diff options
Diffstat (limited to 'libavcodec/aacenc.c')
| -rw-r--r-- | libavcodec/aacenc.c | 231 |
1 files changed, 181 insertions, 50 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index 1e2544182c..043e6fdf35 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -2,20 +2,20 @@ * AAC encoder * Copyright (C) 2008 Konstantin Shishkov * - * This file is part of Libav. + * This file is part of FFmpeg. * - * Libav is free software; you can redistribute it and/or + * 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. * - * Libav is distributed in the hope that it will be useful, + * 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 Libav; if not, write to the Free Software + * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -53,7 +53,11 @@ return AVERROR(EINVAL); \ } -float ff_aac_pow34sf_tab[428]; +#define WARN_IF(cond, ...) \ + if (cond) { \ + av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \ + } + 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, @@ -102,7 +106,8 @@ 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 + swb_size_1024_16, swb_size_1024_16, swb_size_1024_8, + swb_size_1024_8 }; static const uint8_t swb_size_128_96[] = { @@ -131,7 +136,8 @@ static const uint8_t *swb_size_128[] = { 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 + swb_size_128_16, swb_size_128_16, swb_size_128_8, + swb_size_128_8 }; /** default channel configurations */ @@ -145,7 +151,7 @@ static const uint8_t aac_chan_configs[6][5] = { }; /** - * Table to remap channels from Libav's default order to AAC order. + * Table to remap channels from libavcodec's default order to AAC order. */ static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { { 0 }, @@ -165,7 +171,7 @@ 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); + init_put_bits(&pb, avctx->extradata, avctx->extradata_size); put_bits(&pb, 5, 2); //object type - AAC-LC put_bits(&pb, 4, s->samplerate_index); //sample rate index put_bits(&pb, 4, s->channels); @@ -252,7 +258,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, int i; float *output = sce->ret_buf; - apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio); + apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio); if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output); @@ -260,6 +266,7 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, for (i = 0; i < 1024; i += 128) s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2); memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024); + memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs)); } /** @@ -304,27 +311,47 @@ static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe) static void adjust_frame_information(ChannelElement *cpe, int chans) { int i, w, w2, g, ch; - int start, maxsfb, cmaxsfb; + int maxsfb, cmaxsfb; + IndividualChannelStream *ics; + + if (cpe->common_window) { + ics = &cpe->ch[0].ics; + for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { + for (w2 = 0; w2 < ics->group_len[w]; w2++) { + int start = (w+w2) * 128; + for (g = 0; g < ics->num_swb; g++) { + //apply Intensity stereo coeffs transformation + if (cpe->is_mask[w*16 + g]) { + int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14); + float scale = cpe->ch[0].is_ener[w*16+g]; + for (i = 0; i < ics->swb_sizes[g]; i++) { + cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + p*cpe->ch[1].pcoeffs[start+i]) * scale; + cpe->ch[1].coeffs[start+i] = 0.0f; + } + } else if (cpe->ms_mask[w*16 + g] && + cpe->ch[0].band_type[w*16 + g] < NOISE_BT && + cpe->ch[1].band_type[w*16 + g] < NOISE_BT) { + for (i = 0; i < ics->swb_sizes[g]; i++) { + cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f; + cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i]; + } + } + start += ics->swb_sizes[g]; + } + } + } + } 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++) { - //apply M/S - if (cpe->common_window && !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 (w = 0; w < ics->num_windows; w += ics->group_len[w]) { + for (w2 = 0; w2 < ics->group_len[w]; w2++) { + for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--) + ; + maxsfb = FFMAX(maxsfb, cmaxsfb); } - for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--) - ; - maxsfb = FFMAX(maxsfb, cmaxsfb); } ics->max_sfb = maxsfb; @@ -377,16 +404,30 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce) { - int off = sce->sf_idx[0], diff; + int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET; + int off_is = 0, noise_flag = 1; 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]; + if (sce->band_type[w*16 + i] == NOISE_BT) { + diff = sce->sf_idx[w*16 + i] - off_pns; + off_pns = sce->sf_idx[w*16 + i]; + if (noise_flag-- > 0) { + put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE); + continue; + } + } else if (sce->band_type[w*16 + i] == INTENSITY_BT || + sce->band_type[w*16 + i] == INTENSITY_BT2) { + diff = sce->sf_idx[w*16 + i] - off_is; + off_is = sce->sf_idx[w*16 + i]; + } else { + diff = sce->sf_idx[w*16 + i] - off_sf; + off_sf = sce->sf_idx[w*16 + i]; + } + diff += SCALE_DIFF_ZERO; + av_assert0(diff >= 0 && diff <= 120); put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]); } } @@ -431,13 +472,33 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) sce->ics.swb_sizes[i], sce->sf_idx[w*16 + i], sce->band_type[w*16 + i], - s->lambda); + s->lambda, sce->ics.window_clipping[w]); start += sce->ics.swb_sizes[i]; } } } /** + * Downscale spectral coefficients for near-clipping windows to avoid artifacts + */ +static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce) +{ + int start, i, j, w; + + if (sce->ics.clip_avoidance_factor < 1.0f) { + for (w = 0; w < sce->ics.num_windows; w++) { + start = 0; + for (i = 0; i < sce->ics.max_sfb; i++) { + float *swb_coeffs = sce->coeffs + start + w*128; + for (j = 0; j < sce->ics.swb_sizes[i]; j++) + swb_coeffs[j] *= sce->ics.clip_avoidance_factor; + start += sce->ics.swb_sizes[i]; + } + } + } +} + +/** * Encode one channel of audio data. */ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, @@ -478,7 +539,7 @@ static void put_bitstream_info(AACEncContext *s, const char *name) /* * Copy input samples. - * Channels are reordered from Libav's default order to AAC order. + * Channels are reordered from libavcodec's default order to AAC order. */ static void copy_input_samples(AACEncContext *s, const AVFrame *frame) { @@ -508,7 +569,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, AACEncContext *s = avctx->priv_data; float **samples = s->planar_samples, *samples2, *la, *overlap; ChannelElement *cpe; - int i, ch, w, g, chans, tag, start_ch, ret; + int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0, is_mode = 0; int chan_el_counter[4]; FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; @@ -537,6 +598,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, for (ch = 0; ch < chans; ch++) { IndividualChannelStream *ics = &cpe->ch[ch].ics; int cur_channel = start_ch + ch; + float clip_avoidance_factor; overlap = &samples[cur_channel][0]; samples2 = overlap + 1024; la = samples2 + (448+64); @@ -564,18 +626,34 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, ics->num_windows = wi[ch].num_windows; ics->swb_sizes = s->psy.bands [ics->num_windows == 8]; ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8]; + clip_avoidance_factor = 0.0f; for (w = 0; w < ics->num_windows; w++) ics->group_len[w] = wi[ch].grouping[w]; + for (w = 0; w < ics->num_windows; w++) { + if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) { + ics->window_clipping[w] = 1; + clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]); + } else { + ics->window_clipping[w] = 0; + } + } + if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) { + ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor; + } else { + ics->clip_avoidance_factor = 1.0f; + } apply_window_and_mdct(s, &cpe->ch[ch], overlap); + if (isnan(cpe->ch->coeffs[0])) { + av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n"); + return AVERROR(EINVAL); + } + avoid_clipping(s, &cpe->ch[ch]); } start_ch += chans; } - if ((ret = ff_alloc_packet(avpkt, 768 * s->channels))) { - av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); + if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0) return ret; - } - do { int frame_bits; @@ -591,6 +669,8 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, tag = s->chan_map[i+1]; chans = tag == TYPE_CPE ? 2 : 1; cpe = &s->cpe[i]; + memset(cpe->is_mask, 0, sizeof(cpe->is_mask)); + memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask)); put_bits(&s->pb, 3, tag); put_bits(&s->pb, 4, chan_el_counter[tag]++); for (ch = 0; ch < chans; ch++) @@ -613,6 +693,12 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, } } } + if (s->options.pns && s->coder->search_for_pns) { + for (ch = 0; ch < chans; ch++) { + s->cur_channel = start_ch + ch; + s->coder->search_for_pns(s, avctx, &cpe->ch[ch], s->lambda); + } + } s->cur_channel = start_ch; if (s->options.stereo_mode && cpe->common_window) { if (s->options.stereo_mode > 0) { @@ -624,12 +710,20 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, s->coder->search_for_ms(s, cpe, s->lambda); } } + if (chans > 1 && s->options.intensity_stereo && s->coder->search_for_is) { + s->coder->search_for_is(s, avctx, cpe, s->lambda); + if (cpe->is_mode) is_mode = 1; + } + if (s->coder->set_special_band_scalefactors) + for (ch = 0; ch < chans; ch++) + s->coder->set_special_band_scalefactors(s, &cpe->ch[ch]); adjust_frame_information(cpe, chans); 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); + if (cpe->ms_mode) ms_mode = 1; } } for (ch = 0; ch < chans; ch++) { @@ -644,6 +738,15 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, s->psy.bitres.bits = frame_bits / s->channels; break; } + if (is_mode || ms_mode) { + for (i = 0; i < s->chan_map[0]; i++) { + // Must restore coeffs + chans = tag == TYPE_CPE ? 2 : 1; + cpe = &s->cpe[i]; + for (ch = 0; ch < chans; ch++) + memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs)); + } + } s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; @@ -682,6 +785,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) ff_psy_preprocess_end(s->psypp); av_freep(&s->buffer.samples); av_freep(&s->cpe); + av_freep(&s->fdsp); ff_af_queue_close(&s->afq); return 0; } @@ -690,7 +794,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) { int ret = 0; - avpriv_float_dsp_init(&s->fdsp, avctx->flags & AV_CODEC_FLAG_BITEXACT); + s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); + if (!s->fdsp) + return AVERROR(ENOMEM); // window init ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); @@ -698,9 +804,9 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) ff_init_ff_sine_windows(10); ff_init_ff_sine_windows(7); - if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) + if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0) return ret; - if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) + if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0) return ret; return 0; @@ -709,8 +815,8 @@ static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s) { int ch; - FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail); - FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail); + FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail); + FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail); FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); for(ch = 0; ch < s->channels; ch++) @@ -737,14 +843,20 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) s->channels = avctx->channels; - ERROR_IF(i == 16, + ERROR_IF(i == 16 + || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024)) + || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)), "Unsupported sample rate %d\n", avctx->sample_rate); ERROR_IF(s->channels > AAC_MAX_CHANNELS, "Unsupported number of channels: %d\n", s->channels); ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW, "Unsupported profile %d\n", avctx->profile); - ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, - "Too many bits per frame requested\n"); + WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, + "Too many bits per frame requested, clamping to max\n"); + + avctx->bit_rate = (int)FFMIN( + 6144 * s->channels / 1024.0 * avctx->sample_rate, + avctx->bit_rate); s->samplerate_index = i; @@ -769,14 +881,14 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) s->chan_map[0], grouping)) < 0) goto fail; s->psypp = ff_psy_preprocess_init(avctx); - s->coder = &ff_aac_coders[2]; + s->coder = &ff_aac_coders[s->options.aac_coder]; - s->lambda = avctx->global_quality ? avctx->global_quality : 120; + if (HAVE_MIPSDSPR1) + ff_aac_coder_init_mips(s); - ff_aac_tableinit(); + s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; - for (i = 0; i < 428; i++) - ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); + ff_aac_tableinit(); avctx->initial_padding = 1024; ff_af_queue_init(avctx, &s->afq); @@ -793,6 +905,17 @@ static const AVOption aacenc_options[] = { {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, + {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"}, + {"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, + {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"}, + {"disable", "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"}, + {"enable", "Enable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"}, + {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "intensity_stereo"}, + {"disable", "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"}, + {"enable", "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"}, {NULL} }; @@ -803,6 +926,13 @@ static const AVClass aacenc_class = { LIBAVUTIL_VERSION_INT, }; +/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build + * failures */ +static const int mpeg4audio_sample_rates[16] = { + 96000, 88200, 64000, 48000, 44100, 32000, + 24000, 22050, 16000, 12000, 11025, 8000, 7350 +}; + AVCodec ff_aac_encoder = { .name = "aac", .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), @@ -812,6 +942,7 @@ AVCodec ff_aac_encoder = { .init = aac_encode_init, .encode2 = aac_encode_frame, .close = aac_encode_end, + .supported_samplerates = mpeg4audio_sample_rates, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, |