X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Faacenc.c;h=f122fe11acb7eb7002b13e4e760c66697c9a86b1;hb=080ce9071dc1d05fcfd40629eeb6d4a163abd840;hp=838863b8b8e39c2376db5cc408fbbcabe0dbf6d5;hpb=817015e4e2e8d9efbec446033341c9117f889bb8;p=ffmpeg diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 838863b8b8e..f122fe11acb 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -20,24 +20,28 @@ */ /** - * @file aacenc.c + * @file * AAC encoder */ /*********************************** * TODOs: - * psy model selection with some option * add sane pulse detection + * add temporal noise shaping ***********************************/ #include "avcodec.h" -#include "bitstream.h" +#include "put_bits.h" #include "dsputil.h" #include "mpeg4audio.h" -#include "aacpsy.h" #include "aac.h" #include "aactab.h" +#include "aacenc.h" + +#include "psymodel.h" + +#define AAC_MAX_CHANNELS 6 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, @@ -120,24 +124,14 @@ static const uint8_t *swb_size_128[] = { /** default channel configurations */ static const uint8_t aac_chan_configs[6][5] = { - {1, ID_SCE}, // 1 channel - single channel element - {1, ID_CPE}, // 2 channels - channel pair - {2, ID_SCE, ID_CPE}, // 3 channels - center + stereo - {3, ID_SCE, ID_CPE, ID_SCE}, // 4 channels - front center + stereo + back center - {3, ID_SCE, ID_CPE, ID_CPE}, // 5 channels - front center + stereo + back stereo - {4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE + {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 }; -/** - * 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; -} AACEncContext; - /** * Make AAC audio config object. * @see 1.6.2.1 "Syntax - AudioSpecificConfig" @@ -162,83 +156,256 @@ 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]) + for (i = 0; i < 16; i++) + if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) break; - if(i == 16){ + if (i == 16) { av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); return -1; } - if(avctx->channels > 6){ + if (avctx->channels > AAC_MAX_CHANNELS) { 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; - 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); + 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_sine_window_init(ff_sine_1024, 1024); - ff_sine_window_init(ff_sine_128, 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]); - 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); + 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_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE); 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; + + ff_aac_tableinit(); + 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(AVCodecContext *avctx, IndividualChannelStream *info) +static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) { - AACEncContext *s = avctx->priv_data; - int i; + 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){ + 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{ + } 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]); + 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(AVCodecContext *avctx, AACEncContext *s, Pulse *pulse, int channel) +static void encode_pulses(AACEncContext *s, Pulse *pulse) { int i; put_bits(&s->pb, 1, !!pulse->num_pulse); - if(!pulse->num_pulse) return; + 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->offset[i]); + for (i = 0; i < pulse->num_pulse; i++) { + put_bits(&s->pb, 5, pulse->pos[i]); put_bits(&s->pb, 4, pulse->amp[i]); } } @@ -246,54 +413,224 @@ static void encode_pulses(AVCodecContext *avctx, AACEncContext *s, Pulse *pulse, /** * Encode spectral coefficients processed by psychoacoustic model. */ -static void encode_spectral_coeffs(AVCodecContext *avctx, AACEncContext *s, ChannelElement *cpe, int channel) +static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) { - int start, i, w, w2, wg; + int start, i, w, w2; - w = 0; - for(wg = 0; wg < cpe->ch[channel].ics.num_window_groups; wg++){ + for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { start = 0; - for(i = 0; i < cpe->ch[channel].ics.max_sfb; i++){ - if(cpe->ch[channel].zeroes[w][i]){ - start += cpe->ch[channel].ics.swb_sizes[i]; + 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 + 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][i]); - } - start += cpe->ch[channel].ics.swb_sizes[i]; + 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]; } - w += cpe->ch[channel].ics.group_len[wg]; } } +/** + * 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) +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, ID_FIL); + put_bits(&s->pb, 3, TYPE_FIL); put_bits(&s->pb, 4, FFMIN(namelen, 15)); - if(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++) + 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[AAC_MAX_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 + (448+64) * avctx->channels + start_ch; + if (!data) + la = NULL; + for (j = 0; j < chans; j++) { + IndividualChannelStream *ics = &cpe->ch[j].ics; + int k; + if (tag == TYPE_LFE) { + wi[j].window_type[0] = ONLY_LONG_SEQUENCE; + wi[j].window_shape = 0; + wi[j].num_windows = 1; + wi[j].grouping[0] = 1; + } else { + 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 = tag == TYPE_LFE ? 12 : 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_aac_psy_end(&s->psy); + ff_psy_end(&s->psy); + ff_psy_preprocess_end(s->psypp); av_freep(&s->samples); av_freep(&s->cpe); return 0; @@ -301,13 +638,13 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) AVCodec aac_encoder = { "aac", - CODEC_TYPE_AUDIO, + 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, - .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, + .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"), };