/* * 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 #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 "profiles.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; static void put_pce(PutBitContext *pb, AVCodecContext *avctx) { int i, j; AACEncContext *s = avctx->priv_data; AACPCEInfo *pce = &s->pce; const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT; const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT; put_bits(pb, 4, 0); put_bits(pb, 2, avctx->profile); put_bits(pb, 4, s->samplerate_index); put_bits(pb, 4, pce->num_ele[0]); /* Front */ put_bits(pb, 4, pce->num_ele[1]); /* Side */ put_bits(pb, 4, pce->num_ele[2]); /* Back */ put_bits(pb, 2, pce->num_ele[3]); /* LFE */ put_bits(pb, 3, 0); /* Assoc data */ put_bits(pb, 4, 0); /* CCs */ put_bits(pb, 1, 0); /* Stereo mixdown */ put_bits(pb, 1, 0); /* Mono mixdown */ put_bits(pb, 1, 0); /* Something else */ for (i = 0; i < 4; i++) { for (j = 0; j < pce->num_ele[i]; j++) { if (i < 3) put_bits(pb, 1, pce->pairing[i][j]); put_bits(pb, 4, pce->index[i][j]); } } avpriv_align_put_bits(pb); put_bits(pb, 8, strlen(aux_data)); avpriv_put_string(pb, aux_data, 0); } /** * Make AAC audio config object. * @see 1.6.2.1 "Syntax - AudioSpecificConfig" */ static int put_audio_specific_config(AVCodecContext *avctx) { PutBitContext pb; AACEncContext *s = avctx->priv_data; int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0)); const int max_size = 32; avctx->extradata = av_mallocz(max_size); if (!avctx->extradata) return AVERROR(ENOMEM); init_put_bits(&pb, avctx->extradata, max_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 if (s->needs_pce) put_pce(&pb, avctx); //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); avctx->extradata_size = put_bits_count(&pb) >> 3; return 0; } void ff_quantize_band_cost_cache_init(struct AACEncContext *s) { ++s->quantize_band_cost_cache_generation; if (s->quantize_band_cost_cache_generation == 0) { memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache)); s->quantize_band_cost_cache_generation = 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 = s->reorder_map; /* 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]; /* add current frame to queue */ if (frame) { if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) return ret; } else { if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count)) return 0; } 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] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE; wi[ch].window_shape = 0; wi[ch].num_windows = 1; wi[ch].grouping[0] = 1; wi[ch].clipping[0] = 0; /* 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]; for (w = 0; w < ics->num_windows; w++) ics->group_len[w] = wi[ch].grouping[w]; /* Calculate input sample maximums and evaluate clipping risk */ clip_avoidance_factor = 0.0f; for (w = 0; w < ics->num_windows; w++) { const float *wbuf = overlap + w * 128; const int wlen = 2048 / ics->num_windows; float max = 0; int j; /* mdct input is 2 * output */ for (j = 0; j < wlen; j++) max = FFMAX(max, fabsf(wbuf[j])); wi[ch].clipping[w] = max; } 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 (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation av_log(avctx, AV_LOG_ERROR, "Input contains (near) 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 & AV_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 = av_clipf(s->lambda * ratio, FLT_EPSILON, 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++; 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_count ? s->lambda_sum / s->lambda_count : NAN); 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); 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->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; s->needs_pce = 1; for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) { if (avctx->channel_layout == aac_normal_chan_layouts[i]) { s->needs_pce = s->options.pce; break; } } if (s->needs_pce) { char buf[64]; for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++) if (avctx->channel_layout == aac_pce_configs[i].layout) break; av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout); ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf); av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf); s->pce = aac_pce_configs[i]; s->reorder_map = s->pce.reorder_map; s->chan_map = s->pce.config_map; } else { s->reorder_map = aac_chan_maps[s->channels - 1]; 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_ANMR) { ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, "The ANMR coder is considered experimental, add -strict -2 to enable!\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; if ((ret = put_audio_specific_config(avctx))) goto fail; 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); s->random_state = 0x1f2e3d4c; s->abs_pow34 = abs_pow34_v; s->quant_bands = quantize_bands; if (ARCH_X86) ff_aac_dsp_init_x86(s); 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_FAST}, 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", "Default fast search", 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}, {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, FF_AAC_PROFILE_OPTS {NULL} }; static const AVClass aacenc_class = { .class_name = "AAC encoder", .item_name = av_default_item_name, .option = aacenc_options, .version = 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, };