Mercurial > libavcodec.hg
view aacenc.c @ 10874:bcfe2acbf190 libavcodec
AAC: Compress codebook tables and optimise sign bit handling
The codebooks each consist of small number of values repeated in
groups of 2 or 4. Storing the codebooks as a packed list of 2- or
4-bit indexes into a table reduces their size substantially (from 7.5k
to 1.5k), resulting in less cache pressure.
For the band types with sign bits in the bitstream, storing the number
and position of non-zero codebook values using a few bits avoids
multiple get_bits() calls and floating-point comparisons which gcc
handles miserably.
Some float/int type punning also avoids gcc brain damage.
Overall speedup 20-35% on Cortex-A8, 20% on Core i7.
author | mru |
---|---|
date | Wed, 13 Jan 2010 16:46:28 +0000 |
parents | 3d011a01a6a0 |
children | 8a4984c5cacc |
line wrap: on
line source
/* * 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: * add sane pulse detection * add temporal noise shaping ***********************************/ #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" #include "mpeg4audio.h" #include "aac.h" #include "aactab.h" #include "aacenc.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 }; /** 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 }; /** * 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, 1.0); ff_mdct_init(&s->mdct128, 8, 0, 1.0); // 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_init_ff_sine_windows(10); ff_init_ff_sine_windows(7); 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); s->coder = &ff_aac_coders[0]; s->lambda = avctx->global_quality ? avctx->global_quality : 120; #if !CONFIG_HARDCODED_TABLES for (i = 0; i < 428; i++) ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); #endif /* CONFIG_HARDCODED_TABLES */ if (avctx->channels > 5) av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. " "The output will most likely be an illegal bitstream.\n"); 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]); } /** * Produce integer coefficients from scalefactors provided by the model. */ static void adjust_frame_information(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]; } } 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; } } /** * 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]) s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); } /** * 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++) 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"), };