[FFmpeg-devel] [RFC] AAC Encoder, now more optimal
Kostya
kostya.shishkov
Fri Sep 5 15:13:58 CEST 2008
After some time (I'd like to have more free time to spend on it though),
I want to expose my new AAC encoder.
It is slower than realtime since it uses search for optimal quantizers
for given quality.
Known weak points:
* no bitrate management (it uses fixed quality for now)
* no M/S detection
* quantization is not optimal yet
I haven't implemented those because it will slow encoder even more, so
debug session will include an hour rest (and in case of M/S detection
I don't know how to implement it) - 16sec sample coding takes 42 secs
on my PPC G4-1.42GHz and 39 secs on Core2 1.8MHz.
Any comments, suggestions, speedup tricks are extremely welcomed
(especially the latter).
-------------- 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
};
/**
* 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
int bits; ///< number of bit needed to code given number of bands
} BandCodingPath;
/**
* 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;
} 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);
}
static inline float get_approximate_quant_error(const float *q, const int *c, int size, int scale_idx)
{
int i;
float coef, unquant, sum = 0.0f;
const float IQ = ff_aac_pow2sf_tab[200 + scale_idx - SCALE_ONE_POS + SCALE_DIV_512];
for(i = 0; i < size; i++){
coef = fabsf(q[i]);
unquant = (c[i] * cbrt(c[i])) * IQ;
sum += (coef - unquant) * (coef - unquant);
}
return sum;
}
/**
* Convert coefficients to integers.
* @fixme make it RD-optimal
* @return sum of coefficient absolute values
*/
static inline int quantize_band(const float *in, int *out, int size, int scale_idx)
{
int i, sign, sum = 0;
const float Q = ff_aac_pow2sf_tab[200 - scale_idx + SCALE_ONE_POS - SCALE_DIV_512];
for(i = 0; i < size; i++){
sign = in[i] > 0.0;
out[i] = quant(in[i], Q);
sum += out[i];
if(sign) out[i] = -out[i];
}
return sum;
}
static inline int get_approximate_bits(const int *in, int size)
{
int i, bits = 0;
for(i = 0; i < size; i += 2){
int j, idx = 0;
for(j = 0; j < 2; j++){
int t = FFABS(in[i+j]);
if(t)
bits++;
if(t > 16)
bits += av_log2(t)*2 + 4 - 1;
idx = idx*17 + FFMIN(t, 16);
}
bits += ff_aac_spectral_bits[ESC_BT-1][idx];
}
return bits;
}
/**
* Calculate the number of bits needed to code all coefficient signs in current band.
*/
static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
int group_len, int start, int size)
{
int bits = 0;
int i, w;
for(w = 0; w < group_len; w++){
for(i = 0; i < size; i++){
if(sce->icoefs[start + i])
bits++;
}
start += 128;
}
return bits;
}
/**
* Calculate the number of bits needed to code given band with given codebook.
*
* @param s encoder context
* @param sce channel element
* @param group_len window group length
* @param start scalefactor band position in spectral coefficients
* @param size scalefactor band size
* @param cb codebook number
*/
static int calculate_band_bits(AACEncContext *s, SingleChannelElement *sce,
int group_len, int start, int size, int cb)
{
int i, j, w;
int bits = 0, dim, idx;
int range = aac_cb_info[cb].range;
if(range == -1) return 0;
cb--;
dim = cb < FIRST_PAIR_BT ? 4 : 2;
if(cb == ESC_BT){
for(w = 0; w < group_len; w++){
int coef_abs[2];
for(i = 0; i < size; i += 2){
idx = 0;
for(j = 0; j < 2; j++){
coef_abs[j] = FFABS(sce->icoefs[start+i+j]);
idx = idx*17 + FFMIN(coef_abs[j], 16);
if(coef_abs[j] > 15){
bits += av_log2(coef_abs[j])*2 - 4 + 1;
}
}
bits += ff_aac_spectral_bits[cb][idx];
}
start += 128;
}
}else if(IS_CODEBOOK_UNSIGNED(cb)){
for(w = 0; w < group_len; w++){
for(i = 0; i < size; i += dim){
idx = FFABS(sce->icoefs[start+i]);
for(j = 1; j < dim; j++){
idx = idx * range + FFABS(sce->icoefs[start+i+j]);
}
bits += ff_aac_spectral_bits[cb][idx];
}
start += 128;
}
}else{
for(w = 0; w < group_len; w++){
for(i = 0; i < size; i += dim){
idx = sce->icoefs[start+i];
for(j = 1; j < dim; j++)
idx = idx * range + sce->icoefs[start+i+j];
//it turned out that all signed codebooks use the same offset for index coding
idx += 40;
bits += ff_aac_spectral_bits[cb][idx];
}
start += 128;
}
}
return bits;
}
/**
* Encode band info for single window group bands.
*/
static void encode_window_bands_info(AACEncContext *s, SingleChannelElement *sce,
int win, int group_len)
{
BandCodingPath path[64];
int band_bits[64][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 bits, sbits, idx, count;
int stack[64], stack_len;
start = win*128;
for(swb = 0; swb < max_sfb; swb++){
int maxval = 0;
start2 = start;
size = sce->ics.swb_sizes[swb];
if(sce->zeroes[win*16 + swb]){
maxval = 0;
}else{
for(w = 0; w < group_len; w++){
for(i = start2; i < start2 + size; i++){
maxval = FFMAX(maxval, FFABS(sce->icoefs[i]));
}
start2 += 128;
}
}
sbits = calculate_band_sign_bits(s, sce, group_len, start, size);
for(cb = 0; cb < 12; cb++){
if(aac_cb_info[cb].maxval < maxval){
band_bits[swb][cb] = INT_MAX;
}else{
band_bits[swb][cb] = calculate_band_bits(s, sce, group_len, start, size, cb);
if(IS_CODEBOOK_UNSIGNED(cb-1)){
band_bits[swb][cb] += sbits;
}
}
}
start += sce->ics.swb_sizes[swb];
}
path[0].bits = 0;
for(i = 1; i <= max_sfb; i++)
path[i].bits = INT_MAX;
for(i = 0; i < max_sfb; i++){
for(cb = 0; cb < 12; cb++){
int sum = 0;
for(j = 1; j <= max_sfb - i; j++){
if(band_bits[i+j-1][cb] == INT_MAX)
break;
sum += band_bits[i+j-1][cb];
bits = sum + path[i].bits + run_value_bits[sce->ics.num_windows == 8][j];
if(bits < path[i+j].bits){
path[i+j].bits = bits;
path[i+j].codebook = cb;
path[i+j].prev_idx = i;
}
}
}
}
assert(path[max_sfb].bits != INT_MAX);
//convert resulting path from backward-linked list
stack_len = 0;
idx = max_sfb;
while(idx > 0){
stack[stack_len++] = idx;
idx = path[idx].prev_idx;
}
//perform actual band info encoding
start = 0;
for(i = stack_len - 1; i >= 0; i--){
put_bits(&s->pb, 4, path[stack[i]].codebook);
count = stack[i] - path[stack[i]].prev_idx;
memset(sce->zeroes + win*16 + start, !path[stack[i]].codebook, count);
//XXX: memset when band_type is also uint8_t
for(j = 0; j < count; j++){
sce->band_type[win*16 + start] = path[stack[i]].codebook;
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 quantize_coeffs(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];
}
}
if(!cpe->ch[ch].zeroes[w + g])
sum = quantize_band(cpe->ch[ch].coeffs + start,
cpe->ch[ch].icoefs + start,
ics->swb_sizes[g],
cpe->ch[ch].sf_idx[w + g]);
else
memset(cpe->ch[ch].icoefs + start, 0, ics->swb_sizes[g] * sizeof(cpe->ch[0].icoefs[0]));
cpe->ch[ch].zeroes[w + g] = !sum;
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(AACEncContext *s, ChannelElement *cpe, int channels)
{
int q, ch, w, w2, g, start = 0;
int i;
int qcoeffs[128];
int idx;
TrellisPath paths[256*121];
int bandaddr[121];
const float lambda = 5e-7f;
int minq;
float mincost;
for(ch = 0; ch < channels; ch++){
SingleChannelElement *sce = &cpe->ch[ch];
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 mininum 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;
int bits = 0, sum = 0;
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(sce->zeroes[(w+w2)*16+g])
continue;
sum += quantize_band(coefs + w2*128, qcoeffs, sce->ics.swb_sizes[g], q);
dist += get_approximate_quant_error(coefs + w2*128, qcoeffs, sce->ics.swb_sizes[g], q) / band->threshold;
bits += get_approximate_bits(qcoeffs, sce->ics.swb_sizes[g]);
}
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 * lambda + bits
+ 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];
}
}
/**
* Encode the coefficients of one scalefactor band with selected codebook.
*/
static void encode_band_coeffs(AACEncContext *s, SingleChannelElement *sce,
int start, int size, int cb)
{
const uint8_t *bits = ff_aac_spectral_bits [cb - 1];
const uint16_t *codes = ff_aac_spectral_codes[cb - 1];
const int range = aac_cb_info[cb].range;
const int dim = (cb < FIRST_PAIR_BT) ? 4 : 2;
int i, j, idx;
//do not encode zero or special codebooks
if(range == -1) return;
if(cb == ESC_BT){
int coef_abs[2];
for(i = start; i < start + size; i += 2){
idx = 0;
for(j = 0; j < 2; j++){
coef_abs[j] = FFABS(sce->icoefs[i+j]);
idx = idx*17 + FFMIN(coef_abs[j], 16);
}
put_bits(&s->pb, bits[idx], codes[idx]);
//output signs
for(j = 0; j < 2; j++)
if(sce->icoefs[i+j])
put_bits(&s->pb, 1, sce->icoefs[i+j] < 0);
//output escape values
for(j = 0; j < 2; j++){
if(coef_abs[j] > 15){
int len = av_log2(coef_abs[j]);
put_bits(&s->pb, len - 4 + 1, (1 << (len - 4 + 1)) - 2);
put_bits(&s->pb, len, coef_abs[j] & ((1 << len) - 1));
}
}
}
}else if(IS_CODEBOOK_UNSIGNED(cb)){
for(i = start; i < start + size; i += dim){
idx = FFABS(sce->icoefs[i]);
for(j = 1; j < dim; j++)
idx = idx * range + FFABS(sce->icoefs[i+j]);
put_bits(&s->pb, bits[idx], codes[idx]);
//output signs
for(j = 0; j < dim; j++)
if(sce->icoefs[i+j])
put_bits(&s->pb, 1, sce->icoefs[i+j] < 0);
}
}else{
for(i = start; i < start + size; i += dim){
idx = sce->icoefs[i];
for(j = 1; j < dim; j++)
idx = idx * range + sce->icoefs[i+j];
//it turned out that all signed codebooks use the same offset for index coding
idx += 40;
put_bits(&s->pb, bits[idx], codes[idx]);
}
}
}
/**
* 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]);
}
}
/**
* 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++){
encode_band_coeffs(s, sce, start + w2*128,
sce->ics.swb_sizes[i],
sce->band_type[w*16 + i]);
}
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;
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];
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
}
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_quantizers(s, cpe, chans);
quantize_coeffs(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"),
};
-------------- next part --------------
/*
* AAC encoder psychoacoustic model
* 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 aacpsy.c
* AAC encoder psychoacoustic model
*/
#include "avcodec.h"
#include "aactab.h"
#include "psymodel.h"
/***********************************
* TODOs:
* thresholds linearization after their modifications for attaining given bitrate
* try other bitrate controlling mechanism (maybe use ratecontrol.c?)
* control quality for quality-based output
**********************************/
/**
* constants for 3GPP AAC psychoacoustic model
* @{
*/
#define PSY_3GPP_SPREAD_LOW 1.5f // spreading factor for ascending threshold spreading (15 dB/Bark)
#define PSY_3GPP_SPREAD_HI 3.0f // spreading factor for descending threshold spreading (30 dB/Bark)
#define PSY_3GPP_RPEMIN 0.01f
#define PSY_3GPP_RPELEV 2.0f
/**
* @}
*/
/**
* information for single band used by 3GPP TS26.403-inspired psychoacoustic model
*/
typedef struct Psy3gppBand{
float energy; ///< band energy
float ffac; ///< form factor
float thr; ///< energy threshold
float min_snr; ///< minimal SNR
float thr_quiet; ///< threshold in quiet
}Psy3gppBand;
/**
* single/pair channel context for psychoacoustic model
*/
typedef struct Psy3gppChannel{
Psy3gppBand band[128]; ///< bands information
Psy3gppBand prev_band[128]; ///< bands information from the previous frame
float win_energy; ///< sliding average of channel energy
float iir_state[2]; ///< hi-pass IIR filter state
uint8_t next_grouping; ///< stored grouping scheme for the next frame (in case of 8 short window sequence)
enum WindowSequence next_window_seq; ///< window sequence to be used in the next frame
}Psy3gppChannel;
/**
* psychoacoustic model frame type-dependent coefficients
*/
typedef struct Psy3gppCoeffs{
float ath [64]; ///< absolute threshold of hearing per bands
float barks [64]; ///< Bark value for each spectral band in long frame
float spread_low[64]; ///< spreading factor for low-to-high threshold spreading in long frame
float spread_hi [64]; ///< spreading factor for high-to-low threshold spreading in long frame
}Psy3gppCoeffs;
/**
* 3GPP TS26.403-inspired psychoacoustic model specific data
*/
typedef struct Psy3gppContext{
Psy3gppCoeffs psy_coef[2];
Psy3gppChannel *ch;
}Psy3gppContext;
/**
* Calculate Bark value for given line.
*/
static av_cold float calc_bark(float f)
{
return 13.3f * atanf(0.00076f * f) + 3.5f * atanf((f / 7500.0f) * (f / 7500.0f));
}
#define ATH_ADD 4
/**
* Calculate ATH value for given frequency.
* Borrowed from Lame.
*/
static av_cold float ath(float f, float add)
{
f /= 1000.0f;
return 3.64 * pow(f, -0.8)
- 6.8 * exp(-0.6 * (f - 3.4) * (f - 3.4))
+ 6.0 * exp(-0.15 * (f - 8.7) * (f - 8.7))
+ (0.6 + 0.04 * add) * 0.001 * f * f * f * f;
}
static av_cold int psy_3gpp_init(FFPsyContext *ctx){
Psy3gppContext *pctx;
float barks[1024];
int i, j, g, start;
float prev, minscale, minath;
ctx->model_priv_data = av_mallocz(sizeof(Psy3gppContext));
pctx = (Psy3gppContext*) ctx->model_priv_data;
for(i = 0; i < 1024; i++)
barks[i] = calc_bark(i * ctx->avctx->sample_rate / 2048.0);
minath = ath(3410, ATH_ADD);
for(j = 0; j < 2; j++){
Psy3gppCoeffs *coeffs = &pctx->psy_coef[j];
i = 0;
prev = 0.0;
for(g = 0; g < ctx->num_bands[j]; g++){
i += ctx->bands[j][g];
coeffs->barks[g] = (barks[i - 1] + prev) / 2.0;
prev = barks[i - 1];
}
for(g = 0; g < ctx->num_bands[j] - 1; g++){
coeffs->spread_low[g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_LOW);
coeffs->spread_hi [g] = pow(10.0, -(coeffs->barks[g+1] - coeffs->barks[g]) * PSY_3GPP_SPREAD_HI);
}
start = 0;
for(g = 0; g < ctx->num_bands[j]; g++){
minscale = ath(ctx->avctx->sample_rate * start / 1024.0, ATH_ADD);
for(i = 1; i < ctx->bands[j][g]; i++){
minscale = fminf(minscale, ath(ctx->avctx->sample_rate * (start + i) / 1024.0 / 2.0, ATH_ADD));
}
coeffs->ath[g] = minscale - minath;
start += ctx->bands[j][g];
}
}
pctx->ch = av_mallocz(sizeof(Psy3gppChannel) * ctx->avctx->channels);
return 0;
}
/**
* IIR filter used in block switching decision
*/
static float iir_filter(int in, float state[2])
{
float ret;
ret = 0.7548f * (in - state[0]) + 0.5095f * state[1];
state[0] = in;
state[1] = ret;
return ret;
}
/**
* window grouping information stored as bits (0 - new group, 1 - group continues)
*/
static const uint8_t window_grouping[9] = {
0xB6, 0x6C, 0xD8, 0xB2, 0x66, 0xC6, 0x96, 0x36, 0x36
};
/**
* Tell encoder which window types to use.
* @see 3GPP TS26.403 5.4.1 "Blockswitching"
*/
static FFPsyWindowInfo psy_3gpp_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
int i, j;
int br = ctx->avctx->bit_rate / ctx->avctx->channels;
int attack_ratio = br <= 16000 ? 18 : 10;
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
uint8_t grouping = 0;
FFPsyWindowInfo wi;
memset(&wi, 0, sizeof(wi));
if(la){
float s[8], v;
int switch_to_eight = 0;
float sum = 0.0, sum2 = 0.0;
int attack_n = 0;
for(i = 0; i < 8; i++){
for(j = 0; j < 128; j++){
v = iir_filter(audio[(i*128+j)*ctx->avctx->channels], pch->iir_state);
sum += v*v;
}
s[i] = sum;
sum2 += sum;
}
for(i = 0; i < 8; i++){
if(s[i] > pch->win_energy * attack_ratio){
attack_n = i + 1;
switch_to_eight = 1;
break;
}
}
pch->win_energy = pch->win_energy*7/8 + sum2/64;
wi.window_type[1] = prev_type;
switch(prev_type){
case ONLY_LONG_SEQUENCE:
wi.window_type[0] = switch_to_eight ? LONG_START_SEQUENCE : ONLY_LONG_SEQUENCE;
break;
case LONG_START_SEQUENCE:
wi.window_type[0] = EIGHT_SHORT_SEQUENCE;
grouping = pch->next_grouping;
break;
case LONG_STOP_SEQUENCE:
wi.window_type[0] = ONLY_LONG_SEQUENCE;
break;
case EIGHT_SHORT_SEQUENCE:
wi.window_type[0] = switch_to_eight ? EIGHT_SHORT_SEQUENCE : LONG_STOP_SEQUENCE;
grouping = switch_to_eight ? pch->next_grouping : 0;
break;
}
pch->next_grouping = window_grouping[attack_n];
}else{
for(i = 0; i < 3; i++)
wi.window_type[i] = prev_type;
grouping = (prev_type == EIGHT_SHORT_SEQUENCE) ? window_grouping[0] : 0;
}
wi.window_shape = 1;
if(wi.window_type[0] != EIGHT_SHORT_SEQUENCE){
wi.num_windows = 1;
wi.grouping[0] = 1;
}else{
int lastgrp = 0;
wi.num_windows = 8;
for(i = 0; i < 8; i++){
if(!((grouping >> i) & 1))
lastgrp = i;
wi.grouping[lastgrp]++;
}
}
return wi;
}
/**
* Calculate band thresholds as suggested in 3GPP TS26.403
*/
static void psy_3gpp_analyze(FFPsyContext *ctx, int channel, const float *coefs,
FFPsyWindowInfo *wi)
{
Psy3gppContext *pctx = (Psy3gppContext*) ctx->model_priv_data;
Psy3gppChannel *pch = &pctx->ch[channel];
int start = 0;
int i, w, g;
const int num_bands = ctx->num_bands[wi->num_windows == 8];
const uint8_t* band_sizes = ctx->bands[wi->num_windows == 8];
Psy3gppCoeffs *coeffs = &pctx->psy_coef[wi->num_windows == 8];
//calculate energies, initial thresholds and related values - 5.4.2 "Threshold Calculation"
for(w = 0; w < wi->num_windows*16; w += 16){
for(g = 0; g < num_bands; g++){
Psy3gppBand *band = &pch->band[w+g];
for(i = 0; i < band_sizes[g]; i++)
band->energy += coefs[start+i] * coefs[start+i];
band->energy *= 1.0f / (512*512);
band->thr = band->energy * 0.001258925f;
start += band_sizes[g];
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].energy = band->energy;
}
}
//modify thresholds - spread, threshold in quiet - 5.4.3 "Spreaded Energy Calculation"
for(w = 0; w < wi->num_windows*16; w += 16){
Psy3gppBand *band = &pch->band[w];
for(g = 1; g < num_bands; g++){
band[g].thr = FFMAX(band[g].thr, band[g-1].thr * coeffs->spread_low[g-1]);
}
for(g = num_bands - 2; g >= 0; g--){
band[g].thr = FFMAX(band[g].thr, band[g+1].thr * coeffs->spread_hi [g+1]);
}
for(g = 0; g < num_bands; g++){
band[g].thr_quiet = FFMAX(band[g].thr, coeffs->ath[g]);
if(wi->num_windows != 8 && wi->window_type[1] != EIGHT_SHORT_SEQUENCE){
band[g].thr_quiet = fmaxf(PSY_3GPP_RPEMIN*band[g].thr_quiet,
fminf(band[g].thr_quiet,
PSY_3GPP_RPELEV*pch->prev_band[w+g].thr_quiet));
}
band[g].thr = FFMAX(band[g].thr, band[g].thr_quiet * 0.25);
ctx->psy_bands[channel*PSY_MAX_BANDS+w+g].threshold = band[g].thr;
}
}
memcpy(pch->prev_band, pch->band, sizeof(pch->band));
}
static av_cold void psy_3gpp_end(FFPsyContext *apc)
{
Psy3gppContext *pctx = (Psy3gppContext*) apc->model_priv_data;
av_freep(&pctx->ch);
av_freep(&apc->model_priv_data);
}
const FFPsyModel ff_aac_psy_model =
{
.name = "3GPP TS 26.403-inspired model",
.init = psy_3gpp_init,
.window = psy_3gpp_window,
.analyze = psy_3gpp_analyze,
.end = psy_3gpp_end,
};
-------------- next part --------------
/*
* audio encoder psychoacoustic model
* 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
*/
#ifndef AVCODEC_PSYMODEL_H
#define AVCODEC_PSYMODEL_H
#include "avcodec.h"
/** maximum possible number of bands */
#define PSY_MAX_BANDS 128
/**
* single band psychoacoustic information
*/
typedef struct FFPsyBand{
int bits;
float energy;
float threshold;
float distortion;
float perceptual_weight;
}FFPsyBand;
/**
* windowing related information
*/
typedef struct FFPsyWindowInfo{
int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
int window_shape; ///< window shape (sine/KBD/whatever)
int num_windows; ///< number of windows in a frame
int grouping[8]; ///< window grouping (for e.g. AAC)
int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
}FFPsyWindowInfo;
/**
* context used by psychoacoustic model
*/
typedef struct FFPsyContext{
AVCodecContext *avctx; ///< encoder context
const struct FFPsyModel *model; ///< encoder-specific model functions
FFPsyBand *psy_bands; ///< frame bands information
uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
int *num_bands; ///< number of scalefactor bands for possible frame sizes
int num_lens; ///< number of scalefactor band sets
void* model_priv_data; ///< psychoacoustic model implementation private data
}FFPsyContext;
/**
* codec-specific psychoacoustic model implementation
*/
typedef struct FFPsyModel {
const char *name;
int (*init) (FFPsyContext *apc);
FFPsyWindowInfo (*window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type);
void (*analyze)(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi);
void (*end) (FFPsyContext *apc);
}FFPsyModel;
/**
* Initialize psychoacoustic model.
*
* @param ctx model context
* @param avctx codec context
* @param num_lens number of possible frame lengths
* @param bands scalefactor band lengths for all frame lengths
* @param num_bands number of scalefactor bands for all frame lengths
*
* @return zero if successful, a negative value if not
*/
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
int num_lens,
uint8_t **bands, int* num_bands);
/**
* Suggest window sequence for channel.
*
* @param ctx model context
* @param audio samples for the current frame
* @param la lookahead samples (NULL when unavailable)
* @param channel number of channel element to analyze
* @param prev_type previous window type
*
* @return suggested window information in a structure
*/
FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type);
/**
* Perform psychoacoustic analysis and set band info (threshold, energy).
*
* @param ctx model context
* @param channel audio channel number
* @param coeffs pointer to the transformed coefficients
* @param wi window information
*/
void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs,
FFPsyWindowInfo *wi);
/**
* Cleanup model context at the end.
*
* @param ctx model context
*/
av_cold void ff_psy_end(FFPsyContext *ctx);
/**************************************************************************
* Audio preprocessing stuff. *
* This should be moved into some audio filter eventually. *
**************************************************************************/
struct FFPsyPreprocessContext;
/**
* psychoacoustic model audio preprocessing initialization
*/
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);
/**
* Preprocess several channel in audio frame in order to compress it better.
*
* @param ctx preprocessing context
* @param audio samples to preprocess
* @param dest place to put filtered samples
* @param tag channel number
* @param channels number of channel to preprocess (some additional work may be done on stereo pair)
*/
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels);
/**
* Cleanup audio preprocessing module.
*/
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
#endif /* AVCODEC_PSYMODEL_H */
-------------- next part --------------
/*
* audio encoder psychoacoustic model
* 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
*/
#include "avcodec.h"
#include "psymodel.h"
#include "iirfilter.h"
extern const FFPsyModel ff_aac_psy_model;
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
int num_lens,
uint8_t **bands, int* num_bands)
{
ctx->avctx = avctx;
ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
switch(ctx->avctx->codec_id){
case CODEC_ID_AAC:
ctx->model = &ff_aac_psy_model;
break;
}
if(ctx->model->init)
return ctx->model->init(ctx);
return 0;
}
FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
const int16_t *audio, const int16_t *la,
int channel, int prev_type)
{
return ctx->model->window(ctx, audio, la, channel, prev_type);
}
void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
const float *coeffs, FFPsyWindowInfo *wi)
{
ctx->model->analyze(ctx, channel, coeffs, wi);
}
av_cold void ff_psy_end(FFPsyContext *ctx)
{
if(ctx->model->end)
ctx->model->end(ctx);
av_freep(&ctx->bands);
av_freep(&ctx->num_bands);
av_freep(&ctx->psy_bands);
}
typedef struct FFPsyPreprocessContext{
AVCodecContext *avctx;
float stereo_att;
struct FFIIRFilterCoeffs *fcoeffs;
struct FFIIRFilterState **fstate;
}FFPsyPreprocessContext;
#define FILT_ORDER 4
av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
{
FFPsyPreprocessContext *ctx;
int i;
ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
ctx->avctx = avctx;
ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
FILT_ORDER, 0.25, 0.0, 0.0);
if(ctx->fcoeffs){
ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
for(i = 0; i < avctx->channels; i++)
ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
}
return ctx;
}
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
const int16_t *audio, int16_t *dest,
int tag, int channels)
{
int ch, i;
if(ctx->fstate){
for(ch = 0; ch < channels; ch++){
ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
audio + ch, ctx->avctx->channels,
dest + ch, ctx->avctx->channels);
}
}else{
for(ch = 0; ch < channels; ch++){
for(i = 0; i < ctx->avctx->frame_size; i++)
dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
}
}
}
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
{
int i;
ff_iir_filter_free_coeffs(ctx->fcoeffs);
for(i = 0; i < ctx->avctx->channels; i++){
ff_iir_filter_free_state(ctx->fstate[i]);
}
av_freep(&ctx->fstate);
}
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