/* * 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 * 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; } if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) { av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile); return -1; } if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) { av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n"); 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[2]; 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) { for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) { s->output[i+1024] = audio[j] * lwindow[1024 - i - 1]; sce->saved[i] = audio[j] * lwindow[i]; } } else { for (i = 0, j = channel; i < 448; i++, j += avctx->channels) s->output[i+1024] = audio[j]; for (; 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); for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) sce->saved[i] = audio[j]; } ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output); } else { 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); } for (i = 0, j = channel; 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->cur_channel = start_ch + j; ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[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; } } } s->cur_channel = start_ch; 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; 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", AVMEDIA_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 | CODEC_CAP_EXPERIMENTAL, .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), };