| /* |
| * 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 "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 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 }, |
| }; |
| |
| /** |
| * 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; |
| |
| for (int 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 Libav's default order to AAC order. |
| */ |
| static void deinterleave_input_samples(AACEncContext *s, |
| const float *samples) |
| { |
| 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++) { |
| const float *sptr = samples + channel_map[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 */ |
| for (i = 2048; i < 3072; i++) { |
| s->planar_samples[ch][i] = *sptr; |
| sptr += sinc; |
| } |
| } |
| } |
| |
| static int aac_encode_frame(AVCodecContext *avctx, |
| uint8_t *frame, int buf_size, void *data) |
| { |
| AACEncContext *s = avctx->priv_data; |
| float **samples = s->planar_samples, *samples2, *la, *overlap; |
| ChannelElement *cpe; |
| int i, ch, w, g, chans, tag, start_ch; |
| int chan_el_counter[4]; |
| FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; |
| |
| if (s->last_frame) |
| return 0; |
| |
| if (data) { |
| deinterleave_input_samples(s, data); |
| 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 (!data) |
| 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; |
| } |
| do { |
| int frame_bits; |
| init_put_bits(&s->pb, frame, buf_size*8); |
| if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)) |
| put_bitstream_info(avctx, s, LIBAVCODEC_IDENT); |
| start_ch = 0; |
| memset(chan_el_counter, 0, sizeof(chan_el_counter)); |
| for (i = 0; i < 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 (!data) |
| s->last_frame = 1; |
| |
| return put_bits_count(&s->pb)>>3; |
| } |
| |
| static av_cold int aac_encode_end(AVCodecContext *avctx) |
| { |
| AACEncContext *s = avctx->priv_data; |
| |
| ff_mdct_end(&s->mdct1024); |
| ff_mdct_end(&s->mdct128); |
| ff_psy_end(&s->psy); |
| if (s->psypp) |
| ff_psy_preprocess_end(s->psypp); |
| av_freep(&s->buffer.samples); |
| av_freep(&s->cpe); |
| return 0; |
| } |
| |
| static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) |
| { |
| int ret = 0; |
| |
| 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) |
| { |
| 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(int 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]; |
| |
| 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])); |
| |
| 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, |
| .encode = aac_encode_frame, |
| .close = aac_encode_end, |
| .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, |
| }; |
