0f4334df45
This commit adds the ability for a profile to set the default options, as well as for the user to override such options by simply stating them in the command line while still keeping the same profile, as long as those options are still permitted by the profile. Example: setting the profile to aac_low (the default) will turn PNS and IS on. They can be disabled by -aac_pns 0 and -aac_is 0, respectively. Turning on -aac_pred 1 will cause the profile to be elevated to aac_main, as long as no options forbidding aac_main have been entered (like AAC-LTP, which will be pushed soon). A useful feature is that by setting the profile to mpeg2_aac_low, all MPEG4 features will be disabled and if the user tries to enable them then the program will exit with an error. This profile is signalled with the same bitstream as aac_low (MPEG4) but some devices and decoders will fail if any MPEG4 features have been enabled.
1021 lines
37 KiB
C
1021 lines
37 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/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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struct AACProfileOptions {
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int profile;
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struct AACEncOptions opts;
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};
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/**
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* List of currently supported profiles, anything not listed isn't supported.
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*/
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static const struct AACProfileOptions aacenc_profiles[] = {
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{FF_PROFILE_AAC_MAIN,
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{ /* Main profile, all advanced encoding abilities enabled */
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.mid_side = 0,
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.pns = 1,
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.tns = 0,
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.pred = OPT_REQUIRED,
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.intensity_stereo = 1,
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},
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},
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{FF_PROFILE_AAC_LOW,
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{ /* Default profile, these are the settings that get set by default */
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.mid_side = 0,
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.pns = 1,
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.tns = 0,
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.pred = OPT_NEEDS_MAIN,
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.intensity_stereo = 1,
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},
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},
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{FF_PROFILE_MPEG2_AAC_LOW,
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{ /* Strict MPEG 2 Part 7 compliance profile */
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.mid_side = 0,
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.pns = OPT_BANNED,
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.tns = 0,
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.pred = OPT_BANNED,
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.intensity_stereo = 1,
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},
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},
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};
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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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int sf, g;
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for (sf = 0; sf < 256; sf++) {
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for (g = 0; g < 128; g++) {
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s->quantize_band_cost_cache[sf][g].bits = -1;
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}
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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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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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if (!cpe->ms_mask[w*16 + g] && !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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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],
|
|
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);
|
|
}
|
|
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;
|
|
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++) {
|
|
IndividualChannelStream *ics = &cpe->ch[ch].ics;
|
|
int cur_channel = start_ch + ch;
|
|
float clip_avoidance_factor;
|
|
overlap = &samples[cur_channel][0];
|
|
samples2 = overlap + 1024;
|
|
la = samples2 + (448+64);
|
|
if (!frame)
|
|
la = NULL;
|
|
if (tag == TYPE_LFE) {
|
|
wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
|
|
wi[ch].window_shape = 0;
|
|
wi[ch].num_windows = 1;
|
|
wi[ch].grouping[0] = 1;
|
|
|
|
/* Only the lowest 12 coefficients are used in a LFE channel.
|
|
* The expression below results in only the bottom 8 coefficients
|
|
* being used for 11.025kHz to 16kHz sample rates.
|
|
*/
|
|
ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
|
|
} else {
|
|
wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
|
|
ics->window_sequence[0]);
|
|
}
|
|
ics->window_sequence[1] = ics->window_sequence[0];
|
|
ics->window_sequence[0] = wi[ch].window_type[0];
|
|
ics->use_kb_window[1] = ics->use_kb_window[0];
|
|
ics->use_kb_window[0] = wi[ch].window_shape;
|
|
ics->num_windows = wi[ch].num_windows;
|
|
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
|
|
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
|
|
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, &cpe->ch[ch], overlap);
|
|
if (isnan(cpe->ch->coeffs[0])) {
|
|
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
avoid_clipping(s, &cpe->ch[ch]);
|
|
}
|
|
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;
|
|
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 = avctx->frame_bits / 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.pns && s->coder->search_for_pns)
|
|
s->coder->search_for_pns(s, avctx, sce);
|
|
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;
|
|
}
|
|
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_prediction)
|
|
s->coder->adjust_common_prediction(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 (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]);
|
|
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 ((s->lambda < 300.f || ratio > 0.9f) && (s->lambda > 10.f || 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);
|
|
|
|
put_bits(&s->pb, 3, TYPE_END);
|
|
flush_put_bits(&s->pb);
|
|
avctx->frame_bits = 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 int aac_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
const AACEncOptions *p_opt = NULL;
|
|
int i, ret = 0;
|
|
const uint8_t *sizes[2];
|
|
uint8_t grouping[AAC_MAX_CHANNELS];
|
|
int lengths[2];
|
|
|
|
s->channels = avctx->channels;
|
|
s->chan_map = aac_chan_configs[s->channels-1];
|
|
s->random_state = 0x1f2e3d4c;
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
avctx->extradata_size = 5;
|
|
avctx->frame_size = 1024;
|
|
avctx->initial_padding = 1024;
|
|
avctx->bit_rate = (int)FFMIN(
|
|
6144 * s->channels / 1024.0 * avctx->sample_rate,
|
|
avctx->bit_rate);
|
|
avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
|
|
avctx->profile;
|
|
|
|
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);
|
|
ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
|
|
"Unsupported number of channels: %d\n", s->channels);
|
|
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
|
|
"Too many bits per frame requested, clamping to max\n");
|
|
|
|
for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++) {
|
|
if (avctx->profile == aacenc_profiles[i].profile) {
|
|
p_opt = &aacenc_profiles[i].opts;
|
|
break;
|
|
}
|
|
}
|
|
ERROR_IF(!p_opt, "Unsupported encoding profile: %d\n", avctx->profile);
|
|
AAC_OPT_SET(&s->options, p_opt, 1, coder);
|
|
AAC_OPT_SET(&s->options, p_opt, 0, pns);
|
|
AAC_OPT_SET(&s->options, p_opt, 0, tns);
|
|
AAC_OPT_SET(&s->options, p_opt, 0, pred);
|
|
AAC_OPT_SET(&s->options, p_opt, 1, mid_side);
|
|
AAC_OPT_SET(&s->options, p_opt, 0, intensity_stereo);
|
|
if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW)
|
|
s->profile = FF_PROFILE_AAC_LOW;
|
|
else
|
|
s->profile = avctx->profile;
|
|
s->coder = &ff_aac_coders[s->options.coder];
|
|
|
|
if (s->options.coder != AAC_CODER_TWOLOOP) {
|
|
s->options.intensity_stereo = 0;
|
|
s->options.pns = 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);
|
|
|
|
if (HAVE_MIPSDSPR1)
|
|
ff_aac_coder_init_mips(s);
|
|
|
|
ff_aac_tableinit();
|
|
|
|
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}, -1, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
|
|
{"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, 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 = 0}, -1, 1, AACENC_FLAGS},
|
|
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
|
|
{"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
|
|
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
|
|
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = OPT_AUTO}, -1, 1, AACENC_FLAGS},
|
|
{NULL}
|
|
};
|
|
|
|
static const AVClass aacenc_class = {
|
|
"AAC encoder",
|
|
av_default_item_name,
|
|
aacenc_options,
|
|
LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
AVCodec ff_aac_encoder = {
|
|
.name = "aac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_AAC,
|
|
.priv_data_size = sizeof(AACEncContext),
|
|
.init = aac_encode_init,
|
|
.encode2 = aac_encode_frame,
|
|
.close = aac_encode_end,
|
|
.supported_samplerates = mpeg4audio_sample_rates,
|
|
.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
|
|
AV_CODEC_CAP_EXPERIMENTAL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.priv_class = &aacenc_class,
|
|
};
|