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01ecb7172b
This finalizes merging of the work in the patches in ticket #2686. Improvements to twoloop and RC logic are extensive. The non-exhaustive list of twoloop improvments includes: - Tweaks to distortion limits on the RD optimization phase of twoloop - Deeper search in twoloop - PNS information marking to let twoloop decide when to use it (turned out having the decision made separately wasn't working) - Tonal band detection and priorization - Better band energy conservation rules - Strict hole avoidance For rate control: - Use psymodel's bit allocation to allow proper use of the bit reservoir. Don't work against the bit reservoir by moving lambda in the opposite direction when psymodel decides to allocate more/less bits to a frame. - Retry the encode if the effective rate lies outside a reasonable margin of psymodel's allocation or the selected ABR. - Log average lambda at the end. Useful info for everyone, but especially for tuning of the various encoder constants that relate to lambda feedback. Psy: - Do not apply lowpass with a FIR filter, instead just let the coder zero bands above the cutoff. The FIR filter induces group delay, and while zeroing bands causes ripple, it's lost in the quantization noise. - Experimental VBR bit allocation code - Tweak automatic lowpass filter threshold to maximize audio bandwidth at all bitrates while still providing acceptable, stable quality. I/S: - Phase decision fixes. Unrelated to #2686, but the bugs only surfaced when the merge was finalized. Measure I/S band energy accounting for phase, and prevent I/S and M/S from being applied both. PNS: - Avoid marking short bands with PNS when they're part of a window group in which there's a large variation of energy from one window to the next. PNS can't preserve those and the effect is extremely noticeable. M/S: - Implement BMLD protection similar to the specified in ISO-IEC/13818:7-2003, Appendix C Section 6.1. Since M/S decision doesn't conform to section 6.1, a different method had to be implemented, but should provide equivalent protection. - Move the decision logic closer to the method specified in ISO-IEC/13818:7-2003, Appendix C Section 6.1. Specifically, make sure M/S needs less bits than dual stereo. - Don't apply M/S in bands that are using I/S Now, this of course needed adjustments in the compare targets and fuzz factors of the AAC encoder's fate tests, but if wondering why the targets go up (more distortion), consider the previous coder was using too many bits on LF content (far more than required by psy), and thus those signals will now be more distorted, not less. The extra distortion isn't audible though, I carried extensive ABX testing to make sure. A very similar patch was also extensively tested by Kamendo2 in the context of #2686.
969 lines
36 KiB
C
969 lines
36 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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/**
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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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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, s->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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#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],
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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) {
|
|
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.stereo_mode) { /* Mid/Side stereo */
|
|
if (s->options.stereo_mode == -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;
|
|
int i, ret = 0;
|
|
const uint8_t *sizes[2];
|
|
uint8_t grouping[AAC_MAX_CHANNELS];
|
|
int lengths[2];
|
|
|
|
avctx->frame_size = 1024;
|
|
|
|
for (i = 0; i < 16; i++)
|
|
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
|
|
break;
|
|
|
|
s->channels = avctx->channels;
|
|
|
|
ERROR_IF(i == 16 || i >= ff_aac_swb_size_1024_len || i >= ff_aac_swb_size_128_len,
|
|
"Unsupported sample rate %d\n", avctx->sample_rate);
|
|
ERROR_IF(s->channels > AAC_MAX_CHANNELS,
|
|
"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");
|
|
if (avctx->profile == FF_PROFILE_AAC_MAIN) {
|
|
s->options.pred = 1;
|
|
} else if ((avctx->profile == FF_PROFILE_AAC_LOW ||
|
|
avctx->profile == FF_PROFILE_UNKNOWN) && s->options.pred) {
|
|
s->profile = 0; /* Main */
|
|
WARN_IF(1, "Prediction requested, changing profile to AAC-Main\n");
|
|
} else if (avctx->profile == FF_PROFILE_AAC_LOW ||
|
|
avctx->profile == FF_PROFILE_UNKNOWN) {
|
|
s->profile = 1; /* Low */
|
|
} else {
|
|
ERROR_IF(1, "Unsupported profile %d\n", avctx->profile);
|
|
}
|
|
|
|
if (s->options.aac_coder != AAC_CODER_TWOLOOP) {
|
|
s->options.intensity_stereo = 0;
|
|
s->options.pns = 0;
|
|
}
|
|
|
|
avctx->bit_rate = (int)FFMIN(
|
|
6144 * s->channels / 1024.0 * avctx->sample_rate,
|
|
avctx->bit_rate);
|
|
|
|
s->samplerate_index = i;
|
|
|
|
s->chan_map = aac_chan_configs[s->channels-1];
|
|
|
|
if ((ret = dsp_init(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
if ((ret = alloc_buffers(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
avctx->extradata_size = 5;
|
|
put_audio_specific_config(avctx);
|
|
|
|
sizes[0] = ff_aac_swb_size_1024[i];
|
|
sizes[1] = ff_aac_swb_size_128[i];
|
|
lengths[0] = ff_aac_num_swb_1024[i];
|
|
lengths[1] = ff_aac_num_swb_128[i];
|
|
for (i = 0; i < s->chan_map[0]; i++)
|
|
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
|
|
if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
|
|
s->chan_map[0], grouping)) < 0)
|
|
goto fail;
|
|
s->psypp = ff_psy_preprocess_init(avctx);
|
|
s->coder = &ff_aac_coders[s->options.aac_coder];
|
|
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);
|
|
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
s->random_state = 0x1f2e3d4c;
|
|
|
|
ff_aac_tableinit();
|
|
|
|
avctx->initial_padding = 1024;
|
|
ff_af_queue_init(avctx, &s->afq);
|
|
|
|
return 0;
|
|
fail:
|
|
aac_encode_end(avctx);
|
|
return ret;
|
|
}
|
|
|
|
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
|
|
static const AVOption aacenc_options[] = {
|
|
{"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
|
|
{"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
|
|
{"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, AACENC_FLAGS},
|
|
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, 0, 1, AACENC_FLAGS},
|
|
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 1, AACENC_FLAGS},
|
|
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 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,
|
|
};
|