| /* |
| * 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 |
| ***********************************/ |
| |
| #include "libavutil/libm.h" |
| #include "libavutil/thread.h" |
| #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 "aacenctab.h" |
| #include "aacenc_utils.h" |
| |
| #include "psymodel.h" |
| |
| static AVOnce aac_table_init = AV_ONCE_INIT; |
| |
| /** |
| * 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; |
| int channels = s->channels - (s->channels == 8 ? 1 : 0); |
| |
| init_put_bits(&pb, avctx->extradata, avctx->extradata_size); |
| put_bits(&pb, 5, s->profile+1); //profile |
| put_bits(&pb, 4, s->samplerate_index); //sample rate index |
| put_bits(&pb, 4, 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); |
| } |
| |
| void ff_quantize_band_cost_cache_init(struct AACEncContext *s) |
| { |
| int sf, g; |
| for (sf = 0; sf < 256; sf++) { |
| for (g = 0; g < 128; g++) { |
| s->quantize_band_cost_cache[sf][g].bits = -1; |
| } |
| } |
| } |
| |
| #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; |
| const 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); |
| memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs)); |
| } |
| |
| /** |
| * 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, !!info->predictor_present); |
| } 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 maxsfb, cmaxsfb; |
| |
| for (ch = 0; ch < chans; ch++) { |
| IndividualChannelStream *ics = &cpe->ch[ch].ics; |
| maxsfb = 0; |
| cpe->ch[ch].pulse.num_pulse = 0; |
| for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { |
| for (w2 = 0; w2 < ics->group_len[w]; w2++) { |
| for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--) |
| ; |
| maxsfb = FFMAX(maxsfb, cmaxsfb); |
| } |
| } |
| 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; |
| } |
| } |
| |
| static void apply_intensity_stereo(ChannelElement *cpe) |
| { |
| int w, w2, g, i; |
| IndividualChannelStream *ics = &cpe->ch[0].ics; |
| if (!cpe->common_window) |
| return; |
| for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { |
| for (w2 = 0; w2 < ics->group_len[w]; w2++) { |
| int start = (w+w2) * 128; |
| for (g = 0; g < ics->num_swb; g++) { |
| int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14); |
| float scale = cpe->ch[0].is_ener[w*16+g]; |
| if (!cpe->is_mask[w*16 + g]) { |
| start += ics->swb_sizes[g]; |
| continue; |
| } |
| if (cpe->ms_mask[w*16 + g]) |
| p *= -1; |
| for (i = 0; i < ics->swb_sizes[g]; i++) { |
| float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale; |
| cpe->ch[0].coeffs[start+i] = sum; |
| cpe->ch[1].coeffs[start+i] = 0.0f; |
| } |
| start += ics->swb_sizes[g]; |
| } |
| } |
| } |
| } |
| |
| static void apply_mid_side_stereo(ChannelElement *cpe) |
| { |
| int w, w2, g, i; |
| IndividualChannelStream *ics = &cpe->ch[0].ics; |
| if (!cpe->common_window) |
| return; |
| for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { |
| for (w2 = 0; w2 < ics->group_len[w]; w2++) { |
| int start = (w+w2) * 128; |
| for (g = 0; g < ics->num_swb; g++) { |
| /* ms_mask can be used for other purposes in PNS and I/S, |
| * so must not apply M/S if any band uses either, even if |
| * ms_mask is set. |
| */ |
| if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g] |
| || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT |
| || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) { |
| start += ics->swb_sizes[g]; |
| continue; |
| } |
| for (i = 0; i < ics->swb_sizes[g]; i++) { |
| float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f; |
| float R = L - cpe->ch[1].coeffs[start+i]; |
| cpe->ch[0].coeffs[start+i] = L; |
| cpe->ch[1].coeffs[start+i] = R; |
| } |
| start += ics->swb_sizes[g]; |
| } |
| } |
| } |
| } |
| |
| /** |
| * Encode scalefactor band coding type. |
| */ |
| static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) |
| { |
| int w; |
| |
| if (s->coder->set_special_band_scalefactors) |
| s->coder->set_special_band_scalefactors(s, sce); |
| |
| for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) |
| s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); |
| } |
| |
| /** |
| * Encode scalefactors. |
| */ |
| static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, |
| SingleChannelElement *sce) |
| { |
| int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET; |
| int off_is = 0, noise_flag = 1; |
| int i, w; |
| |
| for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { |
| for (i = 0; i < sce->ics.max_sfb; i++) { |
| if (!sce->zeroes[w*16 + i]) { |
| if (sce->band_type[w*16 + i] == NOISE_BT) { |
| diff = sce->sf_idx[w*16 + i] - off_pns; |
| off_pns = sce->sf_idx[w*16 + i]; |
| if (noise_flag-- > 0) { |
| put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE); |
| continue; |
| } |
| } else if (sce->band_type[w*16 + i] == INTENSITY_BT || |
| sce->band_type[w*16 + i] == INTENSITY_BT2) { |
| diff = sce->sf_idx[w*16 + i] - off_is; |
| off_is = sce->sf_idx[w*16 + i]; |
| } else { |
| diff = sce->sf_idx[w*16 + i] - off_sf; |
| off_sf = sce->sf_idx[w*16 + i]; |
| } |
| diff += SCALE_DIFF_ZERO; |
| av_assert0(diff >= 0 && diff <= 120); |
| put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]); |
| } |
| } |
| } |
| } |
| |
| /** |
| * 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], |
| NULL, sce->ics.swb_sizes[i], |
| sce->sf_idx[w*16 + i], |
| sce->band_type[w*16 + i], |
| s->lambda, |
| sce->ics.window_clipping[w]); |
| } |
| start += sce->ics.swb_sizes[i]; |
| } |
| } |
| } |
| |
| /** |
| * Downscale spectral coefficients for near-clipping windows to avoid artifacts |
| */ |
| static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce) |
| { |
| int start, i, j, w; |
| |
| if (sce->ics.clip_avoidance_factor < 1.0f) { |
| for (w = 0; w < sce->ics.num_windows; w++) { |
| start = 0; |
| for (i = 0; i < sce->ics.max_sfb; i++) { |
| float *swb_coeffs = &sce->coeffs[start + w*128]; |
| for (j = 0; j < sce->ics.swb_sizes[i]; j++) |
| swb_coeffs[j] *= sce->ics.clip_avoidance_factor; |
| start += sce->ics.swb_sizes[i]; |
| } |
| } |
| } |
| } |
| |
| /** |
| * Encode one channel of audio data. |
| */ |
| static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, |
| SingleChannelElement *sce, |
| int common_window) |
| { |
| put_bits(&s->pb, 8, sce->sf_idx[0]); |
| if (!common_window) { |
| put_ics_info(s, &sce->ics); |
| if (s->coder->encode_main_pred) |
| s->coder->encode_main_pred(s, sce); |
| if (s->coder->encode_ltp_info) |
| s->coder->encode_ltp_info(s, sce, 0); |
| } |
| encode_band_info(s, sce); |
| encode_scale_factors(avctx, s, sce); |
| encode_pulses(s, &sce->pulse); |
| put_bits(&s->pb, 1, !!sce->tns.present); |
| if (s->coder->encode_tns_info) |
| s->coder->encode_tns_info(s, sce); |
| put_bits(&s->pb, 1, 0); //ssr |
| encode_spectral_coeffs(s, sce); |
| return 0; |
| } |
| |
| /** |
| * 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; |
| SingleChannelElement *sce; |
| IndividualChannelStream *ics; |
| int i, its, ch, w, chans, tag, start_ch, ret, frame_bits; |
| int target_bits, rate_bits, too_many_bits, too_few_bits; |
| int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0; |
| int chan_el_counter[4]; |
| 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++) { |
| int k; |
| float clip_avoidance_factor; |
| sce = &cpe->ch[ch]; |
| ics = &sce->ics; |
| s->cur_channel = start_ch + ch; |
| overlap = &samples[s->cur_channel][0]; |
| samples2 = overlap + 1024; |
| la = samples2 + (448+64); |
| if (!frame) |
| la = NULL; |
| if (tag == TYPE_LFE) { |
| wi[ch].window_type[0] = ONLY_LONG_SEQUENCE; |
| wi[ch].window_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, s->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]; |
| ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb); |
| ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ? |
| ff_swb_offset_128 [s->samplerate_index]: |
| ff_swb_offset_1024[s->samplerate_index]; |
| ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ? |
| ff_tns_max_bands_128 [s->samplerate_index]: |
| ff_tns_max_bands_1024[s->samplerate_index]; |
| clip_avoidance_factor = 0.0f; |
| for (w = 0; w < ics->num_windows; w++) |
| ics->group_len[w] = wi[ch].grouping[w]; |
| for (w = 0; w < ics->num_windows; w++) { |
| if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) { |
| ics->window_clipping[w] = 1; |
| clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]); |
| } else { |
| ics->window_clipping[w] = 0; |
| } |
| } |
| if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) { |
| ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor; |
| } else { |
| ics->clip_avoidance_factor = 1.0f; |
| } |
| |
| apply_window_and_mdct(s, sce, overlap); |
| |
| if (s->options.ltp && s->coder->update_ltp) { |
| s->coder->update_ltp(s, sce); |
| apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]); |
| s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf); |
| } |
| |
| for (k = 0; k < 1024; k++) { |
| if (!isfinite(cpe->ch[ch].coeffs[k])) { |
| av_log(avctx, AV_LOG_ERROR, "Input contains NaN/+-Inf\n"); |
| return AVERROR(EINVAL); |
| } |
| } |
| avoid_clipping(s, sce); |
| } |
| start_ch += chans; |
| } |
| if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0) |
| return ret; |
| frame_bits = its = 0; |
| do { |
| init_put_bits(&s->pb, avpkt->data, avpkt->size); |
| |
| if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT)) |
| put_bitstream_info(s, LIBAVCODEC_IDENT); |
| start_ch = 0; |
| target_bits = 0; |
| memset(chan_el_counter, 0, sizeof(chan_el_counter)); |
| for (i = 0; i < s->chan_map[0]; i++) { |
| FFPsyWindowInfo* wi = windows + start_ch; |
| const float *coeffs[2]; |
| tag = s->chan_map[i+1]; |
| chans = tag == TYPE_CPE ? 2 : 1; |
| cpe = &s->cpe[i]; |
| cpe->common_window = 0; |
| memset(cpe->is_mask, 0, sizeof(cpe->is_mask)); |
| memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask)); |
| put_bits(&s->pb, 3, tag); |
| put_bits(&s->pb, 4, chan_el_counter[tag]++); |
| for (ch = 0; ch < chans; ch++) { |
| sce = &cpe->ch[ch]; |
| coeffs[ch] = sce->coeffs; |
| sce->ics.predictor_present = 0; |
| sce->ics.ltp.present = 0; |
| memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used)); |
| memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used)); |
| memset(&sce->tns, 0, sizeof(TemporalNoiseShaping)); |
| for (w = 0; w < 128; w++) |
| if (sce->band_type[w] > RESERVED_BT) |
| sce->band_type[w] = 0; |
| } |
| s->psy.bitres.alloc = -1; |
| s->psy.bitres.bits = s->last_frame_pb_count / s->channels; |
| s->psy.model->analyze(&s->psy, start_ch, coeffs, wi); |
| if (s->psy.bitres.alloc > 0) { |
| /* Lambda unused here on purpose, we need to take psy's unscaled allocation */ |
| target_bits += s->psy.bitres.alloc |
| * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120)); |
| s->psy.bitres.alloc /= chans; |
| } |
| s->cur_type = tag; |
| for (ch = 0; ch < chans; ch++) { |
| s->cur_channel = start_ch + ch; |
| if (s->options.pns && s->coder->mark_pns) |
| s->coder->mark_pns(s, avctx, &cpe->ch[ch]); |
| s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda); |
| } |
| 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; |
| } |
| } |
| } |
| for (ch = 0; ch < chans; ch++) { /* TNS and PNS */ |
| sce = &cpe->ch[ch]; |
| s->cur_channel = start_ch + ch; |
| if (s->options.tns && s->coder->search_for_tns) |
| s->coder->search_for_tns(s, sce); |
| if (s->options.tns && s->coder->apply_tns_filt) |
| s->coder->apply_tns_filt(s, sce); |
| if (sce->tns.present) |
| tns_mode = 1; |
| if (s->options.pns && s->coder->search_for_pns) |
| s->coder->search_for_pns(s, avctx, sce); |
| } |
| s->cur_channel = start_ch; |
| if (s->options.intensity_stereo) { /* Intensity Stereo */ |
| if (s->coder->search_for_is) |
| s->coder->search_for_is(s, avctx, cpe); |
| if (cpe->is_mode) is_mode = 1; |
| apply_intensity_stereo(cpe); |
| } |
| if (s->options.pred) { /* Prediction */ |
| for (ch = 0; ch < chans; ch++) { |
| sce = &cpe->ch[ch]; |
| s->cur_channel = start_ch + ch; |
| if (s->options.pred && s->coder->search_for_pred) |
| s->coder->search_for_pred(s, sce); |
| if (cpe->ch[ch].ics.predictor_present) pred_mode = 1; |
| } |
| if (s->coder->adjust_common_pred) |
| s->coder->adjust_common_pred(s, cpe); |
| for (ch = 0; ch < chans; ch++) { |
| sce = &cpe->ch[ch]; |
| s->cur_channel = start_ch + ch; |
| if (s->options.pred && s->coder->apply_main_pred) |
| s->coder->apply_main_pred(s, sce); |
| } |
| s->cur_channel = start_ch; |
| } |
| if (s->options.mid_side) { /* Mid/Side stereo */ |
| if (s->options.mid_side == -1 && s->coder->search_for_ms) |
| s->coder->search_for_ms(s, cpe); |
| else if (cpe->common_window) |
| memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask)); |
| apply_mid_side_stereo(cpe); |
| } |
| adjust_frame_information(cpe, chans); |
| if (s->options.ltp) { /* LTP */ |
| for (ch = 0; ch < chans; ch++) { |
| sce = &cpe->ch[ch]; |
| s->cur_channel = start_ch + ch; |
| if (s->coder->search_for_ltp) |
| s->coder->search_for_ltp(s, sce, cpe->common_window); |
| if (sce->ics.ltp.present) pred_mode = 1; |
| } |
| s->cur_channel = start_ch; |
| if (s->coder->adjust_common_ltp) |
| s->coder->adjust_common_ltp(s, cpe); |
| } |
| if (chans == 2) { |
| put_bits(&s->pb, 1, cpe->common_window); |
| if (cpe->common_window) { |
| put_ics_info(s, &cpe->ch[0].ics); |
| if (s->coder->encode_main_pred) |
| s->coder->encode_main_pred(s, &cpe->ch[0]); |
| if (s->coder->encode_ltp_info) |
| s->coder->encode_ltp_info(s, &cpe->ch[0], 1); |
| encode_ms_info(&s->pb, cpe); |
| if (cpe->ms_mode) ms_mode = 1; |
| } |
| } |
| for (ch = 0; ch < chans; ch++) { |
| s->cur_channel = start_ch + ch; |
| encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window); |
| } |
| start_ch += chans; |
| } |
| |
| if (avctx->flags & CODEC_FLAG_QSCALE) { |
| /* When using a constant Q-scale, don't mess with lambda */ |
| break; |
| } |
| |
| /* rate control stuff |
| * allow between the nominal bitrate, and what psy's bit reservoir says to target |
| * but drift towards the nominal bitrate always |
| */ |
| frame_bits = put_bits_count(&s->pb); |
| rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate; |
| rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3); |
| too_many_bits = FFMAX(target_bits, rate_bits); |
| too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3); |
| too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits); |
| |
| /* When using ABR, be strict (but only for increasing) */ |
| too_few_bits = too_few_bits - too_few_bits/8; |
| too_many_bits = too_many_bits + too_many_bits/2; |
| |
| if ( its == 0 /* for steady-state Q-scale tracking */ |
| || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits)) |
| || frame_bits >= 6144 * s->channels - 3 ) |
| { |
| float ratio = ((float)rate_bits) / frame_bits; |
| |
| if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) { |
| /* |
| * This path is for steady-state Q-scale tracking |
| * When frame bits fall within the stable range, we still need to adjust |
| * lambda to maintain it like so in a stable fashion (large jumps in lambda |
| * create artifacts and should be avoided), but slowly |
| */ |
| ratio = sqrtf(sqrtf(ratio)); |
| ratio = av_clipf(ratio, 0.9f, 1.1f); |
| } else { |
| /* Not so fast though */ |
| ratio = sqrtf(ratio); |
| } |
| s->lambda = FFMIN(s->lambda * ratio, 65536.f); |
| |
| /* Keep iterating if we must reduce and lambda is in the sky */ |
| if (ratio > 0.9f && ratio < 1.1f) { |
| break; |
| } else { |
| if (is_mode || ms_mode || tns_mode || pred_mode) { |
| for (i = 0; i < s->chan_map[0]; i++) { |
| // Must restore coeffs |
| chans = tag == TYPE_CPE ? 2 : 1; |
| cpe = &s->cpe[i]; |
| for (ch = 0; ch < chans; ch++) |
| memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs)); |
| } |
| } |
| its++; |
| } |
| } else { |
| break; |
| } |
| } while (1); |
| |
| if (s->options.ltp && s->coder->ltp_insert_new_frame) |
| s->coder->ltp_insert_new_frame(s); |
| |
| put_bits(&s->pb, 3, TYPE_END); |
| flush_put_bits(&s->pb); |
| |
| s->last_frame_pb_count = put_bits_count(&s->pb); |
| |
| s->lambda_sum += s->lambda; |
| s->lambda_count++; |
| |
| 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; |
| |
| av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count); |
| |
| ff_mdct_end(&s->mdct1024); |
| ff_mdct_end(&s->mdct128); |
| ff_psy_end(&s->psy); |
| ff_lpc_end(&s->lpc); |
| if (s->psypp) |
| ff_psy_preprocess_end(s->psypp); |
| av_freep(&s->buffer.samples); |
| av_freep(&s->cpe); |
| av_freep(&s->fdsp); |
| ff_af_queue_close(&s->afq); |
| return 0; |
| } |
| |
| static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) |
| { |
| int ret = 0; |
| |
| s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); |
| if (!s->fdsp) |
| return AVERROR(ENOMEM); |
| |
| // window init |
| ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); |
| 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)) < 0) |
| return ret; |
| if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s) |
| { |
| int ch; |
| FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail); |
| FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail); |
| FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail); |
| |
| for(ch = 0; ch < s->channels; ch++) |
| s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; |
| |
| return 0; |
| alloc_fail: |
| return AVERROR(ENOMEM); |
| } |
| |
| static av_cold void aac_encode_init_tables(void) |
| { |
| ff_aac_tableinit(); |
| } |
| |
| static av_cold int aac_encode_init(AVCodecContext *avctx) |
| { |
| AACEncContext *s = avctx->priv_data; |
| int i, ret = 0; |
| const uint8_t *sizes[2]; |
| uint8_t grouping[AAC_MAX_CHANNELS]; |
| int lengths[2]; |
| |
| /* Constants */ |
| s->last_frame_pb_count = 0; |
| avctx->extradata_size = 5; |
| avctx->frame_size = 1024; |
| avctx->initial_padding = 1024; |
| s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; |
| |
| /* Channel map and unspecified bitrate guessing */ |
| s->channels = avctx->channels; |
| ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7, |
| "Unsupported number of channels: %d\n", s->channels); |
| s->chan_map = aac_chan_configs[s->channels-1]; |
| if (!avctx->bit_rate) { |
| for (i = 1; i <= s->chan_map[0]; i++) { |
| avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */ |
| s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */ |
| 69000 ; /* SCE */ |
| } |
| } |
| |
| /* Samplerate */ |
| for (i = 0; i < 16; i++) |
| if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) |
| break; |
| s->samplerate_index = i; |
| ERROR_IF(s->samplerate_index == 16 || |
| s->samplerate_index >= ff_aac_swb_size_1024_len || |
| s->samplerate_index >= ff_aac_swb_size_128_len, |
| "Unsupported sample rate %d\n", avctx->sample_rate); |
| |
| /* Bitrate limiting */ |
| WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, |
| "Too many bits %f > %d per frame requested, clamping to max\n", |
| 1024.0 * avctx->bit_rate / avctx->sample_rate, |
| 6144 * s->channels); |
| avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate, |
| avctx->bit_rate); |
| |
| /* Profile and option setting */ |
| avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW : |
| avctx->profile; |
| for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++) |
| if (avctx->profile == aacenc_profiles[i]) |
| break; |
| if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) { |
| avctx->profile = FF_PROFILE_AAC_LOW; |
| ERROR_IF(s->options.pred, |
| "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n"); |
| ERROR_IF(s->options.ltp, |
| "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n"); |
| WARN_IF(s->options.pns, |
| "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n"); |
| s->options.pns = 0; |
| } else if (avctx->profile == FF_PROFILE_AAC_LTP) { |
| s->options.ltp = 1; |
| ERROR_IF(s->options.pred, |
| "Main prediction unavailable in the \"aac_ltp\" profile\n"); |
| } else if (avctx->profile == FF_PROFILE_AAC_MAIN) { |
| s->options.pred = 1; |
| ERROR_IF(s->options.ltp, |
| "LTP prediction unavailable in the \"aac_main\" profile\n"); |
| } else if (s->options.ltp) { |
| avctx->profile = FF_PROFILE_AAC_LTP; |
| WARN_IF(1, |
| "Chainging profile to \"aac_ltp\"\n"); |
| ERROR_IF(s->options.pred, |
| "Main prediction unavailable in the \"aac_ltp\" profile\n"); |
| } else if (s->options.pred) { |
| avctx->profile = FF_PROFILE_AAC_MAIN; |
| WARN_IF(1, |
| "Chainging profile to \"aac_main\"\n"); |
| ERROR_IF(s->options.ltp, |
| "LTP prediction unavailable in the \"aac_main\" profile\n"); |
| } |
| s->profile = avctx->profile; |
| |
| /* Coder limitations */ |
| s->coder = &ff_aac_coders[s->options.coder]; |
| if (s->options.coder != AAC_CODER_TWOLOOP) { |
| ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, |
| "Coders other than twoloop require -strict -2 and some may be removed in the future\n"); |
| s->options.intensity_stereo = 0; |
| s->options.pns = 0; |
| } |
| ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, |
| "The LPT profile requires experimental compliance, add -strict -2 to enable!\n"); |
| |
| /* M/S introduces horrible artifacts with multichannel files, this is temporary */ |
| if (s->channels > 3) |
| s->options.mid_side = 0; |
| |
| if ((ret = dsp_init(avctx, s)) < 0) |
| goto fail; |
| |
| if ((ret = alloc_buffers(avctx, s)) < 0) |
| goto fail; |
| |
| put_audio_specific_config(avctx); |
| |
| sizes[0] = ff_aac_swb_size_1024[s->samplerate_index]; |
| sizes[1] = ff_aac_swb_size_128[s->samplerate_index]; |
| lengths[0] = ff_aac_num_swb_1024[s->samplerate_index]; |
| lengths[1] = ff_aac_num_swb_128[s->samplerate_index]; |
| for (i = 0; i < s->chan_map[0]; i++) |
| grouping[i] = s->chan_map[i + 1] == TYPE_CPE; |
| if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, |
| s->chan_map[0], grouping)) < 0) |
| goto fail; |
| s->psypp = ff_psy_preprocess_init(avctx); |
| ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON); |
| av_lfg_init(&s->lfg, 0x72adca55); |
| |
| if (HAVE_MIPSDSP) |
| ff_aac_coder_init_mips(s); |
| |
| if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0) |
| return AVERROR_UNKNOWN; |
| |
| 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[] = { |
| {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"}, |
| {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, |
| {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, |
| {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"}, |
| {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS}, |
| {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, |
| {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, |
| {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS}, |
| {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, |
| {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, |
| {NULL} |
| }; |
| |
| static const AVClass aacenc_class = { |
| "AAC encoder", |
| av_default_item_name, |
| aacenc_options, |
| LIBAVUTIL_VERSION_INT, |
| }; |
| |
| static const AVCodecDefault aac_encode_defaults[] = { |
| { "b", "0" }, |
| { NULL } |
| }; |
| |
| AVCodec ff_aac_encoder = { |
| .name = "aac", |
| .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), |
| .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, |
| .defaults = aac_encode_defaults, |
| .supported_samplerates = mpeg4audio_sample_rates, |
| .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
| .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY, |
| .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, |
| AV_SAMPLE_FMT_NONE }, |
| .priv_class = &aacenc_class, |
| }; |
