/*
* LAME MP3 encoding engine
*
* Copyright (c) 1999 Mark Taylor
* Copyright (c) 2000-2002 Takehiro Tominaga
* Copyright (c) 2000-2005 Robert Hegemann
* Copyright (c) 2001 Gabriel Bouvigne
* Copyright (c) 2001 John Dahlstrom
*
* 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 1.90.2.1 2005/11/20 14:08:24 bouvigne 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"
#include "vbrquantize.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)
*/
static void
adjust_ATH(lame_internal_flags * const gfc)
{
FLOAT gr2_max, max_pow;
if (gfc->ATH->use_adjust == 0) {
gfc->ATH->adjust = 1.0; /* no adjustment */
return;
}
/* jd - 2001 mar 12, 27, jun 30 */
/* loudness based on equal loudness curve; */
/* use granule with maximum combined loudness */
max_pow = gfc->loudness_sq[0][0];
gr2_max = gfc->loudness_sq[1][0];
if (gfc->channels_out == 2) {
max_pow += gfc->loudness_sq[0][1];
gr2_max += gfc->loudness_sq[1][1];
}
else {
max_pow += max_pow;
gr2_max += gr2_max;
}
if (gfc->mode_gr == 2) {
max_pow = Max(max_pow, gr2_max);
}
max_pow *= 0.5; /* max_pow approaches 1.0 for full band noise */
/* jd - 2001 mar 31, jun 30 */
/* user tuning of ATH adjustment region */
max_pow *= gfc->ATH->aa_sensitivity_p;
/* adjust ATH depending on range of maximum value
*/
/* jd - 2001 feb27, mar12,20, jun30, jul22 */
/* continuous curves based on approximation */
/* to GB's original values. */
/* 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. */
/* max_pow is a loudness squared or a power. */
if (max_pow > 0.03125) { /* ((1 - 0.000625)/ 31.98) 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 */
/* about 32 dB maximum adjust (0.000625) */
FLOAT adj_lim_new = 31.98 * max_pow + 0.000625;
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;
}
}
/***********************************************************************
*
* some simple statistics
*
* bitrate index 0: free bitrate -> not allowed in VBR mode
* : bitrates, kbps depending on MPEG version
* bitrate index 15: forbidden
*
* mode_ext:
* 0: LR
* 1: LR-i
* 2: MS
* 3: MS-i
*
***********************************************************************/
static void
updateStats(lame_internal_flags * const gfc)
{
int gr, ch;
assert(gfc->bitrate_index < 16u);
assert(gfc->mode_ext < 4u);
/* count bitrate indices */
gfc->bitrate_stereoMode_Hist[gfc->bitrate_index][4]++;
gfc->bitrate_stereoMode_Hist[15][4]++;
/* count 'em for every mode extension in case of 2 channel encoding */
if (gfc->channels_out == 2) {
gfc->bitrate_stereoMode_Hist[gfc->bitrate_index][gfc->mode_ext]++;
gfc->bitrate_stereoMode_Hist[15][gfc->mode_ext]++;
}
for (gr = 0; gr < gfc->mode_gr; ++gr) {
for (ch = 0; ch < gfc->channels_out; ++ch) {
int bt = gfc->l3_side.tt[gr][ch].block_type;
int mf = gfc->l3_side.tt[gr][ch].mixed_block_flag;
if (mf)
bt = 4;
gfc->bitrate_blockType_Hist[gfc->bitrate_index][bt]++;
gfc->bitrate_blockType_Hist[gfc->bitrate_index][5]++;
gfc->bitrate_blockType_Hist[15][bt]++;
gfc->bitrate_blockType_Hist[15][5]++;
}
}
}
static void
lame_encode_frame_init(lame_global_flags * const gfp, const sample_t * inbuf[2])
{
lame_internal_flags *gfc = gfp->internal_flags;
int ch, gr;
if (gfc->lame_encode_frame_init == 0) {
/* prime the MDCT/polyphase filterbank with a short block */
int i, j;
sample_t primebuff0[286 + 1152 + 576];
sample_t primebuff1[286 + 1152 + 576];
gfc->lame_encode_frame_init = 1;
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.tt[gr][ch].block_type = SHORT_TYPE;
}
}
mdct_sub48(gfc, primebuff0, primebuff1);
/* 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 */
assert(gfc->mf_size >= (512 + gfp->framesize - 32));
}
}
/************************************************************************
*
* encodeframe() Layer 3
*
* encode a single frame
*
************************************************************************
lame_encode_frame()
gr 0 gr 1
inbuf: |--------------|--------------|--------------|
Polyphase (18 windows, each shifted 32)
gr 0:
window1 <----512---->
window18 <----512---->
gr 1:
window1 <----512---->
window18 <----512---->
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 (1328)
MPEG1: FFT ends at: BLKSIZE+2*576-224-MDCTDELAY (1904)
MPEG2: polyphase first window: [0..511]
18th window: [544..1055] (1056)
MPEG1: 36th window: [1120..1631] (1632)
data needed: 512+framesize-32
A close look newmdct.c shows that the polyphase filterbank
only uses data from [0..510] for each window. Perhaps because the window
used by the filterbank is zero for the last point, so Takehiro's
code doesn't bother to compute with it.
FFT starts at 576-224-MDCTDELAY (304) = 576-FFTOFFSET
*/
typedef FLOAT 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 */
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 */
const sample_t *inbuf[2];
lame_internal_flags *gfc = gfp->internal_flags;
FLOAT tot_ener[2][4];
FLOAT ms_ener_ratio[2] = { .5, .5 };
chgrdata pe, pe_MS;
chgrdata *pe_use;
int ch, gr;
FLOAT ms_ratio_next = 0.;
FLOAT ms_ratio_prev = 0.;
inbuf[0] = inbuf_l;
inbuf[1] = inbuf_r;
if (gfc->lame_encode_frame_init == 0) {
/*first run? */
lame_encode_frame_init(gfp, inbuf);
}
/********************** padding *****************************/
/* 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 2000-06-22
*/
gfc->padding = FALSE;
if ((gfc->slot_lag -= gfc->frac_SpF) < 0) {
gfc->slot_lag += gfp->out_samplerate;
gfc->padding = TRUE;
}
/****************************************
* Stage 1: psychoacoustic model *
****************************************/
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 (gfp->psymodel == PSY_NSPSYTUNE) {
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;
if (gfp->mode == JOINT_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];
}
/* block type flags */
for (ch = 0; ch < gfc->channels_out; ch++) {
gr_info *cod_info = &gfc->l3_side.tt[gr][ch];
cod_info->block_type = blocktype[ch];
cod_info->mixed_block_flag = 0;
}
}
}
else {
/*no psy model */
memset((char *) masking_LR, 0, sizeof(masking_LR));
memset((char *) masking_MS, 0, sizeof(masking_MS));
for (gr = 0; gr < gfc->mode_gr; gr++)
for (ch = 0; ch < gfc->channels_out; ch++) {
gfc->l3_side.tt[gr][ch].block_type = NORM_TYPE;
gfc->l3_side.tt[gr][ch].mixed_block_flag = 0;
pe_MS[gr][ch] = pe[gr][ch] = 700;
}
}
/* auto-adjust of ATH, useful for low volume */
adjust_ATH(gfc);
/****************************************
* Stage 2: MDCT *
****************************************/
/* polyphase filtering / mdct */
mdct_sub48(gfc, inbuf[0], inbuf[1]);
/****************************************
* Stage 3: MS/LR decision *
****************************************/
/* Here will be selected MS or LR coding of the 2 stereo channels */
gfc->mode_ext = MPG_MD_LR_LR;
if (gfp->force_ms) {
gfc->mode_ext = MPG_MD_MS_LR;
}
else if (gfp->mode == JOINT_STEREO) {
int check_ms_stereo = 1;
/* 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
*/
if (gfp->psymodel == PSY_GPSYCHO) {
FLOAT ms_ratio_ave1;
FLOAT ms_ratio_ave2;
FLOAT threshold1 = 0.35;
FLOAT 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 (ms_ratio_ave1 >= threshold1 || ms_ratio_ave2 >= threshold2)
check_ms_stereo = 0;
}
if (check_ms_stereo) {
FLOAT sum_pe_MS = 0;
FLOAT sum_pe_LR = 0;
for (gr = 0; gr < gfc->mode_gr; gr++) {
for (ch = 0; ch < gfc->channels_out; ch++) {
sum_pe_MS += pe_MS[gr][ch];
sum_pe_LR += pe[gr][ch];
}
}
/* based on PE: M/S coding would not use much more bits than L/R */
if (((gfp->psymodel == PSY_GPSYCHO) && sum_pe_MS <= 1.07 * sum_pe_LR) ||
((gfp->psymodel == PSY_NSPSYTUNE) && sum_pe_MS <= 1.00 * sum_pe_LR)) {
gr_info *gi0 = &gfc->l3_side.tt[0][0];
gr_info *gi1 = &gfc->l3_side.tt[gfc->mode_gr - 1][0];
if (gi0[0].block_type == gi0[1].block_type &&
gi1[0].block_type == gi1[1].block_type) {
gfc->mode_ext = MPG_MD_MS_LR;
}
}
}
}
/* bit and noise allocation */
if (gfc->mode_ext == MPG_MD_MS_LR) {
masking = &masking_MS; /* use MS masking */
pe_use = &pe_MS;
}
else {
masking = &masking_LR; /* use LR masking */
pe_use = &pe;
}
#if defined(HAVE_GTK)
/* 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.tt[gr][ch].block_type;
gfc->pinfo->pe[gr][ch] = (*pe_use)[gr][ch];
memcpy(gfc->pinfo->xr[gr][ch], &gfc->l3_side.tt[gr][ch].xr, sizeof(FLOAT) * 576);
/* in psymodel, LR and MS data was stored in pinfo.
switch to MS data: */
if (gfc->mode_ext == MPG_MD_MS_LR) {
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]));
}
}
}
}
#endif
/****************************************
* Stage 4: quantization loop *
****************************************/
if (gfp->psymodel == PSY_NSPSYTUNE) {
if (gfp->VBR == vbr_off || gfp->VBR == vbr_abr) {
static FLOAT fircoef[9] = {
-0.0207887 * 5, -0.0378413 * 5, -0.0432472 * 5, -0.031183 * 5,
7.79609e-18 * 5, 0.0467745 * 5, 0.10091 * 5, 0.151365 * 5,
0.187098 * 5
};
int i;
FLOAT f;
for (i = 0; i < 18; i++)
gfc->nsPsy.pefirbuf[i] = gfc->nsPsy.pefirbuf[i + 1];
f = 0.0;
for (gr = 0; gr < gfc->mode_gr; gr++)
for (ch = 0; ch < gfc->channels_out; ch++)
f += (*pe_use)[gr][ch];
gfc->nsPsy.pefirbuf[18] = f;
f = gfc->nsPsy.pefirbuf[9];
for (i = 0; i < 9; i++)
f += (gfc->nsPsy.pefirbuf[i] + gfc->nsPsy.pefirbuf[18 - i]) * fircoef[i];
f = (670 * 5 * gfc->mode_gr * gfc->channels_out) / f;
for (gr = 0; gr < gfc->mode_gr; gr++) {
for (ch = 0; ch < gfc->channels_out; ch++) {
(*pe_use)[gr][ch] *= f;
}
}
}
}
switch (gfp->VBR) {
default:
case vbr_off:
CBR_iteration_loop(gfp, *pe_use, ms_ener_ratio, *masking);
break;
case vbr_mt:
case vbr_rh:
case vbr_mtrh:
VBR_iteration_loop(gfp, *pe_use, ms_ener_ratio, *masking);
break;
case vbr_abr:
ABR_iteration_loop(gfp, *pe_use, ms_ener_ratio, *masking);
break;
}
/****************************************
* Stage 5: bitstream formatting *
****************************************/
/* write the frame to the bitstream */
format_bitstream(gfp);
/* copy mp3 bit buffer into array */
mp3count = copy_buffer(gfc, mp3buf, mp3buf_size, 1);
if (gfp->bWriteVbrTag)
AddVbrFrame(gfp);
#if defined(HAVE_GTK)
if (gfp->analysis && gfc->pinfo != NULL) {
for (ch = 0; ch < gfc->channels_out; ch++) {
int j;
for (j = 0; j < FFTOFFSET; j++)
gfc->pinfo->pcmdata[ch][j] = gfc->pinfo->pcmdata[ch][j + gfp->framesize];
for (j = FFTOFFSET; j < 1600; j++) {
gfc->pinfo->pcmdata[ch][j] = inbuf[ch][j - FFTOFFSET];
}
}
set_frame_pinfo(gfp, *masking);
}
#endif
#ifdef BRHIST
updateStats(gfc);
#endif
return mp3count;
}
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