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Okayed parts of AAC encoder

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Originally committed as revision 14752 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Kostya Shishkov 17 years ago
parent 7ca7d5fae0
commit c03d9d058b
1 changed files with 313 additions and 0 deletions
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      libavcodec/aacenc.c

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libavcodec/aacenc.c
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/*
* 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:
* psy model selection with some option
* change greedy codebook search into something more optimal, like Viterbi algorithm
* determine run lengths along with codebook
***********************************/
#include "avcodec.h"
#include "bitstream.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include "aacpsy.h"
#include "aac.h"
#include "aactab.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
};
#define CB_UNSIGNED 0x01 ///< coefficients are coded as absolute values
#define CB_PAIRS 0x02 ///< coefficients are grouped into pairs before coding (quads by default)
#define CB_ESCAPE 0x04 ///< codebook allows escapes
/** spectral coefficients codebook information */
static const struct {
int16_t maxval; ///< maximum possible value
int8_t cb_num; ///< codebook number
uint8_t flags; ///< codebook features
} aac_cb_info[] = {
{ 0, -1, CB_UNSIGNED }, // zero codebook
{ 1, 0, 0 },
{ 1, 1, 0 },
{ 2, 2, CB_UNSIGNED },
{ 2, 3, CB_UNSIGNED },
{ 4, 4, CB_PAIRS },
{ 4, 5, CB_PAIRS },
{ 7, 6, CB_PAIRS | CB_UNSIGNED },
{ 7, 7, CB_PAIRS | CB_UNSIGNED },
{ 12, 8, CB_PAIRS | CB_UNSIGNED },
{ 12, 9, CB_PAIRS | CB_UNSIGNED },
{ 8191, 10, CB_PAIRS | CB_UNSIGNED | CB_ESCAPE },
{ -1, -1, 0 }, // reserved
{ -1, -1, 0 }, // perceptual noise substitution
{ -1, -1, 0 }, // intensity out-of-phase
{ -1, -1, 0 }, // intensity in-phase
};
/** default channel configurations */
static const uint8_t aac_chan_configs[6][5] = {
{1, ID_SCE}, // 1 channel - single channel element
{1, ID_CPE}, // 2 channels - channel pair
{2, ID_SCE, ID_CPE}, // 3 channels - center + stereo
{3, ID_SCE, ID_CPE, ID_SCE}, // 4 channels - front center + stereo + back center
{3, ID_SCE, ID_CPE, ID_CPE}, // 5 channels - front center + stereo + back stereo
{4, ID_SCE, ID_CPE, ID_CPE, ID_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
DECLARE_ALIGNED_16(FFTSample, tmp[1024]); ///< temporary buffer used by MDCT
int16_t* samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
const uint8_t *swb_sizes1024; ///< scalefactor band sizes for long frame
int swb_num1024; ///< number of scalefactor bands for long frame
const uint8_t *swb_sizes128; ///< scalefactor band sizes for short frame
int swb_num128; ///< number of scalefactor bands for short frame
ChannelElement *cpe; ///< channel elements
AACPsyContext psy; ///< psychoacoustic model context
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;
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;
s->swb_sizes1024 = swb_size_1024[i];
s->swb_num1024 = ff_aac_num_swb_1024[i];
s->swb_sizes128 = swb_size_128[i];
s->swb_num128 = ff_aac_num_swb_128[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_aac_sine_long_1024, 1024);
ff_sine_window_init(ff_aac_sine_short_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]);
if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, aac_chan_configs[avctx->channels-1][0], 0, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128) < 0){
av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
return -1;
}
avctx->extradata = av_malloc(2);
avctx->extradata_size = 2;
put_audio_specific_config(avctx);
return 0;
}
/**
* Encode ics_info element.
* @see Table 4.6 (syntax of ics_info)
*/
static void put_ics_info(AVCodecContext *avctx, IndividualChannelStream *info)
{
AACEncContext *s = avctx->priv_data;
int i;
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(i = 1; i < info->num_windows; i++)
put_bits(&s->pb, 1, info->group_len[i]);
}
}
/**
* 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, ID_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 av_cold int aac_encode_end(AVCodecContext *avctx)
{
AACEncContext *s = avctx->priv_data;
ff_mdct_end(&s->mdct1024);
ff_mdct_end(&s->mdct128);
ff_aac_psy_end(&s->psy);
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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