/* * 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 "libavutil/opt.h" #include "avcodec.h" #include "put_bits.h" #include "dsputil.h" #include "internal.h" #include "mpeg4audio.h" #include "kbdwin.h" #include "sinewin.h" #include "aac.h" #include "aactab.h" #include "aacenc.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 libavcodec'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 }, }; /** * 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, s->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 //Explicitly Mark SBR absent put_bits(&pb, 11, 0x2b7); //sync extension put_bits(&pb, 5, AOT_SBR); put_bits(&pb, 1, 0); flush_put_bits(&pb); } #define WINDOW_FUNC(type) \ static void apply_ ##type ##_window(DSPContext *dsp, SingleChannelElement *sce, const float *audio) WINDOW_FUNC(only_long) { const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; float *out = sce->ret; dsp->vector_fmul (out, audio, lwindow, 1024); dsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024); } WINDOW_FUNC(long_start) { const float *lwindow = sce->ics.use_kb_window[1] ? 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; float *out = sce->ret; dsp->vector_fmul(out, audio, lwindow, 1024); memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448); dsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128); memset(out + 1024 + 576, 0, sizeof(out[0]) * 448); } WINDOW_FUNC(long_stop) { 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[1] ? ff_aac_kbd_short_128 : ff_sine_128; float *out = sce->ret; memset(out, 0, sizeof(out[0]) * 448); dsp->vector_fmul(out + 448, audio + 448, swindow, 128); memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); dsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024); } WINDOW_FUNC(eight_short) { 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; const float *in = audio + 448; float *out = sce->ret; int w; for (w = 0; w < 8; w++) { dsp->vector_fmul (out, in, w ? pwindow : swindow, 128); out += 128; in += 128; dsp->vector_fmul_reverse(out, in, swindow, 128); out += 128; } } static void (*const apply_window[4])(DSPContext *dsp, SingleChannelElement *sce, const float *audio) = { [ONLY_LONG_SEQUENCE] = apply_only_long_window, [LONG_START_SEQUENCE] = apply_long_start_window, [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window, [LONG_STOP_SEQUENCE] = apply_long_stop_window }; static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio) { int i; float *output = sce->ret; apply_window[sce->ics.window_sequence[0]](&s->dsp, 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); memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024); } /** * 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, 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 (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 * ics0->num_windows ? 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 - 14); put_bits(&s->pb, 4, 0); //extension type - filler padbits = -put_bits_count(&s->pb) & 7; avpriv_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); } /* * Deinterleave input samples. * Channels are reordered from libavcodec's default order to AAC order. */ static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame) { int ch, i; const int sinc = s->channels; const uint8_t *channel_map = aac_chan_maps[sinc - 1]; /* deinterleave and remap input samples */ for (ch = 0; ch < sinc; ch++) { /* copy last 1024 samples of previous frame to the start of the current frame */ memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0])); /* deinterleave */ i = 2048; if (frame) { const float *sptr = ((const float *)frame->data[0]) + channel_map[ch]; for (; i < 2048 + frame->nb_samples; i++) { s->planar_samples[ch][i] = *sptr; sptr += sinc; } } memset(&s->planar_samples[ch][i], 0, (3072 - i) * sizeof(s->planar_samples[0][0])); } } static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr) { 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 chan_el_counter[4]; FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; if (s->last_frame == 2) return 0; /* add current frame to queue */ if (frame) { if ((ret = ff_af_queue_add(&s->afq, frame) < 0)) return ret; } deinterleave_input_samples(s, frame); if (s->psypp) ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); if (!avctx->frame_number) return 0; start_ch = 0; for (i = 0; i < s->chan_map[0]; i++) { FFPsyWindowInfo* wi = windows + start_ch; tag = s->chan_map[i+1]; 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]; 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_shape = 0; wi[ch].num_windows = 1; wi[ch].grouping[0] = 1; /* Only the lowest 12 coefficients are used in a LFE channel. * The expression below results in only the bottom 8 coefficients * being used for 11.025kHz to 16kHz sample rates. */ ics->num_swb = s->samplerate_index >= 8 ? 1 : 3; } else { wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel, ics->window_sequence[0]); } ics->window_sequence[1] = ics->window_sequence[0]; ics->window_sequence[0] = wi[ch].window_type[0]; ics->use_kb_window[1] = ics->use_kb_window[0]; ics->use_kb_window[0] = wi[ch].window_shape; 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]; 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); } start_ch += chans; } if ((ret = ff_alloc_packet2(avctx, avpkt, 768 * s->channels))) return ret; do { int frame_bits; init_put_bits(&s->pb, avpkt->data, avpkt->size); 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 < s->chan_map[0]; i++) { FFPsyWindowInfo* wi = windows + start_ch; const float *coeffs[2]; tag = s->chan_map[i+1]; chans = tag == TYPE_CPE ? 2 : 1; cpe = &s->cpe[i]; 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; s->psy.model->analyze(&s->psy, start_ch, coeffs, wi); for (ch = 0; ch < chans; ch++) { s->cur_channel = start_ch * 2 + 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) { cpe->common_window = 1; for (w = 0; w < wi[0].num_windows; w++) { if (wi[0].grouping[w] != wi[1].grouping[w]) { cpe->common_window = 0; break; } } } s->cur_channel = start_ch * 2; 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); } } adjust_frame_information(s, 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); } } for (ch = 0; ch < chans; ch++) { s->cur_channel = start_ch + ch; encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window); } 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; 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 (!frame) s->last_frame++; ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, &avpkt->duration); avpkt->size = put_bits_count(&s->pb) >> 3; *got_packet_ptr = 1; return 0; } 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); if (s->psypp) ff_psy_preprocess_end(s->psypp); av_freep(&s->buffer.samples); av_freep(&s->cpe); ff_af_queue_close(&s->afq); #if FF_API_OLD_ENCODE_AUDIO av_freep(&avctx->coded_frame); #endif return 0; } static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) { int ret = 0; ff_dsputil_init(&s->dsp, avctx); // 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); if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) return ret; if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) return ret; return 0; } 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_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); for(ch = 0; ch < s->channels; ch++) s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; #if FF_API_OLD_ENCODE_AUDIO if (!(avctx->coded_frame = avcodec_alloc_frame())) goto alloc_fail; #endif return 0; alloc_fail: return AVERROR(ENOMEM); } static av_cold int aac_encode_init(AVCodecContext *avctx) { AACEncContext *s = avctx->priv_data; int i, ret = 0; const uint8_t *sizes[2]; uint8_t grouping[AAC_MAX_CHANNELS]; int lengths[2]; avctx->frame_size = 1024; for (i = 0; i < 16; i++) if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) break; s->channels = avctx->channels; ERROR_IF(i == 16, "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"); s->samplerate_index = i; s->chan_map = aac_chan_configs[s->channels-1]; if (ret = dsp_init(avctx, s)) goto fail; if (ret = alloc_buffers(avctx, s)) 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]; 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)) goto fail; s->psypp = ff_psy_preprocess_init(avctx); s->coder = &ff_aac_coders[s->options.aac_coder]; s->lambda = avctx->global_quality ? avctx->global_quality : 120; ff_aac_tableinit(); for (i = 0; i < 428; i++) ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); avctx->delay = 1024; ff_af_queue_init(avctx, &s->afq); return 0; fail: aac_encode_end(avctx); return ret; } #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, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"}, {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.dbl = 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, {.dbl = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.dbl = 2}, 0, AAC_CODER_NB-1, AACENC_FLAGS}, {NULL} }; static const AVClass aacenc_class = { "AAC encoder", av_default_item_name, aacenc_options, LIBAVUTIL_VERSION_INT, }; AVCodec ff_aac_encoder = { .name = "aac", .type = AVMEDIA_TYPE_AUDIO, .id = CODEC_ID_AAC, .priv_data_size = sizeof(AACEncContext), .init = aac_encode_init, .encode2 = aac_encode_frame, .close = aac_encode_end, .supported_samplerates = avpriv_mpeg4audio_sample_rates, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), .priv_class = &aacenc_class, };