust_limit immediately*/ /* after a delay of one frame. */ /* For a loudness decrease, reduce ATH adjust*/ /* towards adjust_limit gradually. */ if( max_val_n > 0.25) { /* sqrt((1 - 0.01) / 15.84) from curve below*/ if( gfc->ATH->adjust >= 1.0) { gfc->ATH->adjust = 1.0; } else { /* preceding frame has lower ATH adjust; */ /* ascend only to the preceding adjust_limit */ /* in case there is leading low volume */ if( gfc->ATH->adjust < gfc->ATH->adjust_limit) { gfc->ATH->adjust = gfc->ATH->adjust_limit; } } gfc->ATH->adjust_limit = 1.0; } else { /* adjustment curve (parabolic) */ adj_lim_new = 15.84 * (max_val_n * max_val_n) + 0.01; if( gfc->ATH->adjust >= adj_lim_new) { /* descend gradually */ gfc->ATH->adjust *= adj_lim_new * 0.075 + 0.925; if( gfc->ATH->adjust < adj_lim_new) { /* stop descent */ gfc->ATH->adjust = adj_lim_new; } } else { /* ascend */ if( gfc->ATH->adjust_limit >= adj_lim_new) { gfc->ATH->adjust = adj_lim_new; } else { /* preceding frame has lower ATH adjust; */ /* ascend only to the preceding adjust_limit */ if( gfc->ATH->adjust < gfc->ATH->adjust_limit) { gfc->ATH->adjust = gfc->ATH->adjust_limit; } } } gfc->ATH->adjust_limit = adj_lim_new; }#endif } }} /************************************************************************** encodeframe() Layer 3** encode a single frame*************************************************************************lame_encode_frame() gr 0 gr 1inbuf: |--------------|---------------|-------------|MDCT output: |--------------|---------------|-------------| FFT's <---------1024----------> <---------1024--------> inbuf = buffer of PCM data size=MP3 framesize encoder acts on inbuf[ch][0], but output is delayed by MDCTDELAY so the MDCT coefficints are from inbuf[ch][-MDCTDELAY] psy-model FFT has a 1 granule delay, so we feed it data for the next granule. FFT is centered over granule: 224+576+224 So FFT starts at: 576-224-MDCTDELAY MPEG2: FFT ends at: BLKSIZE+576-224-MDCTDELAY MPEG1: FFT ends at: BLKSIZE+2*576-224-MDCTDELAY (1904) FFT starts at 576-224-MDCTDELAY (304) = 576-FFTOFFSET */ typedef FLOAT8 chgrdata[2][2]; int lame_encode_mp3_frame ( // Output lame_global_flags* const gfp, // Context sample_t* inbuf_l, // Input sample_t* inbuf_r, // Input unsigned char* mp3buf, // Output int mp3buf_size ) // Output{#ifdef macintosh /* PLL 14/04/2000 */ static FLOAT8 xr[2][2][576]; static int l3_enc[2][2][576];#else FLOAT8 xr[2][2][576]; int l3_enc[2][2][576];#endif int mp3count; III_psy_ratio masking_LR[2][2]; /*LR masking & energy */ III_psy_ratio masking_MS[2][2]; /*MS masking & energy */ III_psy_ratio (*masking)[2][2]; /*pointer to selected maskings*/ III_scalefac_t scalefac[2][2]; const sample_t *inbuf[2]; lame_internal_flags *gfc=gfp->internal_flags; FLOAT8 tot_ener[2][4]; FLOAT8 ms_ener_ratio[2]={.5,.5}; chgrdata pe,pe_MS; chgrdata *pe_use; int ch,gr,mean_bits; int bitsPerFrame; int check_ms_stereo; FLOAT8 ms_ratio_next = 0.; FLOAT8 ms_ratio_prev = 0.; memset((char *) masking_LR, 0, sizeof(masking_LR)); memset((char *) masking_MS, 0, sizeof(masking_MS)); memset((char *) scalefac, 0, sizeof(scalefac)); inbuf[0]=inbuf_l; inbuf[1]=inbuf_r; check_ms_stereo = (gfp->mode == JOINT_STEREO); gfc->mode_ext = MPG_MD_LR_LR; if (gfc->lame_encode_frame_init==0 ) { gfc->lame_encode_frame_init=1; /* padding method as described in * "MPEG-Layer3 / Bitstream Syntax and Decoding" * by Martin Sieler, Ralph Sperschneider * * note: there is no padding for the very first frame * * Robert.Hegemann@gmx.de 2000-06-22 */ gfc->frac_SpF = ((gfp->version+1)*72000L*gfp->brate) % gfp->out_samplerate; gfc->slot_lag = gfc->frac_SpF; /* check FFT will not use a negative starting offset */#if 576 < FFTOFFSET# error FFTOFFSET greater than 576: FFT uses a negative offset#endif /* check if we have enough data for FFT */ assert(gfc->mf_size>=(BLKSIZE+gfp->framesize-FFTOFFSET)); /* check if we have enough data for polyphase filterbank */ /* it needs 1152 samples + 286 samples ignored for one granule */ /* 1152+576+286 samples for two granules */ assert(gfc->mf_size>=(286+576*(1+gfc->mode_gr))); /* prime the MDCT/polyphase filterbank with a short block */ { int i,j; sample_t primebuff0[286+1152+576]; sample_t primebuff1[286+1152+576]; for (i=0, j=0; i<286+576*(1+gfc->mode_gr); ++i) { if (i<576*gfc->mode_gr) { primebuff0[i]=0; if (gfc->channels_out==2) primebuff1[i]=0; }else{ primebuff0[i]=inbuf[0][j]; if (gfc->channels_out==2) primebuff1[i]=inbuf[1][j]; ++j; } } /* polyphase filtering / mdct */ for ( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->l3_side.gr[gr].ch[ch].tt.block_type=SHORT_TYPE; } } mdct_sub48(gfc, primebuff0, primebuff1, xr); } iteration_init(gfp); /* prepare for ATH auto adjustment: * we want to decrease the ATH by 12 dB per second */ { FLOAT8 frame_duration = 576. * gfc->mode_gr / gfp->out_samplerate; gfc->ATH->decay = pow(10., -12./10. * frame_duration); gfc->ATH->adjust = 1.0; gfc->ATH->adjust_limit = 0.01; } } /********************** padding *****************************/ switch (gfp->padding_type) { case 0: gfc->padding=0; break; case 1: gfc->padding=1; break; case 2: default: if (gfp->VBR!=vbr_off) { gfc->padding=0; } else { if (gfp->disable_reservoir) { gfc->padding = 0; /* if the user specified --nores, dont very gfc->padding either */ /* tiny changes in frac_SpF rounding will cause file differences */ }else{ /* padding method as described in * "MPEG-Layer3 / Bitstream Syntax and Decoding" * by Martin Sieler, Ralph Sperschneider * * note: there is no padding for the very first frame * * Robert.Hegemann@gmx.de 2000-06-22 */ gfc->slot_lag -= gfc->frac_SpF; if (gfc->slot_lag < 0) { gfc->slot_lag += gfp->out_samplerate; gfc->padding = 1; } else { gfc->padding = 0; } } /* reservoir enabled */ } } if (gfc->psymodel) { /* psychoacoustic model * psy model has a 1 granule (576) delay that we must compensate for * (mt 6/99). */ int ret; const sample_t *bufp[2]; /* address of beginning of left & right granule */ int blocktype[2]; ms_ratio_prev=gfc->ms_ratio[gfc->mode_gr-1]; for (gr=0; gr < gfc->mode_gr ; gr++) { for ( ch = 0; ch < gfc->channels_out; ch++ ) bufp[ch] = &inbuf[ch][576 + gr*576-FFTOFFSET]; if (gfc->nsPsy.use) { ret=L3psycho_anal_ns( gfp, bufp, gr, &gfc->ms_ratio[gr],&ms_ratio_next, masking_LR, masking_MS, pe[gr],pe_MS[gr],tot_ener[gr],blocktype); } else { ret=L3psycho_anal( gfp, bufp, gr, &gfc->ms_ratio[gr],&ms_ratio_next, masking_LR, masking_MS, pe[gr],pe_MS[gr],tot_ener[gr],blocktype); } if (ret!=0) return -4; for ( ch = 0; ch < gfc->channels_out; ch++ ) gfc->l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch]; if (check_ms_stereo) { ms_ener_ratio[gr] = tot_ener[gr][2]+tot_ener[gr][3]; if (ms_ener_ratio[gr]>0) ms_ener_ratio[gr] = tot_ener[gr][3]/ms_ener_ratio[gr]; } } }else{ for (gr=0; gr < gfc->mode_gr ; gr++) for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE; pe_MS[gr][ch]=pe[gr][ch]=700; } } /* auto-adjust of ATH, useful for low volume */ adjust_ATH( gfp, tot_ener ); /* block type flags */ for( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gr_info *cod_info = &gfc->l3_side.gr[gr].ch[ch].tt; cod_info->mixed_block_flag = 0; /* never used by this model */ if (cod_info->block_type == NORM_TYPE ) cod_info->window_switching_flag = 0; else cod_info->window_switching_flag = 1; } } /* polyphase filtering / mdct */ mdct_sub48(gfc, inbuf[0], inbuf[1], xr); /* re-order the short blocks, for more efficient encoding below */ for (gr = 0; gr < gfc->mode_gr; gr++) { for (ch = 0; ch < gfc->channels_out; ch++) { gr_info *cod_info = &gfc->l3_side.gr[gr].ch[ch].tt; if (cod_info->block_type==SHORT_TYPE) { freorder(gfc->scalefac_band.s,xr[gr][ch]); } } } /* use m/s gfc->channels_out? */ if (check_ms_stereo) { int gr0 = 0, gr1 = gfc->mode_gr-1; /* make sure block type is the same in each channel */ check_ms_stereo = (gfc->l3_side.gr[gr0].ch[0].tt.block_type==gfc->l3_side.gr[gr0].ch[1].tt.block_type) && (gfc->l3_side.gr[gr1].ch[0].tt.block_type==gfc->l3_side.gr[gr1].ch[1].tt.block_type); } /* Here will be selected MS or LR coding of the 2 stereo channels */ assert ( gfc->mode_ext == MPG_MD_LR_LR ); gfc->mode_ext = MPG_MD_LR_LR; if (gfp->force_ms) { gfc->mode_ext = MPG_MD_MS_LR; } else if (check_ms_stereo) { /* ms_ratio = is scaled, for historical reasons, to look like a ratio of side_channel / total. 0 = signal is 100% mono .5 = L & R uncorrelated */ /* [0] and [1] are the results for the two granules in MPEG-1, * in MPEG-2 it's only a faked averaging of the same value * _prev is the value of the last granule of the previous frame * _next is the value of the first granule of the next frame */ FLOAT8 ms_ratio_ave1; FLOAT8 ms_ratio_ave2; FLOAT8 threshold1 = 0.35; FLOAT8 threshold2 = 0.45; /* take an average */ if (gfc->mode_gr==1) { /* MPEG2 - no second granule */ ms_ratio_ave1 = 0.33 * ( gfc->ms_ratio[0] + ms_ratio_prev + ms_ratio_next ); ms_ratio_ave2 = gfc->ms_ratio[0]; }else{ ms_ratio_ave1 = 0.25 * ( gfc->ms_ratio[0] + gfc->ms_ratio[1] + ms_ratio_prev + ms_ratio_next ); ms_ratio_ave2 = 0.50 * ( gfc->ms_ratio[0] + gfc->ms_ratio[1] ); } if (gfp->mode_automs) { if ( gfp->compression_ratio < 11.025 ) { /* 11.025 => 1, 6.3 => 0 */ double thr = (gfp->compression_ratio - 6.3) / (11.025 - 6.3); if (thr<0) thr=0; threshold1 *= thr; threshold2 *= thr; } } if ((ms_ratio_ave1 < threshold1 && ms_ratio_ave2 < threshold2) || gfc->nsPsy.use) { int sum_pe_MS = pe_MS[0][0] + pe_MS[0][1] + pe_MS[1][0] + pe_MS[1][1]; int sum_pe_LR = pe [0][0] + pe [0][1] + pe [1][0] + pe [1][1]; /* based on PE: M/S coding would not use much more bits than L/R coding */ if (sum_pe_MS <= 1.07 * sum_pe_LR && !gfc->nsPsy.use) gfc->mode_ext = MPG_MD_MS_LR; if (sum_pe_MS <= 1.00 * sum_pe_LR && gfc->nsPsy.use) gfc->mode_ext = MPG_MD_MS_LR; } } /* copy data for MP3 frame analyzer */ if (gfp->analysis && gfc->pinfo != NULL) { for ( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->pinfo->ms_ratio[gr]=gfc->ms_ratio[gr]; gfc->pinfo->ms_ener_ratio[gr]=ms_ener_ratio[gr]; gfc->pinfo->blocktype[gr][ch]= gfc->l3_side.gr[gr].ch[ch].tt.block_type; memcpy(gfc->pinfo->xr[gr][ch],xr[gr][ch],sizeof(xr[gr][ch])); /* in psymodel, LR and MS data was stored in pinfo. switch to MS data: */ if (gfc->mode_ext==MPG_MD_MS_LR) { gfc->pinfo->pe[gr][ch]=gfc->pinfo->pe[gr][ch+2]; gfc->pinfo->ers[gr][ch]=gfc->pinfo->ers[gr][ch+2]; memcpy(gfc->pinfo->energy[gr][ch],gfc->pinfo->energy[gr][ch+2], sizeof(gfc->pinfo->energy[gr][ch])); } } } } /* bit and noise allocation */ if (MPG_MD_MS_LR == gfc->mode_ext) { masking = &masking_MS; /* use MS masking */ pe_use = &pe_MS;Lines 1–500 of 584. 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/* * LAME MP3 encoding engine * * Copyright (c) 1999 Mark Taylor * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /* $Id: encoder.c,v 1.43 2001/03/12 04:38:35 markt Exp $ */ #ifdef HAVE_CONFIG_H #include <config.h> #endif #include <assert.h> #include "lame.h" #include "util.h" #include "newmdct.h" #include "psymodel.h" #include "quantize.h" #include "quantize_pvt.h" #include "bitstream.h" #include "VbrTag.h" #ifdef WITH_DMALLOC #include <dmalloc.h> #endif /* * auto-adjust of ATH, useful for low volume * Gabriel Bouvigne 3 feb 2001 * * modifies some values in * gfp->internal_flags->ATH * (gfc->ATH) */ void adjust_ATH( lame_global_flags* const gfp, FLOAT8 tot_ener[2][4] ) { lame_internal_flags* const gfc = gfp->internal_flags; int gr, channel; if (gfc->ATH->use_adjust) { FLOAT8 max_val = 0; for ( gr = 0; gr < gfc->mode_gr; ++gr ) for ( channel = 0; channel < gfc->channels_out; ++channel ) max_val = Max( max_val, tot_ener[gr][channel] ); /* scale to 0..1, and then rescale to 0..32767 */ max_val *= 32767/1e13; /* adjust ATH depending on range of maximum value */ if (vbr_mtrh == gfp->VBR) { /* this code reduces slowly the ATH (speed of 12 dB per second) * with some supporting stages to limit the reduction * 640 -> ~17 dB * : * 32640 -> ~0.01 dB */ FLOAT8 x = Max (640, 320*(int)(max_val/320)); x = x/32768; gfc->ATH->adjust *= gfc->ATH->decay; if (gfc->ATH->adjust < x) /* but not more than f(x) dB */ gfc->ATH->adjust = x; } else { #ifdef OLD_ATH_AUTO_ADJUST if (0.5 < max_val / 32768) { /* value above 50 % */ gfc->ATH->adjust = 1.0; /* do not reduce ATH */ } else if (0.3 < max_val / 32768) { /* value above 30 % */ gfc->ATH->adjust *= 0.955; /* reduce by ~0.2 dB */ if (gfc->ATH->adjust < 0.3) /* but ~5 dB in maximum */ gfc->ATH->adjust = 0.3; } else { /* value below 30 % */ gfc->ATH->adjust *= 0.93; /* reduce by ~0.3 dB */ if (gfc->ATH->adjust < 0.01) /* but 20 dB in maximum */ gfc->ATH->adjust = 0.01; } #else /* jd - 27 feb 2001 */ /* continuous curves based on approximation */ /* to GB's original values */ FLOAT8 max_val_n = max_val / 32768; FLOAT8 adj_lim_new; /* For an increase in approximate loudness, */ /* set ATH adjust to adjust_limit immediately*/ /* after a delay of one frame. */ /* For a loudness decrease, reduce ATH adjust*/ /* towards adjust_limit gradually. */ if( max_val_n > 0.25) { /* sqrt((1 - 0.01) / 15.84) from curve below*/ if( gfc->ATH->adjust >= 1.0) { gfc->ATH->adjust = 1.0; } else { /* preceding frame has lower ATH adjust; */ /* ascend only to the preceding adjust_limit */ /* in case there is leading low volume */ if( gfc->ATH->adjust < gfc->ATH->adjust_limit) { gfc->ATH->adjust = gfc->ATH->adjust_limit; } } gfc->ATH->adjust_limit = 1.0; } else { /* adjustment curve (parabolic) */ adj_lim_new = 15.84 * (max_val_n * max_val_n) + 0.01; if( gfc->ATH->adjust >= adj_lim_new) { /* descend gradually */ gfc->ATH->adjust *= adj_lim_new * 0.075 + 0.925; if( gfc->ATH->adjust < adj_lim_new) { /* stop descent */ gfc->ATH->adjust = adj_lim_new; } } else { /* ascend */ if( gfc->ATH->adjust_limit >= adj_lim_new) { gfc->ATH->adjust = adj_lim_new; } else { /* preceding frame has lower ATH adjust; */ /* ascend only to the preceding adjust_limit */ if( gfc->ATH->adjust < gfc->ATH->adjust_limit) { gfc->ATH->adjust = gfc->ATH->adjust_limit; } } } gfc->ATH->adjust_limit = adj_lim_new; } #endif } } } /************************************************************************ * * encodeframe() Layer 3 * * encode a single frame * ************************************************************************ lame_encode_frame() gr 0 gr 1 inbuf: |--------------|---------------|-------------| MDCT output: |--------------|---------------|-------------| FFT's <---------1024----------> <---------1024--------> inbuf = buffer of PCM data size=MP3 framesize encoder acts on inbuf[ch][0], but output is delayed by MDCTDELAY so the MDCT coefficints are from inbuf[ch][-MDCTDELAY] psy-model FFT has a 1 granule delay, so we feed it data for the next granule. FFT is centered over granule: 224+576+224 So FFT starts at: 576-224-MDCTDELAY MPEG2: FFT ends at: BLKSIZE+576-224-MDCTDELAY MPEG1: FFT ends at: BLKSIZE+2*576-224-MDCTDELAY (1904) FFT starts at 576-224-MDCTDELAY (304) = 576-FFTOFFSET */ typedef FLOAT8 chgrdata[2][2]; int lame_encode_mp3_frame ( // Output lame_global_flags* const gfp, // Context sample_t* inbuf_l, // Input sample_t* inbuf_r, // Input unsigned char* mp3buf, // Output int mp3buf_size ) // Output { #ifdef macintosh /* PLL 14/04/2000 */ static FLOAT8 xr[2][2][576]; static int l3_enc[2][2][576]; #else FLOAT8 xr[2][2][576]; int l3_enc[2][2][576]; #endif int mp3count; III_psy_ratio masking_LR[2][2]; /*LR masking & energy */ III_psy_ratio masking_MS[2][2]; /*MS masking & energy */ III_psy_ratio (*masking)[2][2]; /*pointer to selected maskings*/ III_scalefac_t scalefac[2][2]; const sample_t *inbuf[2]; lame_internal_flags *gfc=gfp->internal_flags; FLOAT8 tot_ener[2][4]; FLOAT8 ms_ener_ratio[2]={.5,.5}; chgrdata pe,pe_MS; chgrdata *pe_use; int ch,gr,mean_bits; int bitsPerFrame; int check_ms_stereo; FLOAT8 ms_ratio_next = 0.; FLOAT8 ms_ratio_prev = 0.; memset((char *) masking_LR, 0, sizeof(masking_LR)); memset((char *) masking_MS, 0, sizeof(masking_MS)); memset((char *) scalefac, 0, sizeof(scalefac)); inbuf[0]=inbuf_l; inbuf[1]=inbuf_r; check_ms_stereo = (gfp->mode == JOINT_STEREO); gfc->mode_ext = MPG_MD_LR_LR; if (gfc->lame_encode_frame_init==0 ) { gfc->lame_encode_frame_init=1; /* padding method as described in * "MPEG-Layer3 / Bitstream Syntax and Decoding" * by Martin Sieler, Ralph Sperschneider * * note: there is no padding for the very first frame * * Robert.Hegemann@gmx.de 2000-06-22 */ gfc->frac_SpF = ((gfp->version+1)*72000L*gfp->brate) % gfp->out_samplerate; gfc->slot_lag = gfc->frac_SpF; /* check FFT will not use a negative starting offset */ #if 576 < FFTOFFSET # error FFTOFFSET greater than 576: FFT uses a negative offset #endif /* check if we have enough data for FFT */ assert(gfc->mf_size>=(BLKSIZE+gfp->framesize-FFTOFFSET)); /* check if we have enough data for polyphase filterbank */ /* it needs 1152 samples + 286 samples ignored for one granule */ /* 1152+576+286 samples for two granules */ assert(gfc->mf_size>=(286+576*(1+gfc->mode_gr))); /* prime the MDCT/polyphase filterbank with a short block */ { int i,j; sample_t primebuff0[286+1152+576]; sample_t primebuff1[286+1152+576]; for (i=0, j=0; i<286+576*(1+gfc->mode_gr); ++i) { if (i<576*gfc->mode_gr) { primebuff0[i]=0; if (gfc->channels_out==2) primebuff1[i]=0; }else{ primebuff0[i]=inbuf[0][j]; if (gfc->channels_out==2) primebuff1[i]=inbuf[1][j]; ++j; } } /* polyphase filtering / mdct */ for ( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->l3_side.gr[gr].ch[ch].tt.block_type=SHORT_TYPE; } } mdct_sub48(gfc, primebuff0, primebuff1, xr); } iteration_init(gfp); /* prepare for ATH auto adjustment: * we want to decrease the ATH by 12 dB per second */ { FLOAT8 frame_duration = 576. * gfc->mode_gr / gfp->out_samplerate; gfc->ATH->decay = pow(10., -12./10. * frame_duration); gfc->ATH->adjust = 1.0; gfc->ATH->adjust_limit = 0.01; } } /********************** padding *****************************/ switch (gfp->padding_type) { case 0: gfc->padding=0; break; case 1: gfc->padding=1; break; case 2: default: if (gfp->VBR!=vbr_off) { gfc->padding=0; } else { if (gfp->disable_reservoir) { gfc->padding = 0; /* if the user specified --nores, dont very gfc->padding either */ /* tiny changes in frac_SpF rounding will cause file differences */ }else{ /* padding method as described in * "MPEG-Layer3 / Bitstream Syntax and Decoding" * by Martin Sieler, Ralph Sperschneider * * note: there is no padding for the very first frame * * Robert.Hegemann@gmx.de 2000-06-22 */ gfc->slot_lag -= gfc->frac_SpF; if (gfc->slot_lag < 0) { gfc->slot_lag += gfp->out_samplerate; gfc->padding = 1; } else { gfc->padding = 0; } } /* reservoir enabled */ } } if (gfc->psymodel) { /* psychoacoustic model * psy model has a 1 granule (576) delay that we must compensate for * (mt 6/99). */ int ret; const sample_t *bufp[2]; /* address of beginning of left & right granule */ int blocktype[2]; ms_ratio_prev=gfc->ms_ratio[gfc->mode_gr-1]; for (gr=0; gr < gfc->mode_gr ; gr++) { for ( ch = 0; ch < gfc->channels_out; ch++ ) bufp[ch] = &inbuf[ch][576 + gr*576-FFTOFFSET]; if (gfc->nsPsy.use) { ret=L3psycho_anal_ns( gfp, bufp, gr, &gfc->ms_ratio[gr],&ms_ratio_next, masking_LR, masking_MS, pe[gr],pe_MS[gr],tot_ener[gr],blocktype); } else { ret=L3psycho_anal( gfp, bufp, gr, &gfc->ms_ratio[gr],&ms_ratio_next, masking_LR, masking_MS, pe[gr],pe_MS[gr],tot_ener[gr],blocktype); } if (ret!=0) return -4; for ( ch = 0; ch < gfc->channels_out; ch++ ) gfc->l3_side.gr[gr].ch[ch].tt.block_type=blocktype[ch]; if (check_ms_stereo) { ms_ener_ratio[gr] = tot_ener[gr][2]+tot_ener[gr][3]; if (ms_ener_ratio[gr]>0) ms_ener_ratio[gr] = tot_ener[gr][3]/ms_ener_ratio[gr]; } } }else{ for (gr=0; gr < gfc->mode_gr ; gr++) for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->l3_side.gr[gr].ch[ch].tt.block_type=NORM_TYPE; pe_MS[gr][ch]=pe[gr][ch]=700; } } /* auto-adjust of ATH, useful for low volume */ adjust_ATH( gfp, tot_ener ); /* block type flags */ for( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gr_info *cod_info = &gfc->l3_side.gr[gr].ch[ch].tt; cod_info->mixed_block_flag = 0; /* never used by this model */ if (cod_info->block_type == NORM_TYPE ) cod_info->window_switching_flag = 0; else cod_info->window_switching_flag = 1; } } /* polyphase filtering / mdct */ mdct_sub48(gfc, inbuf[0], inbuf[1], xr); /* re-order the short blocks, for more efficient encoding below */ for (gr = 0; gr < gfc->mode_gr; gr++) { for (ch = 0; ch < gfc->channels_out; ch++) { gr_info *cod_info = &gfc->l3_side.gr[gr].ch[ch].tt; if (cod_info->block_type==SHORT_TYPE) { freorder(gfc->scalefac_band.s,xr[gr][ch]); } } } /* use m/s gfc->channels_out? */ if (check_ms_stereo) { int gr0 = 0, gr1 = gfc->mode_gr-1; /* make sure block type is the same in each channel */ check_ms_stereo = (gfc->l3_side.gr[gr0].ch[0].tt.block_type==gfc->l3_side.gr[gr0].ch[1].tt.block_type) && (gfc->l3_side.gr[gr1].ch[0].tt.block_type==gfc->l3_side.gr[gr1].ch[1].tt.block_type); } /* Here will be selected MS or LR coding of the 2 stereo channels */ assert ( gfc->mode_ext == MPG_MD_LR_LR ); gfc->mode_ext = MPG_MD_LR_LR; if (gfp->force_ms) { gfc->mode_ext = MPG_MD_MS_LR; } else if (check_ms_stereo) { /* ms_ratio = is scaled, for historical reasons, to look like a ratio of side_channel / total. 0 = signal is 100% mono .5 = L & R uncorrelated */ /* [0] and [1] are the results for the two granules in MPEG-1, * in MPEG-2 it's only a faked averaging of the same value * _prev is the value of the last granule of the previous frame * _next is the value of the first granule of the next frame */ FLOAT8 ms_ratio_ave1; FLOAT8 ms_ratio_ave2; FLOAT8 threshold1 = 0.35; FLOAT8 threshold2 = 0.45; /* take an average */ if (gfc->mode_gr==1) { /* MPEG2 - no second granule */ ms_ratio_ave1 = 0.33 * ( gfc->ms_ratio[0] + ms_ratio_prev + ms_ratio_next ); ms_ratio_ave2 = gfc->ms_ratio[0]; }else{ ms_ratio_ave1 = 0.25 * ( gfc->ms_ratio[0] + gfc->ms_ratio[1] + ms_ratio_prev + ms_ratio_next ); ms_ratio_ave2 = 0.50 * ( gfc->ms_ratio[0] + gfc->ms_ratio[1] ); } if (gfp->mode_automs) { if ( gfp->compression_ratio < 11.025 ) { /* 11.025 => 1, 6.3 => 0 */ double thr = (gfp->compression_ratio - 6.3) / (11.025 - 6.3); if (thr<0) thr=0; threshold1 *= thr; threshold2 *= thr; } } if ((ms_ratio_ave1 < threshold1 && ms_ratio_ave2 < threshold2) || gfc->nsPsy.use) { int sum_pe_MS = pe_MS[0][0] + pe_MS[0][1] + pe_MS[1][0] + pe_MS[1][1]; int sum_pe_LR = pe [0][0] + pe [0][1] + pe [1][0] + pe [1][1]; /* based on PE: M/S coding would not use much more bits than L/R coding */ if (sum_pe_MS <= 1.07 * sum_pe_LR && !gfc->nsPsy.use) gfc->mode_ext = MPG_MD_MS_LR; if (sum_pe_MS <= 1.00 * sum_pe_LR && gfc->nsPsy.use) gfc->mode_ext = MPG_MD_MS_LR; } } /* copy data for MP3 frame analyzer */ if (gfp->analysis && gfc->pinfo != NULL) { for ( gr = 0; gr < gfc->mode_gr; gr++ ) { for ( ch = 0; ch < gfc->channels_out; ch++ ) { gfc->pinfo->ms_ratio[gr]=gfc->ms_ratio[gr]; gfc->pinfo->ms_ener_ratio[gr]=ms_ener_ratio[gr]; gfc->pinfo->blocktype[gr][ch]= gfc->l3_side.gr[gr].ch[ch].tt.block_type; memcpy(gfc->pinfo->xr[gr][ch],xr[gr][ch],sizeof(xr[gr][ch])); /* in psymodel, LR and MS data was stored in pinfo. switch to MS data: */ if (gfc->mode_ext==MPG_MD_MS_LR) { gfc->pinfo->pe[gr][ch]=gfc->pinfo->pe[gr][ch+2]; gfc->pinfo->ers[gr][ch]=gfc->pinfo->ers[gr][ch+2]; memcpy(gfc->pinfo->energy[gr][ch],gfc->pinfo->energy[gr][ch+2], sizeof(gfc->pinfo->energy[gr][ch])); } } } } /* bit and noise allocation */ if (MPG_MD_MS_LR == gfc->mode_ext) { masking = &masking_MS; /* use MS masking */ pe_use = &pe_MS;