mirror of
https://github.com/FFmpeg/FFmpeg.git
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642 lines
22 KiB
C
642 lines
22 KiB
C
/*
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* AAC encoder
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* Copyright (C) 2008 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file libavcodec/aacenc.c
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* AAC encoder
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*/
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/***********************************
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* TODOs:
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* add sane pulse detection
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* add temporal noise shaping
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***********************************/
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#include "avcodec.h"
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#include "put_bits.h"
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#include "dsputil.h"
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#include "mpeg4audio.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacenc.h"
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#include "psymodel.h"
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static const uint8_t swb_size_1024_96[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_64[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
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40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
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};
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static const uint8_t swb_size_1024_48[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
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96
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};
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static const uint8_t swb_size_1024_32[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
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};
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static const uint8_t swb_size_1024_24[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_16[] = {
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8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
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32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
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};
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static const uint8_t swb_size_1024_8[] = {
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12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
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16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
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};
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static const uint8_t *swb_size_1024[] = {
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swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
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swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
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swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
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};
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static const uint8_t swb_size_128_96[] = {
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4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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};
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static const uint8_t swb_size_128_48[] = {
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4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
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};
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static const uint8_t swb_size_128_24[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
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};
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static const uint8_t swb_size_128_16[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
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};
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static const uint8_t swb_size_128_8[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
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};
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static const uint8_t *swb_size_128[] = {
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/* the last entry on the following row is swb_size_128_64 but is a
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duplicate of swb_size_128_96 */
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swb_size_128_96, swb_size_128_96, swb_size_128_96,
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swb_size_128_48, swb_size_128_48, swb_size_128_48,
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swb_size_128_24, swb_size_128_24, swb_size_128_16,
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swb_size_128_16, swb_size_128_16, swb_size_128_8
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};
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/** default channel configurations */
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static const uint8_t aac_chan_configs[6][5] = {
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{1, TYPE_SCE}, // 1 channel - single channel element
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{1, TYPE_CPE}, // 2 channels - channel pair
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{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
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{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
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{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
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{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
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};
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/**
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* Make AAC audio config object.
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* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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*/
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static void put_audio_specific_config(AVCodecContext *avctx)
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{
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PutBitContext pb;
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AACEncContext *s = avctx->priv_data;
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init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
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put_bits(&pb, 5, 2); //object type - AAC-LC
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put_bits(&pb, 4, s->samplerate_index); //sample rate index
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put_bits(&pb, 4, avctx->channels);
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//GASpecificConfig
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put_bits(&pb, 1, 0); //frame length - 1024 samples
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put_bits(&pb, 1, 0); //does not depend on core coder
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put_bits(&pb, 1, 0); //is not extension
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flush_put_bits(&pb);
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}
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static av_cold int aac_encode_init(AVCodecContext *avctx)
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{
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AACEncContext *s = avctx->priv_data;
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int i;
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const uint8_t *sizes[2];
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int lengths[2];
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avctx->frame_size = 1024;
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for (i = 0; i < 16; i++)
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if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
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break;
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if (i == 16) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
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return -1;
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}
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if (avctx->channels > 6) {
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av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
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return -1;
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}
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s->samplerate_index = i;
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dsputil_init(&s->dsp, avctx);
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ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
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ff_mdct_init(&s->mdct128, 8, 0, 1.0);
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// window init
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ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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ff_init_ff_sine_windows(10);
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ff_init_ff_sine_windows(7);
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s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
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s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
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avctx->extradata = av_malloc(2);
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avctx->extradata_size = 2;
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put_audio_specific_config(avctx);
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sizes[0] = swb_size_1024[i];
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sizes[1] = swb_size_128[i];
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lengths[0] = ff_aac_num_swb_1024[i];
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lengths[1] = ff_aac_num_swb_128[i];
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ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
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s->psypp = ff_psy_preprocess_init(avctx);
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s->coder = &ff_aac_coders[0];
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s->lambda = avctx->global_quality ? avctx->global_quality : 120;
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#if !CONFIG_HARDCODED_TABLES
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for (i = 0; i < 428; i++)
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ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
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#endif /* CONFIG_HARDCODED_TABLES */
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if (avctx->channels > 5)
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av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
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"The output will most likely be an illegal bitstream.\n");
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return 0;
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}
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static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
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SingleChannelElement *sce, short *audio, int channel)
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{
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int i, j, k;
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const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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memcpy(s->output, sce->saved, sizeof(float)*1024);
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if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
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memset(s->output, 0, sizeof(s->output[0]) * 448);
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for (i = 448; i < 576; i++)
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s->output[i] = sce->saved[i] * pwindow[i - 448];
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for (i = 576; i < 704; i++)
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s->output[i] = sce->saved[i];
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}
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if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
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j = channel;
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for (i = 0; i < 1024; i++, j += avctx->channels) {
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s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
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sce->saved[i] = audio[j] * lwindow[i];
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}
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} else {
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j = channel;
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for (i = 0; i < 448; i++, j += avctx->channels)
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s->output[i+1024] = audio[j];
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for (i = 448; i < 576; i++, j += avctx->channels)
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s->output[i+1024] = audio[j] * swindow[576 - i - 1];
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memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
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j = channel;
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for (i = 0; i < 1024; i++, j += avctx->channels)
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sce->saved[i] = audio[j];
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}
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ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
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} else {
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j = channel;
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for (k = 0; k < 1024; k += 128) {
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for (i = 448 + k; i < 448 + k + 256; i++)
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s->output[i - 448 - k] = (i < 1024)
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? sce->saved[i]
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: audio[channel + (i-1024)*avctx->channels];
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s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
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s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
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ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
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}
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j = channel;
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for (i = 0; i < 1024; i++, j += avctx->channels)
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sce->saved[i] = audio[j];
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}
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}
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/**
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* Encode ics_info element.
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* @see Table 4.6 (syntax of ics_info)
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*/
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static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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{
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int w;
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put_bits(&s->pb, 1, 0); // ics_reserved bit
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put_bits(&s->pb, 2, info->window_sequence[0]);
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put_bits(&s->pb, 1, info->use_kb_window[0]);
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if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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put_bits(&s->pb, 6, info->max_sfb);
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put_bits(&s->pb, 1, 0); // no prediction
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} else {
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put_bits(&s->pb, 4, info->max_sfb);
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for (w = 1; w < 8; w++)
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put_bits(&s->pb, 1, !info->group_len[w]);
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}
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}
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/**
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* Encode MS data.
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* @see 4.6.8.1 "Joint Coding - M/S Stereo"
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*/
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static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
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{
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int i, w;
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put_bits(pb, 2, cpe->ms_mode);
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if (cpe->ms_mode == 1)
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for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
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for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
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put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
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}
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/**
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* Produce integer coefficients from scalefactors provided by the model.
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*/
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static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
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{
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int i, w, w2, g, ch;
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int start, sum, maxsfb, cmaxsfb;
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for (ch = 0; ch < chans; ch++) {
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IndividualChannelStream *ics = &cpe->ch[ch].ics;
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start = 0;
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maxsfb = 0;
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cpe->ch[ch].pulse.num_pulse = 0;
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for (w = 0; w < ics->num_windows*16; w += 16) {
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for (g = 0; g < ics->num_swb; g++) {
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sum = 0;
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//apply M/S
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if (!ch && cpe->ms_mask[w + g]) {
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
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cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
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}
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}
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start += ics->swb_sizes[g];
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}
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for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
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;
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maxsfb = FFMAX(maxsfb, cmaxsfb);
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}
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ics->max_sfb = maxsfb;
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//adjust zero bands for window groups
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (g = 0; g < ics->max_sfb; g++) {
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i = 1;
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for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
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if (!cpe->ch[ch].zeroes[w2*16 + g]) {
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i = 0;
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break;
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}
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}
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cpe->ch[ch].zeroes[w*16 + g] = i;
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}
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}
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}
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if (chans > 1 && cpe->common_window) {
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IndividualChannelStream *ics0 = &cpe->ch[0].ics;
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IndividualChannelStream *ics1 = &cpe->ch[1].ics;
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int msc = 0;
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ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
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ics1->max_sfb = ics0->max_sfb;
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for (w = 0; w < ics0->num_windows*16; w += 16)
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for (i = 0; i < ics0->max_sfb; i++)
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if (cpe->ms_mask[w+i])
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msc++;
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if (msc == 0 || ics0->max_sfb == 0)
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cpe->ms_mode = 0;
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else
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cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
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}
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}
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/**
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* Encode scalefactor band coding type.
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*/
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static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
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{
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int w;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
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s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
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}
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/**
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* Encode scalefactors.
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*/
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static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
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SingleChannelElement *sce)
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{
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int off = sce->sf_idx[0], diff;
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int i, w;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (i = 0; i < sce->ics.max_sfb; i++) {
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if (!sce->zeroes[w*16 + i]) {
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diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
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if (diff < 0 || diff > 120)
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av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
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off = sce->sf_idx[w*16 + i];
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put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
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}
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}
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}
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}
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/**
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* Encode pulse data.
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*/
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static void encode_pulses(AACEncContext *s, Pulse *pulse)
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{
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int i;
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put_bits(&s->pb, 1, !!pulse->num_pulse);
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if (!pulse->num_pulse)
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return;
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put_bits(&s->pb, 2, pulse->num_pulse - 1);
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put_bits(&s->pb, 6, pulse->start);
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for (i = 0; i < pulse->num_pulse; i++) {
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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++)
|
|
s->coder->quantize_and_encode_band(s, &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];
|
|
FFPsyWindowInfo windows[avctx->channels];
|
|
|
|
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, 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;
|
|
}
|
|
|
|
start_ch = 0;
|
|
for (i = 0; i < chan_map[0]; i++) {
|
|
FFPsyWindowInfo* wi = windows + start_ch;
|
|
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];
|
|
|
|
s->cur_channel = start_ch + j;
|
|
apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
do {
|
|
int frame_bits;
|
|
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 = windows + start_ch;
|
|
tag = chan_map[i+1];
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
for (j = 0; j < chans; j++) {
|
|
s->coder->search_for_quantizers(avctx, 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;
|
|
}
|
|
}
|
|
}
|
|
if (cpe->common_window && s->coder->search_for_ms)
|
|
s->coder->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;
|
|
}
|
|
|
|
frame_bits = put_bits_count(&s->pb);
|
|
if (frame_bits <= 6144 * avctx->channels - 3)
|
|
break;
|
|
|
|
s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
|
|
|
|
} while (1);
|
|
|
|
put_bits(&s->pb, 3, TYPE_END);
|
|
flush_put_bits(&s->pb);
|
|
avctx->frame_bits = put_bits_count(&s->pb);
|
|
|
|
// rate control stuff
|
|
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
|
|
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
|
|
s->lambda *= ratio;
|
|
s->lambda = FFMIN(s->lambda, 65536.f);
|
|
}
|
|
|
|
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 = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
|
|
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
|
|
};
|