Mercurial > hg > pmhd
diff ffmpeg/libavcodec/aacenc.c @ 10:6840f77b83aa
commit
| author | Yading Song <yading.song@eecs.qmul.ac.uk> |
|---|---|
| date | Sun, 21 Apr 2013 10:55:35 +0200 |
| parents | |
| children |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/ffmpeg/libavcodec/aacenc.c Sun Apr 21 10:55:35 2013 +0200 @@ -0,0 +1,827 @@ +/* + * 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/float_dsp.h" +#include "libavutil/opt.h" +#include "avcodec.h" +#include "put_bits.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(AVFloatDSPContext *fdsp, \ + 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_buf; + + fdsp->vector_fmul (out, audio, lwindow, 1024); + fdsp->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_buf; + + fdsp->vector_fmul(out, audio, lwindow, 1024); + memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448); + fdsp->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_buf; + + memset(out, 0, sizeof(out[0]) * 448); + fdsp->vector_fmul(out + 448, audio + 448, swindow, 128); + memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); + fdsp->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_buf; + int w; + + for (w = 0; w < 8; w++) { + fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128); + out += 128; + in += 128; + fdsp->vector_fmul_reverse(out, in, swindow, 128); + out += 128; + } +} + +static void (*const apply_window[4])(AVFloatDSPContext *fdsp, + 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_buf; + + 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); + 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(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; + av_assert0(diff >= 0 && diff <= 120); + 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(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); +} + +/* + * Copy input samples. + * Channels are reordered from libavcodec's default order to AAC order. + */ +static void copy_input_samples(AACEncContext *s, const AVFrame *frame) +{ + int ch; + int end = 2048 + (frame ? frame->nb_samples : 0); + const uint8_t *channel_map = aac_chan_maps[s->channels - 1]; + + /* copy and remap input samples */ + for (ch = 0; ch < s->channels; 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])); + + /* copy new samples and zero any remaining samples */ + if (frame) { + memcpy(&s->planar_samples[ch][2048], + frame->extended_data[channel_map[ch]], + frame->nb_samples * sizeof(s->planar_samples[0][0])); + } + memset(&s->planar_samples[ch][end], 0, + (3072 - end) * 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; + } + + copy_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, 8192 * s->channels)) < 0) + 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(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(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); + return 0; +} + +static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) +{ + int ret = 0; + + avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); + + // 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; + + 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]; + + if (HAVE_MIPSDSPR1) + ff_aac_coder_init_mips(s); + + 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, {.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", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = 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, +}; + +/* 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", + .type = AVMEDIA_TYPE_AUDIO, + .id = AV_CODEC_ID_AAC, + .priv_data_size = sizeof(AACEncContext), + .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, + AV_SAMPLE_FMT_NONE }, + .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), + .priv_class = &aacenc_class, +};