/* * 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 * add sane pulse detection ***********************************/ #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 }; /** 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 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_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]); 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]); } } /** * Encode pulse data. */ static void encode_pulses(AACEncContext *s, Pulse *pulse, int channel) { 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, ChannelElement *cpe, int channel) { int start, i, w, w2, wg; w = 0; for(wg = 0; wg < cpe->ch[channel].ics.num_window_groups; wg++){ start = 0; for(i = 0; i < cpe->ch[channel].ics.max_sfb; i++){ if(cpe->ch[channel].zeroes[w*16 + i]){ start += cpe->ch[channel].ics.swb_sizes[i]; continue; } for(w2 = w; w2 < w + cpe->ch[channel].ics.group_len[wg]; w2++){ encode_band_coeffs(s, cpe, channel, start + w2*128, cpe->ch[channel].ics.swb_sizes[i], cpe->ch[channel].band_type[w*16 + i]); } start += cpe->ch[channel].ics.swb_sizes[i]; } w += cpe->ch[channel].ics.group_len[wg]; } } /** * 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 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"), };