diff options
Diffstat (limited to 'libavcodec/aacenc.c')
-rw-r--r-- | libavcodec/aacenc.c | 717 |
1 files changed, 492 insertions, 225 deletions
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index c247c5b..2653cef 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 */ @@ -27,9 +27,10 @@ /*********************************** * TODOs: * add sane pulse detection - * add temporal noise shaping ***********************************/ +#include "libavutil/libm.h" +#include "libavutil/thread.h" #include "libavutil/float_dsp.h" #include "libavutil/opt.h" #include "avcodec.h" @@ -42,119 +43,12 @@ #include "aac.h" #include "aactab.h" #include "aacenc.h" +#include "aacenctab.h" +#include "aacenc_utils.h" #include "psymodel.h" -#define AAC_MAX_CHANNELS 6 - -#define ERROR_IF(cond, ...) \ - if (cond) { \ - av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \ - return AVERROR(EINVAL); \ - } - -float ff_aac_pow34sf_tab[428]; - -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 -}; - -/** - * Table to remap channels from Libav's default order to AAC order. - */ -static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { - { 0 }, - { 0, 1 }, - { 2, 0, 1 }, - { 2, 0, 1, 3 }, - { 2, 0, 1, 3, 4 }, - { 2, 0, 1, 4, 5, 3 }, -}; +static AVOnce aac_table_init = AV_ONCE_INIT; /** * Make AAC audio config object. @@ -164,11 +58,12 @@ static void put_audio_specific_config(AVCodecContext *avctx) { PutBitContext pb; AACEncContext *s = avctx->priv_data; + int channels = s->channels - (s->channels == 8 ? 1 : 0); - init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); - put_bits(&pb, 5, 2); //object type - AAC-LC + init_put_bits(&pb, avctx->extradata, avctx->extradata_size); + put_bits(&pb, 5, s->profile+1); //profile put_bits(&pb, 4, s->samplerate_index); //sample rate index - put_bits(&pb, 4, s->channels); + put_bits(&pb, 4, channels); //GASpecificConfig put_bits(&pb, 1, 0); //frame length - 1024 samples put_bits(&pb, 1, 0); //does not depend on core coder @@ -181,6 +76,15 @@ static void put_audio_specific_config(AVCodecContext *avctx) flush_put_bits(&pb); } +void ff_quantize_band_cost_cache_init(struct AACEncContext *s) +{ + ++s->quantize_band_cost_cache_generation; + if (s->quantize_band_cost_cache_generation == 0) { + memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache)); + s->quantize_band_cost_cache_generation = 1; + } +} + #define WINDOW_FUNC(type) \ static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \ SingleChannelElement *sce, \ @@ -250,16 +154,17 @@ static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio) { int i; - float *output = sce->ret_buf; + const 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); else for (i = 0; i < 1024; i += 128) - s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2); + 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)); } /** @@ -275,7 +180,7 @@ static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) 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 + put_bits(&s->pb, 1, !!info->predictor_present); } else { put_bits(&s->pb, 4, info->max_sfb); for (w = 1; w < 8; w++) @@ -304,27 +209,18 @@ 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; 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; @@ -360,6 +256,67 @@ static void adjust_frame_information(ChannelElement *cpe, int chans) } } +static void apply_intensity_stereo(ChannelElement *cpe) +{ + int w, w2, g, i; + IndividualChannelStream *ics = &cpe->ch[0].ics; + if (!cpe->common_window) + return; + 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++) { + int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14); + float scale = cpe->ch[0].is_ener[w*16+g]; + if (!cpe->is_mask[w*16 + g]) { + start += ics->swb_sizes[g]; + continue; + } + if (cpe->ms_mask[w*16 + g]) + p *= -1; + for (i = 0; i < ics->swb_sizes[g]; i++) { + float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale; + cpe->ch[0].coeffs[start+i] = sum; + cpe->ch[1].coeffs[start+i] = 0.0f; + } + start += ics->swb_sizes[g]; + } + } + } +} + +static void apply_mid_side_stereo(ChannelElement *cpe) +{ + int w, w2, g, i; + IndividualChannelStream *ics = &cpe->ch[0].ics; + if (!cpe->common_window) + return; + 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++) { + /* ms_mask can be used for other purposes in PNS and I/S, + * so must not apply M/S if any band uses either, even if + * ms_mask is set. + */ + if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g] + || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT + || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) { + start += ics->swb_sizes[g]; + continue; + } + for (i = 0; i < ics->swb_sizes[g]; i++) { + float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f; + float R = L - cpe->ch[1].coeffs[start+i]; + cpe->ch[0].coeffs[start+i] = L; + cpe->ch[1].coeffs[start+i] = R; + } + start += ics->swb_sizes[g]; + } + } + } +} + /** * Encode scalefactor band coding type. */ @@ -367,6 +324,9 @@ static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) { int w; + if (s->coder->set_special_band_scalefactors) + s->coder->set_special_band_scalefactors(s, sce); + 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); } @@ -377,16 +337,30 @@ 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]); } } @@ -426,18 +400,41 @@ static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) 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], + 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], + NULL, 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, @@ -445,12 +442,19 @@ static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, int common_window) { put_bits(&s->pb, 8, sce->sf_idx[0]); - if (!common_window) + if (!common_window) { put_ics_info(s, &sce->ics); + if (s->coder->encode_main_pred) + s->coder->encode_main_pred(s, sce); + if (s->coder->encode_ltp_info) + s->coder->encode_ltp_info(s, sce, 0); + } 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, !!sce->tns.present); + if (s->coder->encode_tns_info) + s->coder->encode_tns_info(s, sce); put_bits(&s->pb, 1, 0); //ssr encode_spectral_coeffs(s, sce); return 0; @@ -478,7 +482,7 @@ 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) { @@ -508,9 +512,12 @@ 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; + SingleChannelElement *sce; + IndividualChannelStream *ics; + int i, its, ch, w, chans, tag, start_ch, ret, frame_bits; + int target_bits, rate_bits, too_many_bits, too_few_bits; + int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0; int chan_el_counter[4]; - int frame_bits; FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; if (s->last_frame == 2) @@ -536,18 +543,22 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, chans = tag == TYPE_CPE ? 2 : 1; cpe = &s->cpe[i]; for (ch = 0; ch < chans; ch++) { - IndividualChannelStream *ics = &cpe->ch[ch].ics; - int cur_channel = start_ch + ch; - overlap = &samples[cur_channel][0]; + int k; + float clip_avoidance_factor; + sce = &cpe->ch[ch]; + ics = &sce->ics; + s->cur_channel = start_ch + ch; + overlap = &samples[s->cur_channel][0]; samples2 = overlap + 1024; la = samples2 + (448+64); if (!frame) la = NULL; if (tag == TYPE_LFE) { - wi[ch].window_type[0] = ONLY_LONG_SEQUENCE; + wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE; wi[ch].window_shape = 0; wi[ch].num_windows = 1; wi[ch].grouping[0] = 1; + wi[ch].clipping[0] = 0; /* Only the lowest 12 coefficients are used in a LFE channel. * The expression below results in only the bottom 8 coefficients @@ -555,7 +566,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, */ ics->num_swb = s->samplerate_index >= 8 ? 1 : 3; } else { - wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel, + wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel, ics->window_sequence[0]); } ics->window_sequence[1] = ics->window_sequence[0]; @@ -565,24 +576,71 @@ 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]; + ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb); + ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ? + ff_swb_offset_128 [s->samplerate_index]: + ff_swb_offset_1024[s->samplerate_index]; + ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ? + ff_tns_max_bands_128 [s->samplerate_index]: + ff_tns_max_bands_1024[s->samplerate_index]; + for (w = 0; w < ics->num_windows; w++) ics->group_len[w] = wi[ch].grouping[w]; - apply_window_and_mdct(s, &cpe->ch[ch], overlap); + /* Calculate input sample maximums and evaluate clipping risk */ + clip_avoidance_factor = 0.0f; + for (w = 0; w < ics->num_windows; w++) { + const float *wbuf = overlap + w * 128; + const int wlen = 2048 / ics->num_windows; + float max = 0; + int j; + /* mdct input is 2 * output */ + for (j = 0; j < wlen; j++) + max = FFMAX(max, fabsf(wbuf[j])); + wi[ch].clipping[w] = max; + } + 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, sce, overlap); + + if (s->options.ltp && s->coder->update_ltp) { + s->coder->update_ltp(s, sce); + apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]); + s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf); + } + + for (k = 0; k < 1024; k++) { + if (!isfinite(cpe->ch[ch].coeffs[k])) { + av_log(avctx, AV_LOG_ERROR, "Input contains NaN/+-Inf\n"); + return AVERROR(EINVAL); + } + } + avoid_clipping(s, sce); } 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; - } - + frame_bits = its = 0; do { init_put_bits(&s->pb, avpkt->data, avpkt->size); if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT)) put_bitstream_info(s, LIBAVCODEC_IDENT); start_ch = 0; + target_bits = 0; memset(chan_el_counter, 0, sizeof(chan_el_counter)); for (i = 0; i < s->chan_map[0]; i++) { FFPsyWindowInfo* wi = windows + start_ch; @@ -590,16 +648,39 @@ 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]; + cpe->common_window = 0; + 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++) - coeffs[ch] = cpe->ch[ch].coeffs; + for (ch = 0; ch < chans; ch++) { + sce = &cpe->ch[ch]; + coeffs[ch] = sce->coeffs; + sce->ics.predictor_present = 0; + sce->ics.ltp.present = 0; + memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used)); + memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used)); + memset(&sce->tns, 0, sizeof(TemporalNoiseShaping)); + for (w = 0; w < 128; w++) + if (sce->band_type[w] > RESERVED_BT) + sce->band_type[w] = 0; + } + s->psy.bitres.alloc = -1; + s->psy.bitres.bits = s->last_frame_pb_count / s->channels; s->psy.model->analyze(&s->psy, start_ch, coeffs, wi); + if (s->psy.bitres.alloc > 0) { + /* Lambda unused here on purpose, we need to take psy's unscaled allocation */ + target_bits += s->psy.bitres.alloc + * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120)); + s->psy.bitres.alloc /= chans; + } + s->cur_type = tag; for (ch = 0; ch < chans; ch++) { s->cur_channel = start_ch + ch; + if (s->options.pns && s->coder->mark_pns) + s->coder->mark_pns(s, avctx, &cpe->ch[ch]); s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], 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) { @@ -612,23 +693,73 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, } } } + for (ch = 0; ch < chans; ch++) { /* TNS and PNS */ + sce = &cpe->ch[ch]; + s->cur_channel = start_ch + ch; + if (s->options.tns && s->coder->search_for_tns) + s->coder->search_for_tns(s, sce); + if (s->options.tns && s->coder->apply_tns_filt) + s->coder->apply_tns_filt(s, sce); + if (sce->tns.present) + tns_mode = 1; + if (s->options.pns && s->coder->search_for_pns) + s->coder->search_for_pns(s, avctx, sce); + } s->cur_channel = start_ch; - if (s->options.stereo_mode && cpe->common_window) { - if (s->options.stereo_mode > 0) { - IndividualChannelStream *ics = &cpe->ch[0].ics; - for (w = 0; w < ics->num_windows; w += ics->group_len[w]) - for (g = 0; g < ics->num_swb; g++) - cpe->ms_mask[w*16+g] = 1; - } else if (s->coder->search_for_ms) { - s->coder->search_for_ms(s, cpe, s->lambda); + if (s->options.intensity_stereo) { /* Intensity Stereo */ + if (s->coder->search_for_is) + s->coder->search_for_is(s, avctx, cpe); + if (cpe->is_mode) is_mode = 1; + apply_intensity_stereo(cpe); + } + if (s->options.pred) { /* Prediction */ + for (ch = 0; ch < chans; ch++) { + sce = &cpe->ch[ch]; + s->cur_channel = start_ch + ch; + if (s->options.pred && s->coder->search_for_pred) + s->coder->search_for_pred(s, sce); + if (cpe->ch[ch].ics.predictor_present) pred_mode = 1; } + if (s->coder->adjust_common_pred) + s->coder->adjust_common_pred(s, cpe); + for (ch = 0; ch < chans; ch++) { + sce = &cpe->ch[ch]; + s->cur_channel = start_ch + ch; + if (s->options.pred && s->coder->apply_main_pred) + s->coder->apply_main_pred(s, sce); + } + s->cur_channel = start_ch; + } + if (s->options.mid_side) { /* Mid/Side stereo */ + if (s->options.mid_side == -1 && s->coder->search_for_ms) + s->coder->search_for_ms(s, cpe); + else if (cpe->common_window) + memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask)); + apply_mid_side_stereo(cpe); } adjust_frame_information(cpe, chans); + if (s->options.ltp) { /* LTP */ + for (ch = 0; ch < chans; ch++) { + sce = &cpe->ch[ch]; + s->cur_channel = start_ch + ch; + if (s->coder->search_for_ltp) + s->coder->search_for_ltp(s, sce, cpe->common_window); + if (sce->ics.ltp.present) pred_mode = 1; + } + s->cur_channel = start_ch; + if (s->coder->adjust_common_ltp) + s->coder->adjust_common_ltp(s, cpe); + } if (chans == 2) { put_bits(&s->pb, 1, cpe->common_window); if (cpe->common_window) { put_ics_info(s, &cpe->ch[0].ics); + if (s->coder->encode_main_pred) + s->coder->encode_main_pred(s, &cpe->ch[0]); + if (s->coder->encode_ltp_info) + s->coder->encode_ltp_info(s, &cpe->ch[0], 1); encode_ms_info(&s->pb, cpe); + if (cpe->ms_mode) ms_mode = 1; } } for (ch = 0; ch < chans; ch++) { @@ -638,31 +769,77 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, start_ch += chans; } - frame_bits = put_bits_count(&s->pb); - if (frame_bits <= 6144 * s->channels - 3) { - s->psy.bitres.bits = frame_bits / s->channels; + if (avctx->flags & CODEC_FLAG_QSCALE) { + /* When using a constant Q-scale, don't mess with lambda */ break; } - s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; + /* rate control stuff + * allow between the nominal bitrate, and what psy's bit reservoir says to target + * but drift towards the nominal bitrate always + */ + frame_bits = put_bits_count(&s->pb); + rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate; + rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3); + too_many_bits = FFMAX(target_bits, rate_bits); + too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3); + too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits); + + /* When using ABR, be strict (but only for increasing) */ + too_few_bits = too_few_bits - too_few_bits/8; + too_many_bits = too_many_bits + too_many_bits/2; + + if ( its == 0 /* for steady-state Q-scale tracking */ + || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits)) + || frame_bits >= 6144 * s->channels - 3 ) + { + float ratio = ((float)rate_bits) / frame_bits; + + if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) { + /* + * This path is for steady-state Q-scale tracking + * When frame bits fall within the stable range, we still need to adjust + * lambda to maintain it like so in a stable fashion (large jumps in lambda + * create artifacts and should be avoided), but slowly + */ + ratio = sqrtf(sqrtf(ratio)); + ratio = av_clipf(ratio, 0.9f, 1.1f); + } else { + /* Not so fast though */ + ratio = sqrtf(ratio); + } + s->lambda = FFMIN(s->lambda * ratio, 65536.f); + /* Keep iterating if we must reduce and lambda is in the sky */ + if (ratio > 0.9f && ratio < 1.1f) { + break; + } else { + if (is_mode || ms_mode || tns_mode || pred_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)); + } + } + its++; + } + } else { + break; + } } while (1); + if (s->options.ltp && s->coder->ltp_insert_new_frame) + s->coder->ltp_insert_new_frame(s); + put_bits(&s->pb, 3, TYPE_END); flush_put_bits(&s->pb); - frame_bits = put_bits_count(&s->pb); -#if FF_API_STAT_BITS -FF_DISABLE_DEPRECATION_WARNINGS - avctx->frame_bits = frame_bits; -FF_ENABLE_DEPRECATION_WARNINGS -#endif - - // rate control stuff - if (!(avctx->flags & AV_CODEC_FLAG_QSCALE)) { - float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; - s->lambda *= ratio; - s->lambda = FFMIN(s->lambda, 65536.f); - } + + s->last_frame_pb_count = put_bits_count(&s->pb); + + s->lambda_sum += s->lambda; + s->lambda_count++; if (!frame) s->last_frame++; @@ -679,13 +856,17 @@ static av_cold int aac_encode_end(AVCodecContext *avctx) { AACEncContext *s = avctx->priv_data; + av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count); + ff_mdct_end(&s->mdct1024); ff_mdct_end(&s->mdct128); ff_psy_end(&s->psy); + ff_lpc_end(&s->lpc); if (s->psypp) 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; } @@ -694,7 +875,9 @@ 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); @@ -702,9 +885,9 @@ 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; @@ -713,8 +896,8 @@ 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 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail); for(ch = 0; ch < s->channels; ch++) @@ -725,6 +908,11 @@ alloc_fail: return AVERROR(ENOMEM); } +static av_cold void aac_encode_init_tables(void) +{ + ff_aac_tableinit(); +} + static av_cold int aac_encode_init(AVCodecContext *avctx) { AACEncContext *s = avctx->priv_data; @@ -733,28 +921,96 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) uint8_t grouping[AAC_MAX_CHANNELS]; int lengths[2]; + /* Constants */ + s->last_frame_pb_count = 0; + avctx->extradata_size = 5; avctx->frame_size = 1024; + avctx->initial_padding = 1024; + s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; + /* Channel map and unspecified bitrate guessing */ + s->channels = avctx->channels; + ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7, + "Unsupported number of channels: %d\n", s->channels); + s->chan_map = aac_chan_configs[s->channels-1]; + if (!avctx->bit_rate) { + for (i = 1; i <= s->chan_map[0]; i++) { + avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */ + s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */ + 69000 ; /* SCE */ + } + } + + /* Samplerate */ for (i = 0; i < 16; i++) if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) break; - - s->channels = avctx->channels; - - ERROR_IF(i == 16, + s->samplerate_index = i; + ERROR_IF(s->samplerate_index == 16 || + s->samplerate_index >= ff_aac_swb_size_1024_len || + s->samplerate_index >= ff_aac_swb_size_128_len, "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 %f > %d per frame requested\n", + + /* Bitrate limiting */ + WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, + "Too many bits %f > %d per frame requested, clamping to max\n", 1024.0 * avctx->bit_rate / avctx->sample_rate, 6144 * s->channels); + avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate, + avctx->bit_rate); + + /* Profile and option setting */ + avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW : + avctx->profile; + for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++) + if (avctx->profile == aacenc_profiles[i]) + break; + if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) { + avctx->profile = FF_PROFILE_AAC_LOW; + ERROR_IF(s->options.pred, + "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n"); + ERROR_IF(s->options.ltp, + "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n"); + WARN_IF(s->options.pns, + "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n"); + s->options.pns = 0; + } else if (avctx->profile == FF_PROFILE_AAC_LTP) { + s->options.ltp = 1; + ERROR_IF(s->options.pred, + "Main prediction unavailable in the \"aac_ltp\" profile\n"); + } else if (avctx->profile == FF_PROFILE_AAC_MAIN) { + s->options.pred = 1; + ERROR_IF(s->options.ltp, + "LTP prediction unavailable in the \"aac_main\" profile\n"); + } else if (s->options.ltp) { + avctx->profile = FF_PROFILE_AAC_LTP; + WARN_IF(1, + "Chainging profile to \"aac_ltp\"\n"); + ERROR_IF(s->options.pred, + "Main prediction unavailable in the \"aac_ltp\" profile\n"); + } else if (s->options.pred) { + avctx->profile = FF_PROFILE_AAC_MAIN; + WARN_IF(1, + "Chainging profile to \"aac_main\"\n"); + ERROR_IF(s->options.ltp, + "LTP prediction unavailable in the \"aac_main\" profile\n"); + } + s->profile = avctx->profile; + + /* Coder limitations */ + s->coder = &ff_aac_coders[s->options.coder]; + if (s->options.coder != AAC_CODER_TWOLOOP) { + ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, + "Coders other than twoloop require -strict -2 and some may be removed in the future\n"); + s->options.intensity_stereo = 0; + s->options.pns = 0; + } + ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, + "The LPT profile requires experimental compliance, add -strict -2 to enable!\n"); - s->samplerate_index = i; - - s->chan_map = aac_chan_configs[s->channels-1]; + /* M/S introduces horrible artifacts with multichannel files, this is temporary */ + if (s->channels > 3) + s->options.mid_side = 0; if ((ret = dsp_init(avctx, s)) < 0) goto fail; @@ -762,29 +1018,27 @@ static av_cold int aac_encode_init(AVCodecContext *avctx) if ((ret = alloc_buffers(avctx, s)) < 0) goto fail; - avctx->extradata_size = 5; 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]; + sizes[0] = ff_aac_swb_size_1024[s->samplerate_index]; + sizes[1] = ff_aac_swb_size_128[s->samplerate_index]; + lengths[0] = ff_aac_num_swb_1024[s->samplerate_index]; + lengths[1] = ff_aac_num_swb_128[s->samplerate_index]; for (i = 0; i < s->chan_map[0]; i++) grouping[i] = s->chan_map[i + 1] == TYPE_CPE; if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping)) < 0) goto fail; s->psypp = ff_psy_preprocess_init(avctx); - s->coder = &ff_aac_coders[2]; + ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON); + av_lfg_init(&s->lfg, 0x72adca55); - s->lambda = avctx->global_quality ? avctx->global_quality : 120; + if (HAVE_MIPSDSP) + ff_aac_coder_init_mips(s); - ff_aac_tableinit(); + if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0) + return AVERROR_UNKNOWN; - for (i = 0; i < 428; i++) - ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); - - avctx->initial_padding = 1024; ff_af_queue_init(avctx, &s->afq); return 0; @@ -795,10 +1049,16 @@ fail: #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM static const AVOption aacenc_options[] = { - {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"}, - {"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", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"}, + {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, + {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, + {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, + {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS}, + {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, + {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, + {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, + {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, + {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, {NULL} }; @@ -809,6 +1069,11 @@ static const AVClass aacenc_class = { LIBAVUTIL_VERSION_INT, }; +static const AVCodecDefault aac_encode_defaults[] = { + { "b", "0" }, + { NULL } +}; + AVCodec ff_aac_encoder = { .name = "aac", .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), @@ -818,8 +1083,10 @@ AVCodec ff_aac_encoder = { .init = aac_encode_init, .encode2 = aac_encode_frame, .close = aac_encode_end, - .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY | - AV_CODEC_CAP_EXPERIMENTAL, + .defaults = aac_encode_defaults, + .supported_samplerates = mpeg4audio_sample_rates, + .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, + .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .priv_class = &aacenc_class, |