[FFmpeg-devel] [RFC] AAC encoder optimizations
Kostya
kostya.shishkov
Sat Sep 13 09:29:27 CEST 2008
Here's the next experimental version.
I've replaced quantization and codebook determining functions
and reworked scalefactor search code.
Now it's ~30 time slower than realtime on Core2.
Please comment on those functions and give your suggestions.
-------------- next part --------------
/*
* AAC encoder
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file aacenc.c
* AAC encoder
*/
/***********************************
* TODOs:
* speedup quantizer selection
* add sane pulse detection
* add temporal noise shaping
***********************************/
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include "aac.h"
#include "aactab.h"
#include "psymodel.h"
static const uint8_t swb_size_1024_96[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
};
static const uint8_t swb_size_1024_64[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
};
static const uint8_t swb_size_1024_48[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
96
};
static const uint8_t swb_size_1024_32[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
};
static const uint8_t swb_size_1024_24[] = {
4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
};
static const uint8_t swb_size_1024_16[] = {
8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
};
static const uint8_t swb_size_1024_8[] = {
12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
};
static const uint8_t *swb_size_1024[] = {
swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
};
static const uint8_t swb_size_128_96[] = {
4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
};
static const uint8_t swb_size_128_48[] = {
4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
};
static const uint8_t swb_size_128_24[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
};
static const uint8_t swb_size_128_16[] = {
4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
};
static const uint8_t swb_size_128_8[] = {
4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
};
static const uint8_t *swb_size_128[] = {
/* the last entry on the following row is swb_size_128_64 but is a
duplicate of swb_size_128_96 */
swb_size_128_96, swb_size_128_96, swb_size_128_96,
swb_size_128_48, swb_size_128_48, swb_size_128_48,
swb_size_128_24, swb_size_128_24, swb_size_128_16,
swb_size_128_16, swb_size_128_16, swb_size_128_8
};
/** spectral coefficients codebook information */
static const struct {
int16_t maxval; ///< maximum possible value
int8_t range; ///< value used in vector calculation
} aac_cb_info[] = {
{ 0, -1 }, // zero codebook
{ 1, 3 },
{ 1, 3 },
{ 2, 3 },
{ 2, 3 },
{ 4, 9 },
{ 4, 9 },
{ 7, 8 },
{ 7, 8 },
{ 12, 13 },
{ 12, 13 },
{ 8191, 17 },
{ -1, -1 }, // reserved
{ -1, -1 }, // perceptual noise substitution
{ -1, -1 }, // intensity out-of-phase
{ -1, -1 }, // intensity in-phase
};
/** bits needed to code codebook run value for long windows */
static const uint8_t run_value_bits_long[64] = {
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
};
/** bits needed to code codebook run value for short windows */
static const uint8_t run_value_bits_short[16] = {
3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
};
static const uint8_t* run_value_bits[2] = {
run_value_bits_long, run_value_bits_short
};
/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
{1, TYPE_SCE}, // 1 channel - single channel element
{1, TYPE_CPE}, // 2 channels - channel pair
{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
};
/**
* AAC encoder context
*/
typedef struct {
PutBitContext pb;
MDCTContext mdct1024; ///< long (1024 samples) frame transform context
MDCTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
int16_t* samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
int cur_channel;
int last_frame;
float lambda;
} AACEncContext;
/**
* Make AAC audio config object.
* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
*/
static void put_audio_specific_config(AVCodecContext *avctx)
{
PutBitContext pb;
AACEncContext *s = avctx->priv_data;
init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
put_bits(&pb, 5, 2); //object type - AAC-LC
put_bits(&pb, 4, s->samplerate_index); //sample rate index
put_bits(&pb, 4, avctx->channels);
//GASpecificConfig
put_bits(&pb, 1, 0); //frame length - 1024 samples
put_bits(&pb, 1, 0); //does not depend on core coder
put_bits(&pb, 1, 0); //is not extension
flush_put_bits(&pb);
}
static av_cold int aac_encode_init(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
int i;
const uint8_t *sizes[2];
int lengths[2];
avctx->frame_size = 1024;
for(i = 0; i < 16; i++)
if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
break;
if(i == 16){
av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
return -1;
}
if(avctx->channels > 6){
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
return -1;
}
s->samplerate_index = i;
dsputil_init(&s->dsp, avctx);
ff_mdct_init(&s->mdct1024, 11, 0);
ff_mdct_init(&s->mdct128, 8, 0);
// window init
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
ff_sine_window_init(ff_sine_1024, 1024);
ff_sine_window_init(ff_sine_128, 128);
s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
sizes[0] = swb_size_1024[i];
sizes[1] = swb_size_128[i];
lengths[0] = ff_aac_num_swb_1024[i];
lengths[1] = ff_aac_num_swb_128[i];
ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
s->psypp = ff_psy_preprocess_init(avctx);
#ifndef CONFIG_HARDCODED_TABLES
for (i = 0; i < 316; i++)
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
#endif /* CONFIG_HARDCODED_TABLES */
return 0;
}
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
SingleChannelElement *sce, short *audio, int channel)
{
int i, j, k;
const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
memcpy(s->output, sce->saved, sizeof(float)*1024);
if(sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE){
memset(s->output, 0, sizeof(s->output[0]) * 448);
for(i = 448; i < 576; i++)
s->output[i] = sce->saved[i] * pwindow[i - 448];
for(i = 576; i < 704; i++)
s->output[i] = sce->saved[i];
}
if(sce->ics.window_sequence[0] != LONG_START_SEQUENCE){
j = channel;
for (i = 0; i < 1024; i++, j += avctx->channels){
s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
sce->saved[i] = audio[j] * lwindow[i];
}
}else{
j = channel;
for(i = 0; i < 448; i++, j += avctx->channels)
s->output[i+1024] = audio[j];
for(i = 448; i < 576; i++, j += avctx->channels)
s->output[i+1024] = audio[j] * swindow[576 - i - 1];
memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
j = channel;
for(i = 0; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
}else{
j = channel;
for (k = 0; k < 1024; k += 128) {
for(i = 448 + k; i < 448 + k + 256; i++)
s->output[i - 448 - k] = (i < 1024)
? sce->saved[i]
: audio[channel + (i-1024)*avctx->channels];
s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
}
j = channel;
for(i = 0; i < 1024; i++, j += avctx->channels)
sce->saved[i] = audio[j];
}
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
{
int w;
put_bits(&s->pb, 1, 0); // ics_reserved bit
put_bits(&s->pb, 2, info->window_sequence[0]);
put_bits(&s->pb, 1, info->use_kb_window[0]);
if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
put_bits(&s->pb, 6, info->max_sfb);
put_bits(&s->pb, 1, 0); // no prediction
}else{
put_bits(&s->pb, 4, info->max_sfb);
for(w = 1; w < 8; w++){
put_bits(&s->pb, 1, !info->group_len[w]);
}
}
}
/**
* Encode MS data.
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
*/
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
{
int i, w;
put_bits(pb, 2, cpe->ms_mode);
if(cpe->ms_mode == 1){
for(w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]){
for(i = 0; i < cpe->ch[0].ics.max_sfb; i++)
put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
}
}
}
/**
* Quantize one coefficient.
* @return absolute value of the quantized coefficient
* @see 3GPP TS26.403 5.6.2 "Scalefactor determination"
*/
static av_always_inline int quant(float coef, const float Q)
{
return av_clip((int)(pow(fabsf(coef) * Q, 0.75) + 0.4054), 0, 8191);
}
/**
* Calculate rate distortion cost for quantizing with given codebook
*
* @return quantization distortion
*/
static float quantize_band_cost(const float *in, int size, int scale_idx, int cb,
const float lambda, const float uplim)
{
const float Q = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
int i, j, k;
float cost = 0;
const int dim = cb < FIRST_PAIR_BT ? 4 : 2;
START_TIMER
if(!cb){
for(i = 0; i < size; i++)
cost += in[i]*in[i]*lambda;
return cost;
}
for(i = 0; i < size; i += dim){
float mincost = INFINITY;
int minidx = 0;
int minbits = 0;
const float *vec = ff_aac_codebook_vectors[cb-1];
for(j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim){
float rd = 0.0f;
int curbits = ff_aac_spectral_bits[cb-1][minidx];
if(IS_CODEBOOK_UNSIGNED(cb)){
for(k = 0; k < dim; k++){
float t = fabsf(in[i+k]);
float di;
//do not code with escape sequence small values
if(vec[k] == 64.0f && t < 39.0f*Q){
rd = INFINITY;
break;
}
if(vec[k] == 64.0f){//FIXME: slow
if(t >= 165140.0f*Q){ // clipped value
di = t - 165140.0f;
curbits += 21;
}else{
int c = quant(t, 1.0/Q);
di = t - c*cbrt(c)*Q;
curbits += av_log2(c)*2 - 4 + 1;
}
}else{
di = t - vec[k]*Q;
}
if(vec[k] != 0.0f)
curbits++;
rd += di*di*lambda;
}
}else{
for(k = 0; k < dim; k++){
float di = in[i+k] - vec[k]*Q;
rd += di*di*lambda;
}
}
rd += curbits;
if(rd < mincost){
mincost = rd;
minidx = j;
minbits = curbits;
}
}
cost += mincost;
if(cost >= uplim)
return uplim;
minbits = 0;
if(IS_CODEBOOK_UNSIGNED(cb))
for(j = 0; j < dim; j++)
if(ff_aac_codebook_vectors[cb-1][minidx*2+j] != 0.0f)
minbits++;
cost += minbits;
}
STOP_TIMER("quantize_band")
return cost;
}
/**
* Prepare coefficients for encoding.
*
* @return sum of coefficient absolute values
*/
static void quantize_and_encode_band(PutBitContext *pb, const float *in, int size,
int scale_idx, int cb, const float lambda)
{
const float Q = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
const float IQ = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
int i, j, k;
const int dim = cb < FIRST_PAIR_BT ? 4 : 2;
if(!cb)
return;
for(i = 0; i < size; i += dim){
float mincost = INFINITY;
int minidx = 0;
int minbits = 0;
const float *vec = ff_aac_codebook_vectors[cb-1];
for(j = 0; j < ff_aac_spectral_sizes[cb-1]; j++, vec += dim){
float rd = 0.0f;
int curbits = ff_aac_spectral_bits[cb-1][minidx];
if(IS_CODEBOOK_UNSIGNED(cb)){
for(k = 0; k < dim; k++){
float t = fabsf(in[i+k]);
float di;
if(vec[k] != 0.0f)
curbits++;
//do not code with escape sequence small values
if(vec[k] == 64.0f && t < 39.0f*Q){
rd = INFINITY;
break;
}
if(vec[k] == 64.0f){//FIXME: slow
if(t*IQ >= 165140.0f){ // clipped value
di = t - 165140.0f;
curbits += 21;
}else{
int c = quant(t, IQ);
di = t - c*cbrt(c)*Q;
curbits += av_log2(c)*2 - 4 + 1;
}
}else{
di = t - vec[k]*Q;
}
rd += di*di*lambda;
}
}else{
for(k = 0; k < dim; k++){
float di = in[i+k] - vec[k]*Q;
rd += di*di*lambda;
}
}
rd += curbits;
if(rd < mincost){
mincost = rd;
minidx = j;
minbits = curbits;
}
}
put_bits(pb, ff_aac_spectral_bits[cb-1][minidx], ff_aac_spectral_codes[cb-1][minidx]);
if(IS_CODEBOOK_UNSIGNED(cb))
for(j = 0; j < dim; j++)
if(ff_aac_codebook_vectors[cb-1][minidx*dim+j] != 0.0f)
put_bits(pb, 1, in[i+j] < 0.0f);
if(cb == ESC_BT){
for(j = 0; j < 2; j++){
if(ff_aac_codebook_vectors[cb-1][minidx*2+j] == 64.0f){
int coef = quant(in[i+j], IQ);
int len = av_log2(coef);
put_bits(pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
put_bits(pb, len, coef & ((1 << len) - 1));
}
}
}
}
}
/**
* structure used in optimal codebook search
*/
typedef struct BandCodingPath {
int prev_idx; ///< pointer to the previous path point
int codebook; ///< codebook for coding band run
float cost; ///< path cost
int run;
} BandCodingPath;
/**
* Encode band info for single window group bands.
*/
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda)
{
BandCodingPath path[120][12];
int w, swb, cb, start, start2, size;
int i, j;
const int max_sfb = sce->ics.max_sfb;
const int run_bits = sce->ics.num_windows == 1 ? 5 : 3;
const int run_esc = (1 << run_bits) - 1;
int idx, ppos, count;
int stackrun[120], stackcb[120], stack_len;
start = win*128;
for(cb = 0; cb < 12; cb++){
path[0][cb].cost = 0.0f;
path[0][cb].prev_idx = -1;
path[0][cb].run = 0;
}
for(swb = 0; swb < max_sfb; swb++){
start2 = start;
size = sce->ics.swb_sizes[swb];
if(sce->zeroes[win*16 + swb]){
for(cb = 0; cb < 12; cb++){
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost;
path[swb+1][cb].run = path[swb][cb].run + 1;
}
}else{
float minrd = INFINITY;
int mincb = 0;
for(cb = 0; cb < 12; cb++){
float rd = 0.0f;
for(w = 0; w < group_len; w++){
rd += quantize_band_cost(sce->coeffs + start + w*128, size, sce->sf_idx[(win+w)*16+swb], cb, lambda, minrd);
}
path[swb+1][cb].prev_idx = cb;
path[swb+1][cb].cost = path[swb][cb].cost + rd;
path[swb+1][cb].run = path[swb][cb].run + 1;
if(rd < minrd){
minrd = rd;
mincb = cb;
}
}
for(cb = 0; cb < 12; cb++){
float cost = path[swb][cb].cost + minrd + run_bits + 4;
if(cost < path[swb+1][cb].cost){
path[swb+1][cb].prev_idx = mincb;
path[swb+1][cb].cost = cost;
path[swb+1][cb].run = 1;
}
}
}
start += sce->ics.swb_sizes[swb];
}
//convert resulting path from backward-linked list
stack_len = 0;
idx = 0;
for(cb = 1; cb < 12; cb++){
if(path[max_sfb][cb].cost < path[max_sfb][idx].cost)
idx = cb;
}
ppos = max_sfb;
while(ppos > 0){
cb = idx;
stackrun[stack_len] = path[ppos][cb].run;
stackcb [stack_len] = cb;
idx = path[ppos][cb].prev_idx;
ppos -= path[ppos][cb].run;
stack_len++;
}
//perform actual band info encoding
start = 0;
for(i = stack_len - 1; i >= 0; i--){
put_bits(&s->pb, 4, stackcb[i]);
count = stackrun[i];
memset(sce->zeroes + win*16 + start, !stackcb[i], count);
//XXX: memset when band_type is also uint8_t
for(j = 0; j < count; j++){
sce->band_type[win*16 + start] = stackcb[i];
start++;
}
while(count >= run_esc){
put_bits(&s->pb, run_bits, run_esc);
count -= run_esc;
}
put_bits(&s->pb, run_bits, count);
}
}
/**
* Produce integer coefficients from scalefactors provided by the model.
*/
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
{
int i, w, w2, g, ch;
int start, sum, maxsfb, cmaxsfb;
for(ch = 0; ch < chans; ch++){
IndividualChannelStream *ics = &cpe->ch[ch].ics;
start = 0;
maxsfb = 0;
cpe->ch[ch].pulse.num_pulse = 0;
for(w = 0; w < ics->num_windows*16; w += 16){
for(g = 0; g < ics->num_swb; g++){
sum = 0;
//apply M/S
if(!ch && cpe->ms_mask[w + g]){
for(i = 0; i < ics->swb_sizes[g]; i++){
cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
}
}
start += ics->swb_sizes[g];
}
for(cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--);
maxsfb = FFMAX(maxsfb, cmaxsfb);
}
ics->max_sfb = maxsfb;
//adjust zero bands for window groups
for(w = 0; w < ics->num_windows; w += ics->group_len[w]){
for(g = 0; g < ics->max_sfb; g++){
i = 1;
for(w2 = w; w2 < w + ics->group_len[w]; w2++){
if(!cpe->ch[ch].zeroes[w2*16 + g]){
i = 0;
break;
}
}
cpe->ch[ch].zeroes[w*16 + g] = i;
}
}
}
if(chans > 1 && cpe->common_window){
IndividualChannelStream *ics0 = &cpe->ch[0].ics;
IndividualChannelStream *ics1 = &cpe->ch[1].ics;
int msc = 0;
ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
ics1->max_sfb = ics0->max_sfb;
for(w = 0; w < ics0->num_windows*16; w += 16)
for(i = 0; i < ics0->max_sfb; i++)
if(cpe->ms_mask[w+i]) msc++;
if(msc == 0 || ics0->max_sfb == 0) cpe->ms_mode = 0;
else cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
}
}
typedef struct TrellisPath {
float cost;
int prev;
int min_val;
int max_val;
} TrellisPath;
static void search_for_quantizers_anmr(AACEncContext *s, SingleChannelElement *sce, const float lambda)
{
int q, w, w2, g, start = 0;
int i;
int idx;
TrellisPath paths[256*121];
int bandaddr[121];
int minq;
float mincost;
START_TIMER
for(i = 0; i < 256; i++){
paths[i].cost = 0.0f;
paths[i].prev = -1;
paths[i].min_val = i;
paths[i].max_val = i;
}
for(i = 256; i < 256*121; i++){
paths[i].cost = INFINITY;
paths[i].prev = -2;
paths[i].min_val = INT_MAX;
paths[i].max_val = 0;
}
idx = 256;
for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
start = w*128;
for(g = 0; g < sce->ics.num_swb; g++){
const float *coefs = sce->coeffs + start;
float qmin, qmax;
int nz = 0;
bandaddr[idx >> 8] = w*16+g;
qmin = INT_MAX;
qmax = 0.0f;
for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
if(band->energy <= band->threshold || band->threshold == 0.0f){
sce->zeroes[(w+w2)*16+g] = 1;
continue;
}
sce->zeroes[(w+w2)*16+g] = 0;
nz = 1;
for(i = 0; i < sce->ics.swb_sizes[g]; i++){
float t = fabsf(coefs[w2*128+i]);
if(t > 0.0f) qmin = fminf(qmin, t);
qmax = fmaxf(qmax, t);
}
}
if(nz){
int minscale, maxscale;
//minimum scalefactor index is when minimum nonzero coefficient after quantizing is not clipped
minscale = av_clip_uint8(log2(qmin)*4 - 69 + SCALE_ONE_POS - SCALE_DIV_512);
//maximum scalefactor index is when maximum coefficient after quantizing is still not zero
maxscale = av_clip_uint8(log2(qmax)*4 + 6 + SCALE_ONE_POS - SCALE_DIV_512);
for(q = minscale; q < maxscale; q++){
float dist = 0.0f;
for(w2 = 0; w2 < sce->ics.group_len[w]; w2++){
FFPsyBand *band = &s->psy.psy_bands[s->cur_channel*PSY_MAX_BANDS+(w+w2)*16+g];
int cb;
float mincost = INFINITY;
if(sce->zeroes[(w+w2)*16+g])
continue;
for(cb = 0; cb <= ESC_BT; cb++){
mincost = fminf(mincost, quantize_band_cost(coefs + w2*128, sce->ics.swb_sizes[g],
q, cb, s->lambda / band->threshold, mincost));
}
dist += mincost;
}
for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
float cost;
int minv, maxv;
if(isinf(paths[idx - 256 + i].cost))
continue;
cost = paths[idx - 256 + i].cost + dist
+ ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 256 + i].min_val, q);
maxv = FFMAX(paths[idx - 256 + i].max_val, q);
if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
paths[idx + q].cost = cost;
paths[idx + q].prev = idx - 256 + i;
paths[idx + q].min_val = minv;
paths[idx + q].max_val = maxv;
}
}
}
}else{
for(q = 0; q < 256; q++){
if(!isinf(paths[idx - 256 + q].cost)){
paths[idx + q].cost = paths[idx - 256 + q].cost + 1;
paths[idx + q].prev = idx - 256 + q;
paths[idx + q].min_val = FFMIN(paths[idx - 256 + q].min_val, q);
paths[idx + q].max_val = FFMAX(paths[idx - 256 + q].max_val, q);
continue;
}
for(i = FFMAX(q - SCALE_MAX_DIFF, 0); i < FFMIN(q + SCALE_MAX_DIFF, 256); i++){
float cost;
int minv, maxv;
if(isinf(paths[idx - 256 + i].cost))
continue;
cost = paths[idx - 256 + i].cost + ff_aac_scalefactor_bits[q - i + SCALE_DIFF_ZERO];
minv = FFMIN(paths[idx - 256 + i].min_val, q);
maxv = FFMAX(paths[idx - 256 + i].max_val, q);
if(cost < paths[idx + q].cost && maxv-minv < SCALE_MAX_DIFF){
paths[idx + q].cost = cost;
paths[idx + q].prev = idx - 256 + i;
paths[idx + q].min_val = minv;
paths[idx + q].max_val = maxv;
}
}
}
}
sce->zeroes[w*16+g] = !nz;
start += sce->ics.swb_sizes[g];
idx += 256;
}
}
idx -= 256;
mincost = paths[idx].cost;
minq = idx;
for(i = 1; i < 256; i++){
if(paths[idx + i].cost < mincost){
mincost = paths[idx + i].cost;
minq = idx + i;
}
}
while(minq >= 256){
sce->sf_idx[bandaddr[minq>>8]] = minq & 0xFF;
minq = paths[minq].prev;
}
//set the same quantizers inside window groups
for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
for(g = 0; g < sce->ics.num_swb; g++)
for(w2 = 1; w2 < sce->ics.group_len[w]; w2++)
sce->sf_idx[(w+w2)*16+g] = sce->sf_idx[w*16+g];
STOP_TIMER("search_for_quantizers_anmr")
}
static void search_for_ms(AACEncContext *s, ChannelElement *cpe, const float lambda)
{
int start = 0, i, w, w2, g;
float M[128], S[128];
SingleChannelElement *sce0 = &cpe->ch[0];
SingleChannelElement *sce1 = &cpe->ch[1];
if(!cpe->common_window)
return;
for(w = 0; w < sce0->ics.num_windows; w += sce0->ics.group_len[w]){
for(g = 0; g < sce0->ics.num_swb; g++){
if(!cpe->ch[0].zeroes[w*16+g] && !cpe->ch[1].zeroes[w*16+g]){
float dist1 = 0.0f, dist2 = 0.0f;
for(w2 = 0; w2 < sce0->ics.group_len[w]; w2++){
FFPsyBand *band0 = &s->psy.psy_bands[(s->cur_channel+0)*PSY_MAX_BANDS+(w+w2)*16+g];
FFPsyBand *band1 = &s->psy.psy_bands[(s->cur_channel+1)*PSY_MAX_BANDS+(w+w2)*16+g];
float minthr = fminf(band0->threshold, band1->threshold);
float maxthr = fmaxf(band0->threshold, band1->threshold);
for(i = 0; i < sce0->ics.swb_sizes[g]; i++){
M[i] = (sce0->coeffs[start+w2*128+i]
+ sce1->coeffs[start+w2*128+i])*0.5;
S[i] = sce0->coeffs[start+w2*128+i]
- sce1->coeffs[start+w2*128+i];
}
dist1 += quantize_band_cost(sce0->coeffs + start + w2*128,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / band0->threshold, INFINITY);
dist1 += quantize_band_cost(sce1->coeffs + start + w2*128,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / band1->threshold, INFINITY);
dist2 += quantize_band_cost(M,
sce0->ics.swb_sizes[g],
sce0->sf_idx[(w+w2)*16+g],
sce0->band_type[(w+w2)*16+g],
lambda / maxthr, INFINITY);
dist2 += quantize_band_cost(S,
sce1->ics.swb_sizes[g],
sce1->sf_idx[(w+w2)*16+g],
sce1->band_type[(w+w2)*16+g],
lambda / minthr, INFINITY);
}
cpe->ms_mask[w*16+g] = dist2 < dist1;
}
start += sce0->ics.swb_sizes[g];
}
}
}
/**
* Encode scalefactor band coding type.
*/
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
{
int w;
for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
}
}
/**
* Encode scalefactors.
*/
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
{
int off = sce->sf_idx[0], diff;
int i, w;
for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
for(i = 0; i < sce->ics.max_sfb; i++){
if(!sce->zeroes[w*16 + i]){
diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
if(diff < 0 || diff > 120) av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
off = sce->sf_idx[w*16 + i];
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
}
}
}
}
/**
* Encode pulse data.
*/
static void encode_pulses(AACEncContext *s, Pulse *pulse)
{
int i;
put_bits(&s->pb, 1, !!pulse->num_pulse);
if(!pulse->num_pulse) return;
put_bits(&s->pb, 2, pulse->num_pulse - 1);
put_bits(&s->pb, 6, pulse->start);
for(i = 0; i < pulse->num_pulse; i++){
put_bits(&s->pb, 5, pulse->pos[i]);
put_bits(&s->pb, 4, pulse->amp[i]);
}
}
/**
* Encode spectral coefficients processed by psychoacoustic model.
*/
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
{
int start, i, w, w2;
for(w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]){
start = 0;
for(i = 0; i < sce->ics.max_sfb; i++){
if(sce->zeroes[w*16 + i]){
start += sce->ics.swb_sizes[i];
continue;
}
for(w2 = w; w2 < w + sce->ics.group_len[w]; w2++){
quantize_and_encode_band(&s->pb, sce->coeffs + start + w2*128,
sce->ics.swb_sizes[i],
sce->sf_idx[w*16 + i],
sce->band_type[w*16 + i],
s->lambda);
}
start += sce->ics.swb_sizes[i];
}
}
}
/**
* Encode one channel of audio data.
*/
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
{
put_bits(&s->pb, 8, sce->sf_idx[0]);
if(!common_window) put_ics_info(s, &sce->ics);
encode_band_info(s, sce);
encode_scale_factors(avctx, s, sce);
encode_pulses(s, &sce->pulse);
put_bits(&s->pb, 1, 0); //tns
put_bits(&s->pb, 1, 0); //ssr
encode_spectral_coeffs(s, sce);
return 0;
}
/**
* Write some auxiliary information about the created AAC file.
*/
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
{
int i, namelen, padbits;
namelen = strlen(name) + 2;
put_bits(&s->pb, 3, TYPE_FIL);
put_bits(&s->pb, 4, FFMIN(namelen, 15));
if(namelen >= 15)
put_bits(&s->pb, 8, namelen - 16);
put_bits(&s->pb, 4, 0); //extension type - filler
padbits = 8 - (put_bits_count(&s->pb) & 7);
align_put_bits(&s->pb);
for(i = 0; i < namelen - 2; i++)
put_bits(&s->pb, 8, name[i]);
put_bits(&s->pb, 12 - padbits, 0);
}
static int aac_encode_frame(AVCodecContext *avctx,
uint8_t *frame, int buf_size, void *data)
{
AACEncContext *s = avctx->priv_data;
int16_t *samples = s->samples, *samples2, *la;
ChannelElement *cpe;
int i, j, chans, tag, start_ch;
const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
int chan_el_counter[4];
if(s->last_frame)
return 0;
if(data){
if(!s->psypp){
memcpy(s->samples + 1024 * avctx->channels, data, 1024 * avctx->channels * sizeof(s->samples[0]));
}else{
start_ch = 0;
samples2 = s->samples + 1024 * avctx->channels;
for(i = 0; i < chan_map[0]; i++){
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, samples2 + start_ch, start_ch + i, chans);
start_ch += chans;
}
}
}
if(!avctx->frame_number){
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
return 0;
}
init_put_bits(&s->pb, frame, buf_size*8);
if((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)){
put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
}
start_ch = 0;
memset(chan_el_counter, 0, sizeof(chan_el_counter));
for(i = 0; i < chan_map[0]; i++){
FFPsyWindowInfo wi[2];
tag = chan_map[i+1];
chans = tag == TYPE_CPE ? 2 : 1;
cpe = &s->cpe[i];
samples2 = samples + start_ch;
la = samples2 + 1024 * avctx->channels + start_ch;
if(!data) la = NULL;
s->lambda = 5e-7f;
for(j = 0; j < chans; j++){
IndividualChannelStream *ics = &cpe->ch[j].ics;
int k;
wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
ics->window_sequence[1] = ics->window_sequence[0];
ics->window_sequence[0] = wi[j].window_type[0];
ics->use_kb_window[1] = ics->use_kb_window[0];
ics->use_kb_window[0] = wi[j].window_shape;
ics->num_windows = wi[j].num_windows;
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
for(k = 0; k < ics->num_windows; k++)
ics->group_len[k] = wi[j].grouping[k];
s->cur_channel = start_ch + j;
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
search_for_quantizers_anmr(s, &cpe->ch[j], s->lambda);
}
cpe->common_window = 0;
if(chans > 1
&& wi[0].window_type[0] == wi[1].window_type[0]
&& wi[0].window_shape == wi[1].window_shape){
cpe->common_window = 1;
for(j = 0; j < wi[0].num_windows; j++){
if(wi[0].grouping[j] != wi[1].grouping[j]){
cpe->common_window = 0;
break;
}
}
}
search_for_ms(s, cpe, s->lambda);
adjust_frame_information(s, cpe, chans);
put_bits(&s->pb, 3, tag);
put_bits(&s->pb, 4, chan_el_counter[tag]++);
if(chans == 2){
put_bits(&s->pb, 1, cpe->common_window);
if(cpe->common_window){
put_ics_info(s, &cpe->ch[0].ics);
encode_ms_info(&s->pb, cpe);
}
}
for(j = 0; j < chans; j++){
s->cur_channel = start_ch + j;
ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
}
start_ch += chans;
}
put_bits(&s->pb, 3, TYPE_END);
flush_put_bits(&s->pb);
avctx->frame_bits = put_bits_count(&s->pb);
if(!data)
s->last_frame = 1;
memcpy(s->samples, s->samples + 1024 * avctx->channels, 1024 * avctx->channels * sizeof(s->samples[0]));
return put_bits_count(&s->pb)>>3;
}
static av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_psy_end(&s->psy);
ff_psy_preprocess_end(s->psypp);
av_freep(&s->samples);
av_freep(&s->cpe);
return 0;
}
AVCodec aac_encoder = {
"aac",
CODEC_TYPE_AUDIO,
CODEC_ID_AAC,
sizeof(AACEncContext),
aac_encode_init,
aac_encode_frame,
aac_encode_end,
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
};
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