| /* |
| * 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 libavcodec/aacenc.c |
| * AAC encoder |
| */ |
| |
| /*********************************** |
| * TODOs: |
| * psy model selection with some option |
| * add sane pulse detection |
| * add temporal noise shaping |
| ***********************************/ |
| |
| #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 * const 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 * const 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* const 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; |
| |
| 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, |
| swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 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(AACEncContext *s, IndividualChannelStream *info) |
| { |
| 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]); |
| } |
| } |
| |
| /** |
| * 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; |
| } |
| |
| /** |
| * 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, wg; |
| |
| w = 0; |
| for(wg = 0; wg < sce->ics.num_window_groups; wg++){ |
| 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[wg]; 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]; |
| } |
| w += sce->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"), |
| }; |
