mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
d2ae5f77c6
Performance improvements: quant_bands: with: 681 decicycles in quant_bands, 8388453 runs, 155 skips without: 1190 decicycles in quant_bands, 8388386 runs, 222 skips Around 42% for the function Twoloop coder: abs_pow34: with/without: 7.82s/8.17s Around 4% for the entire encoder Both: with/without: 7.15s/8.17s Around 12% for the entire encoder Fast coder: abs_pow34: with/without: 3.40s/3.77s Around 10% for the entire encoder Both: with/without: 3.02s/3.77s Around 20% faster for the entire encoder Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com> Tested-by: Michael Niedermayer <michael@niedermayer.cc> Reviewed-by: James Almer <jamrial@gmail.com>
1100 lines
41 KiB
C
1100 lines
41 KiB
C
/*
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* AAC encoder
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* Copyright (C) 2008 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* AAC encoder
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*/
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/***********************************
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* TODOs:
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* add sane pulse detection
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***********************************/
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#include "libavutil/libm.h"
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#include "libavutil/thread.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "internal.h"
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#include "mpeg4audio.h"
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#include "kbdwin.h"
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#include "sinewin.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacenc.h"
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#include "aacenctab.h"
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#include "aacenc_utils.h"
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#include "psymodel.h"
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static AVOnce aac_table_init = AV_ONCE_INIT;
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/**
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* Make AAC audio config object.
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* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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*/
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static void put_audio_specific_config(AVCodecContext *avctx)
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{
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PutBitContext pb;
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AACEncContext *s = avctx->priv_data;
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int channels = s->channels - (s->channels == 8 ? 1 : 0);
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init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
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put_bits(&pb, 5, s->profile+1); //profile
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put_bits(&pb, 4, s->samplerate_index); //sample rate index
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put_bits(&pb, 4, channels);
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//GASpecificConfig
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put_bits(&pb, 1, 0); //frame length - 1024 samples
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put_bits(&pb, 1, 0); //does not depend on core coder
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put_bits(&pb, 1, 0); //is not extension
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//Explicitly Mark SBR absent
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put_bits(&pb, 11, 0x2b7); //sync extension
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put_bits(&pb, 5, AOT_SBR);
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put_bits(&pb, 1, 0);
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flush_put_bits(&pb);
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}
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void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
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{
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++s->quantize_band_cost_cache_generation;
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if (s->quantize_band_cost_cache_generation == 0) {
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memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
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s->quantize_band_cost_cache_generation = 1;
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}
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}
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#define WINDOW_FUNC(type) \
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static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
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SingleChannelElement *sce, \
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const float *audio)
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WINDOW_FUNC(only_long)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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float *out = sce->ret_buf;
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fdsp->vector_fmul (out, audio, lwindow, 1024);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
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}
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WINDOW_FUNC(long_start)
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{
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const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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fdsp->vector_fmul(out, audio, lwindow, 1024);
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memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
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memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
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}
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WINDOW_FUNC(long_stop)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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memset(out, 0, sizeof(out[0]) * 448);
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fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
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memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
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}
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WINDOW_FUNC(eight_short)
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{
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *in = audio + 448;
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float *out = sce->ret_buf;
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int w;
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for (w = 0; w < 8; w++) {
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fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
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out += 128;
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in += 128;
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fdsp->vector_fmul_reverse(out, in, swindow, 128);
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out += 128;
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}
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}
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static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
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SingleChannelElement *sce,
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const float *audio) = {
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[ONLY_LONG_SEQUENCE] = apply_only_long_window,
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[LONG_START_SEQUENCE] = apply_long_start_window,
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[EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
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[LONG_STOP_SEQUENCE] = apply_long_stop_window
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};
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static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
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float *audio)
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{
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int i;
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const float *output = sce->ret_buf;
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apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
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s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
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else
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for (i = 0; i < 1024; i += 128)
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s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
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memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
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memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
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}
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/**
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* Encode ics_info element.
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* @see Table 4.6 (syntax of ics_info)
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*/
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static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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{
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int w;
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put_bits(&s->pb, 1, 0); // ics_reserved bit
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put_bits(&s->pb, 2, info->window_sequence[0]);
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put_bits(&s->pb, 1, info->use_kb_window[0]);
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if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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put_bits(&s->pb, 6, info->max_sfb);
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put_bits(&s->pb, 1, !!info->predictor_present);
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} else {
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put_bits(&s->pb, 4, info->max_sfb);
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for (w = 1; w < 8; w++)
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put_bits(&s->pb, 1, !info->group_len[w]);
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}
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}
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/**
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* Encode MS data.
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* @see 4.6.8.1 "Joint Coding - M/S Stereo"
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*/
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static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
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{
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int i, w;
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put_bits(pb, 2, cpe->ms_mode);
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if (cpe->ms_mode == 1)
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for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
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for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
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put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
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}
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/**
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* Produce integer coefficients from scalefactors provided by the model.
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*/
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static void adjust_frame_information(ChannelElement *cpe, int chans)
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{
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int i, w, w2, g, ch;
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int maxsfb, cmaxsfb;
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for (ch = 0; ch < chans; ch++) {
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IndividualChannelStream *ics = &cpe->ch[ch].ics;
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maxsfb = 0;
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cpe->ch[ch].pulse.num_pulse = 0;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
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;
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maxsfb = FFMAX(maxsfb, cmaxsfb);
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}
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}
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ics->max_sfb = maxsfb;
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//adjust zero bands for window groups
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (g = 0; g < ics->max_sfb; g++) {
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i = 1;
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for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
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if (!cpe->ch[ch].zeroes[w2*16 + g]) {
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i = 0;
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break;
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}
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}
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cpe->ch[ch].zeroes[w*16 + g] = i;
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}
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}
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}
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if (chans > 1 && cpe->common_window) {
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IndividualChannelStream *ics0 = &cpe->ch[0].ics;
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IndividualChannelStream *ics1 = &cpe->ch[1].ics;
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int msc = 0;
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ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
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ics1->max_sfb = ics0->max_sfb;
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for (w = 0; w < ics0->num_windows*16; w += 16)
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for (i = 0; i < ics0->max_sfb; i++)
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if (cpe->ms_mask[w+i])
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msc++;
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if (msc == 0 || ics0->max_sfb == 0)
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cpe->ms_mode = 0;
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else
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cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
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}
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}
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static void apply_intensity_stereo(ChannelElement *cpe)
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{
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int w, w2, g, i;
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IndividualChannelStream *ics = &cpe->ch[0].ics;
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if (!cpe->common_window)
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return;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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int start = (w+w2) * 128;
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for (g = 0; g < ics->num_swb; g++) {
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int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
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float scale = cpe->ch[0].is_ener[w*16+g];
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if (!cpe->is_mask[w*16 + g]) {
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start += ics->swb_sizes[g];
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continue;
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}
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if (cpe->ms_mask[w*16 + g])
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p *= -1;
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
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cpe->ch[0].coeffs[start+i] = sum;
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cpe->ch[1].coeffs[start+i] = 0.0f;
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}
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start += ics->swb_sizes[g];
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}
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}
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}
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}
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static void apply_mid_side_stereo(ChannelElement *cpe)
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{
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int w, w2, g, i;
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IndividualChannelStream *ics = &cpe->ch[0].ics;
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if (!cpe->common_window)
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return;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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int start = (w+w2) * 128;
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for (g = 0; g < ics->num_swb; g++) {
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/* ms_mask can be used for other purposes in PNS and I/S,
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* so must not apply M/S if any band uses either, even if
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* ms_mask is set.
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*/
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if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
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|| cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
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|| cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
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start += ics->swb_sizes[g];
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continue;
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}
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
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float R = L - cpe->ch[1].coeffs[start+i];
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cpe->ch[0].coeffs[start+i] = L;
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cpe->ch[1].coeffs[start+i] = R;
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}
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start += ics->swb_sizes[g];
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}
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}
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}
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}
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/**
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* Encode scalefactor band coding type.
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*/
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static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
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{
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int w;
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if (s->coder->set_special_band_scalefactors)
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s->coder->set_special_band_scalefactors(s, sce);
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
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s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
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}
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/**
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* Encode scalefactors.
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*/
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static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
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SingleChannelElement *sce)
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{
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int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
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int off_is = 0, noise_flag = 1;
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int i, w;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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for (i = 0; i < sce->ics.max_sfb; i++) {
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if (!sce->zeroes[w*16 + i]) {
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if (sce->band_type[w*16 + i] == NOISE_BT) {
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diff = sce->sf_idx[w*16 + i] - off_pns;
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off_pns = sce->sf_idx[w*16 + i];
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if (noise_flag-- > 0) {
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put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
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continue;
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}
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} else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
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sce->band_type[w*16 + i] == INTENSITY_BT2) {
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diff = sce->sf_idx[w*16 + i] - off_is;
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off_is = sce->sf_idx[w*16 + i];
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} else {
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diff = sce->sf_idx[w*16 + i] - off_sf;
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off_sf = sce->sf_idx[w*16 + i];
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}
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diff += SCALE_DIFF_ZERO;
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av_assert0(diff >= 0 && diff <= 120);
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put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
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}
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}
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}
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}
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/**
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* Encode pulse data.
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*/
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static void encode_pulses(AACEncContext *s, Pulse *pulse)
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{
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int i;
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put_bits(&s->pb, 1, !!pulse->num_pulse);
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if (!pulse->num_pulse)
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return;
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put_bits(&s->pb, 2, pulse->num_pulse - 1);
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put_bits(&s->pb, 6, pulse->start);
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for (i = 0; i < pulse->num_pulse; i++) {
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put_bits(&s->pb, 5, pulse->pos[i]);
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put_bits(&s->pb, 4, pulse->amp[i]);
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}
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}
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/**
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* Encode spectral coefficients processed by psychoacoustic model.
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*/
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static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
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{
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int start, i, w, w2;
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for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
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start = 0;
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for (i = 0; i < sce->ics.max_sfb; i++) {
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if (sce->zeroes[w*16 + i]) {
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start += sce->ics.swb_sizes[i];
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continue;
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}
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for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
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s->coder->quantize_and_encode_band(s, &s->pb,
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&sce->coeffs[start + w2*128],
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NULL, sce->ics.swb_sizes[i],
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sce->sf_idx[w*16 + i],
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sce->band_type[w*16 + i],
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s->lambda,
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sce->ics.window_clipping[w]);
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}
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start += sce->ics.swb_sizes[i];
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}
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}
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}
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/**
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* Downscale spectral coefficients for near-clipping windows to avoid artifacts
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*/
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static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
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{
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int start, i, j, w;
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if (sce->ics.clip_avoidance_factor < 1.0f) {
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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] = 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 & 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_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;
|
|
|
|
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);
|
|
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_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,
|
|
};
|