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Diffstat (limited to 'ffmpeg/libavcodec/aacenc.c')
| -rw-r--r-- | ffmpeg/libavcodec/aacenc.c | 831 |
1 files changed, 0 insertions, 831 deletions
diff --git a/ffmpeg/libavcodec/aacenc.c b/ffmpeg/libavcodec/aacenc.c deleted file mode 100644 index 5596b4b..0000000 --- a/ffmpeg/libavcodec/aacenc.c +++ /dev/null @@ -1,831 +0,0 @@ -/* - * 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 + 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; - 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 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"}, - {"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, - {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, - {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, - {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, - {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", - .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), - .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 }, - .priv_class = &aacenc_class, -}; |