ffmpeg / libavcodec / aacdec.c @ 79ce1078
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/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-ffmpeg@jannau.net>
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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 decoder
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* @author Oded Shimon ( ods15 ods15 dyndns org )
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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*/
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/*
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* supported tools
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*
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* Support? Name
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* N (code in SoC repo) gain control
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* Y block switching
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* Y window shapes - standard
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* N window shapes - Low Delay
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* Y filterbank - standard
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* N (code in SoC repo) filterbank - Scalable Sample Rate
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* Y Temporal Noise Shaping
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* N (code in SoC repo) Long Term Prediction
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* Y intensity stereo
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* Y channel coupling
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* Y frequency domain prediction
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* Y Perceptual Noise Substitution
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* Y Mid/Side stereo
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* N Scalable Inverse AAC Quantization
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* N Frequency Selective Switch
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* N upsampling filter
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* Y quantization & coding - AAC
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* N quantization & coding - TwinVQ
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* N quantization & coding - BSAC
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* N AAC Error Resilience tools
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* N Error Resilience payload syntax
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* N Error Protection tool
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* N CELP
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* N Silence Compression
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* N HVXC
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* N HVXC 4kbits/s VR
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* N Structured Audio tools
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* N Structured Audio Sample Bank Format
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* N MIDI
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* N Harmonic and Individual Lines plus Noise
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* N Text-To-Speech Interface
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* Y Spectral Band Replication
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* Y (not in this code) Layer-1
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* Y (not in this code) Layer-2
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* Y (not in this code) Layer-3
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* N SinuSoidal Coding (Transient, Sinusoid, Noise)
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* Y Parametric Stereo
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* N Direct Stream Transfer
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*
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* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
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* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
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Parametric Stereo.
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*/
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#include "avcodec.h" |
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#include "internal.h" |
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#include "get_bits.h" |
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#include "dsputil.h" |
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#include "fft.h" |
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#include "lpc.h" |
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#include "aac.h" |
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#include "aactab.h" |
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#include "aacdectab.h" |
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#include "cbrt_tablegen.h" |
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#include "sbr.h" |
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#include "aacsbr.h" |
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#include "mpeg4audio.h" |
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#include "aacadtsdec.h" |
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#include <assert.h> |
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#include <errno.h> |
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#include <math.h> |
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#include <string.h> |
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#if ARCH_ARM
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# include "arm/aac.h" |
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#endif
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union float754 {
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float f;
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uint32_t i; |
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}; |
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static VLC vlc_scalefactors;
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static VLC vlc_spectral[11]; |
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static const char overread_err[] = "Input buffer exhausted before END element found\n"; |
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static ChannelElement *get_che(AACContext *ac, int type, int elem_id) |
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{ |
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// For PCE based channel configurations map the channels solely based on tags.
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if (!ac->m4ac.chan_config) {
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return ac->tag_che_map[type][elem_id];
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} |
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// For indexed channel configurations map the channels solely based on position.
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switch (ac->m4ac.chan_config) {
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case 7: |
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if (ac->tags_mapped == 3 && type == TYPE_CPE) { |
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ac->tags_mapped++; |
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; |
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} |
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case 6: |
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/* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
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instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
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encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
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if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { |
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ac->tags_mapped++; |
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return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; |
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} |
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case 5: |
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if (ac->tags_mapped == 2 && type == TYPE_CPE) { |
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ac->tags_mapped++; |
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; |
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} |
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case 4: |
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if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { |
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ac->tags_mapped++; |
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return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; |
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} |
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case 3: |
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case 2: |
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if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { |
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ac->tags_mapped++; |
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return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; |
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} else if (ac->m4ac.chan_config == 2) { |
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return NULL; |
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} |
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case 1: |
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if (!ac->tags_mapped && type == TYPE_SCE) {
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ac->tags_mapped++; |
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return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; |
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} |
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default:
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return NULL; |
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} |
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} |
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/**
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* Check for the channel element in the current channel position configuration.
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* If it exists, make sure the appropriate element is allocated and map the
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* channel order to match the internal FFmpeg channel layout.
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*
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* @param che_pos current channel position configuration
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* @param type channel element type
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* @param id channel element id
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* @param channels count of the number of channels in the configuration
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int che_configure(AACContext *ac, |
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enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
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int type, int id, |
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int *channels)
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{ |
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if (che_pos[type][id]) {
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if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) |
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return AVERROR(ENOMEM);
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ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr); |
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if (type != TYPE_CCE) {
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
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if (type == TYPE_CPE ||
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(type == TYPE_SCE && ac->m4ac.ps == 1)) {
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ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
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} |
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} |
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} else {
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if (ac->che[type][id])
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ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr); |
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av_freep(&ac->che[type][id]); |
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} |
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return 0; |
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} |
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/**
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* Configure output channel order based on the current program configuration element.
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*
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* @param che_pos current channel position configuration
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int output_configure(AACContext *ac, |
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enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
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int channel_config, enum OCStatus oc_type) |
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{ |
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AVCodecContext *avctx = ac->avctx; |
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int i, type, channels = 0, ret; |
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if (new_che_pos != che_pos)
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memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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if (channel_config) {
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for (i = 0; i < tags_per_config[channel_config]; i++) { |
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if ((ret = che_configure(ac, che_pos,
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aac_channel_layout_map[channel_config - 1][i][0], |
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aac_channel_layout_map[channel_config - 1][i][1], |
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&channels))) |
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return ret;
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} |
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memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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avctx->channel_layout = aac_channel_layout[channel_config - 1];
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} else {
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/* Allocate or free elements depending on if they are in the
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* current program configuration.
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*
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* Set up default 1:1 output mapping.
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*
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* For a 5.1 stream the output order will be:
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* [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
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*/
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for (i = 0; i < MAX_ELEM_ID; i++) { |
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for (type = 0; type < 4; type++) { |
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if ((ret = che_configure(ac, che_pos, type, i, &channels)))
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return ret;
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} |
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} |
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memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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avctx->channel_layout = 0;
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} |
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avctx->channels = channels; |
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ac->output_configured = oc_type; |
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return 0; |
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} |
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/**
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* Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
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*
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* @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
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* @param sce_map mono (Single Channel Element) map
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* @param type speaker type/position for these channels
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*/
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static void decode_channel_map(enum ChannelPosition *cpe_map, |
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enum ChannelPosition *sce_map,
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enum ChannelPosition type,
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GetBitContext *gb, int n)
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{ |
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while (n--) {
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enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map |
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map[get_bits(gb, 4)] = type;
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} |
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} |
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/**
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* Decode program configuration element; reference: table 4.2.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
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GetBitContext *gb) |
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{ |
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int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
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int comment_len;
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skip_bits(gb, 2); // object_type |
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sampling_index = get_bits(gb, 4);
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if (m4ac->sampling_index != sampling_index)
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av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
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num_front = get_bits(gb, 4);
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num_side = get_bits(gb, 4);
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num_back = get_bits(gb, 4);
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num_lfe = get_bits(gb, 2);
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num_assoc_data = get_bits(gb, 3);
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num_cc = get_bits(gb, 4);
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if (get_bits1(gb))
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skip_bits(gb, 4); // mono_mixdown_tag |
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if (get_bits1(gb))
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skip_bits(gb, 4); // stereo_mixdown_tag |
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if (get_bits1(gb))
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skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
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skip_bits_long(gb, 4 * num_assoc_data);
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decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
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align_get_bits(gb); |
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/* comment field, first byte is length */
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comment_len = get_bits(gb, 8) * 8; |
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if (get_bits_left(gb) < comment_len) {
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av_log(avctx, AV_LOG_ERROR, overread_err); |
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return -1; |
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} |
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skip_bits_long(gb, comment_len); |
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return 0; |
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} |
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/**
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* Set up channel positions based on a default channel configuration
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* as specified in table 1.17.
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*
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* @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static av_cold int set_default_channel_config(AVCodecContext *avctx, |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
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int channel_config)
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{ |
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if (channel_config < 1 || channel_config > 7) { |
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av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
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channel_config); |
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return -1; |
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} |
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/* default channel configurations:
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*
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* 1ch : front center (mono)
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* 2ch : L + R (stereo)
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* 3ch : front center + L + R
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* 4ch : front center + L + R + back center
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* 5ch : front center + L + R + back stereo
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* 6ch : front center + L + R + back stereo + LFE
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* 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
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*/
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if (channel_config != 2) |
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new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) |
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if (channel_config > 1) |
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new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) |
370 |
if (channel_config == 4) |
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new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center |
372 |
if (channel_config > 4) |
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new_che_pos[TYPE_CPE][(channel_config == 7) + 1] |
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= AAC_CHANNEL_BACK; // back stereo
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if (channel_config > 5) |
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new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE |
377 |
if (channel_config == 7) |
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new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right |
379 |
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return 0; |
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} |
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/**
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* Decode GA "General Audio" specific configuration; reference: table 4.1.
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*
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* @param ac pointer to AACContext, may be null
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* @param avctx pointer to AVCCodecContext, used for logging
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*
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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, |
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GetBitContext *gb, |
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MPEG4AudioConfig *m4ac, |
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int channel_config)
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{ |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
397 |
int extension_flag, ret;
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|
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if (get_bits1(gb)) { // frameLengthFlag |
400 |
av_log_missing_feature(avctx, "960/120 MDCT window is", 1); |
401 |
return -1; |
402 |
} |
403 |
|
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if (get_bits1(gb)) // dependsOnCoreCoder |
405 |
skip_bits(gb, 14); // coreCoderDelay |
406 |
extension_flag = get_bits1(gb); |
407 |
|
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if (m4ac->object_type == AOT_AAC_SCALABLE ||
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m4ac->object_type == AOT_ER_AAC_SCALABLE) |
410 |
skip_bits(gb, 3); // layerNr |
411 |
|
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memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
413 |
if (channel_config == 0) { |
414 |
skip_bits(gb, 4); // element_instance_tag |
415 |
if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
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return ret;
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} else {
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if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
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return ret;
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420 |
} |
421 |
if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
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return ret;
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|
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if (extension_flag) {
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switch (m4ac->object_type) {
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case AOT_ER_BSAC:
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skip_bits(gb, 5); // numOfSubFrame |
428 |
skip_bits(gb, 11); // layer_length |
429 |
break;
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case AOT_ER_AAC_LC:
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case AOT_ER_AAC_LTP:
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case AOT_ER_AAC_SCALABLE:
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case AOT_ER_AAC_LD:
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skip_bits(gb, 3); /* aacSectionDataResilienceFlag |
435 |
* aacScalefactorDataResilienceFlag
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* aacSpectralDataResilienceFlag
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*/
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break;
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} |
440 |
skip_bits1(gb); // extensionFlag3 (TBD in version 3)
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} |
442 |
return 0; |
443 |
} |
444 |
|
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/**
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* Decode audio specific configuration; reference: table 1.13.
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*
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* @param ac pointer to AACContext, may be null
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449 |
* @param avctx pointer to AVCCodecContext, used for logging
|
450 |
* @param m4ac pointer to MPEG4AudioConfig, used for parsing
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451 |
* @param data pointer to AVCodecContext extradata
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* @param data_size size of AVCCodecContext extradata
|
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*
|
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* @return Returns error status or number of consumed bits. <0 - error
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*/
|
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static int decode_audio_specific_config(AACContext *ac, |
457 |
AVCodecContext *avctx, |
458 |
MPEG4AudioConfig *m4ac, |
459 |
const uint8_t *data, int data_size) |
460 |
{ |
461 |
GetBitContext gb; |
462 |
int i;
|
463 |
|
464 |
init_get_bits(&gb, data, data_size * 8);
|
465 |
|
466 |
if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0) |
467 |
return -1; |
468 |
if (m4ac->sampling_index > 12) { |
469 |
av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
|
470 |
return -1; |
471 |
} |
472 |
if (m4ac->sbr == 1 && m4ac->ps == -1) |
473 |
m4ac->ps = 1;
|
474 |
|
475 |
skip_bits_long(&gb, i); |
476 |
|
477 |
switch (m4ac->object_type) {
|
478 |
case AOT_AAC_MAIN:
|
479 |
case AOT_AAC_LC:
|
480 |
if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
|
481 |
return -1; |
482 |
break;
|
483 |
default:
|
484 |
av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
|
485 |
m4ac->sbr == 1? "SBR+" : "", m4ac->object_type); |
486 |
return -1; |
487 |
} |
488 |
|
489 |
return get_bits_count(&gb);
|
490 |
} |
491 |
|
492 |
/**
|
493 |
* linear congruential pseudorandom number generator
|
494 |
*
|
495 |
* @param previous_val pointer to the current state of the generator
|
496 |
*
|
497 |
* @return Returns a 32-bit pseudorandom integer
|
498 |
*/
|
499 |
static av_always_inline int lcg_random(int previous_val) |
500 |
{ |
501 |
return previous_val * 1664525 + 1013904223; |
502 |
} |
503 |
|
504 |
static av_always_inline void reset_predict_state(PredictorState *ps) |
505 |
{ |
506 |
ps->r0 = 0.0f; |
507 |
ps->r1 = 0.0f; |
508 |
ps->cor0 = 0.0f; |
509 |
ps->cor1 = 0.0f; |
510 |
ps->var0 = 1.0f; |
511 |
ps->var1 = 1.0f; |
512 |
} |
513 |
|
514 |
static void reset_all_predictors(PredictorState *ps) |
515 |
{ |
516 |
int i;
|
517 |
for (i = 0; i < MAX_PREDICTORS; i++) |
518 |
reset_predict_state(&ps[i]); |
519 |
} |
520 |
|
521 |
static void reset_predictor_group(PredictorState *ps, int group_num) |
522 |
{ |
523 |
int i;
|
524 |
for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) |
525 |
reset_predict_state(&ps[i]); |
526 |
} |
527 |
|
528 |
#define AAC_INIT_VLC_STATIC(num, size) \
|
529 |
INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
|
530 |
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ |
531 |
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ |
532 |
size); |
533 |
|
534 |
static av_cold int aac_decode_init(AVCodecContext *avctx) |
535 |
{ |
536 |
AACContext *ac = avctx->priv_data; |
537 |
|
538 |
ac->avctx = avctx; |
539 |
ac->m4ac.sample_rate = avctx->sample_rate; |
540 |
|
541 |
if (avctx->extradata_size > 0) { |
542 |
if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
|
543 |
avctx->extradata, |
544 |
avctx->extradata_size) < 0)
|
545 |
return -1; |
546 |
} |
547 |
|
548 |
avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
549 |
|
550 |
AAC_INIT_VLC_STATIC( 0, 304); |
551 |
AAC_INIT_VLC_STATIC( 1, 270); |
552 |
AAC_INIT_VLC_STATIC( 2, 550); |
553 |
AAC_INIT_VLC_STATIC( 3, 300); |
554 |
AAC_INIT_VLC_STATIC( 4, 328); |
555 |
AAC_INIT_VLC_STATIC( 5, 294); |
556 |
AAC_INIT_VLC_STATIC( 6, 306); |
557 |
AAC_INIT_VLC_STATIC( 7, 268); |
558 |
AAC_INIT_VLC_STATIC( 8, 510); |
559 |
AAC_INIT_VLC_STATIC( 9, 366); |
560 |
AAC_INIT_VLC_STATIC(10, 462); |
561 |
|
562 |
ff_aac_sbr_init(); |
563 |
|
564 |
dsputil_init(&ac->dsp, avctx); |
565 |
|
566 |
ac->random_state = 0x1f2e3d4c;
|
567 |
|
568 |
// -1024 - Compensate wrong IMDCT method.
|
569 |
// 60 - Required to scale values to the correct range [-32768,32767]
|
570 |
// for float to int16 conversion. (1 << (60 / 4)) == 32768
|
571 |
ac->sf_scale = 1. / -1024.; |
572 |
ac->sf_offset = 60;
|
573 |
|
574 |
ff_aac_tableinit(); |
575 |
|
576 |
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
|
577 |
ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
578 |
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), |
579 |
352);
|
580 |
|
581 |
ff_mdct_init(&ac->mdct, 11, 1, 1.0); |
582 |
ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); |
583 |
// window initialization
|
584 |
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); |
585 |
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); |
586 |
ff_init_ff_sine_windows(10);
|
587 |
ff_init_ff_sine_windows( 7);
|
588 |
|
589 |
cbrt_tableinit(); |
590 |
|
591 |
return 0; |
592 |
} |
593 |
|
594 |
/**
|
595 |
* Skip data_stream_element; reference: table 4.10.
|
596 |
*/
|
597 |
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb) |
598 |
{ |
599 |
int byte_align = get_bits1(gb);
|
600 |
int count = get_bits(gb, 8); |
601 |
if (count == 255) |
602 |
count += get_bits(gb, 8);
|
603 |
if (byte_align)
|
604 |
align_get_bits(gb); |
605 |
|
606 |
if (get_bits_left(gb) < 8 * count) { |
607 |
av_log(ac->avctx, AV_LOG_ERROR, overread_err); |
608 |
return -1; |
609 |
} |
610 |
skip_bits_long(gb, 8 * count);
|
611 |
return 0; |
612 |
} |
613 |
|
614 |
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, |
615 |
GetBitContext *gb) |
616 |
{ |
617 |
int sfb;
|
618 |
if (get_bits1(gb)) {
|
619 |
ics->predictor_reset_group = get_bits(gb, 5);
|
620 |
if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
621 |
av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
|
622 |
return -1; |
623 |
} |
624 |
} |
625 |
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
626 |
ics->prediction_used[sfb] = get_bits1(gb); |
627 |
} |
628 |
return 0; |
629 |
} |
630 |
|
631 |
/**
|
632 |
* Decode Individual Channel Stream info; reference: table 4.6.
|
633 |
*
|
634 |
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
635 |
*/
|
636 |
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, |
637 |
GetBitContext *gb, int common_window)
|
638 |
{ |
639 |
if (get_bits1(gb)) {
|
640 |
av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
|
641 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
642 |
return -1; |
643 |
} |
644 |
ics->window_sequence[1] = ics->window_sequence[0]; |
645 |
ics->window_sequence[0] = get_bits(gb, 2); |
646 |
ics->use_kb_window[1] = ics->use_kb_window[0]; |
647 |
ics->use_kb_window[0] = get_bits1(gb);
|
648 |
ics->num_window_groups = 1;
|
649 |
ics->group_len[0] = 1; |
650 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
651 |
int i;
|
652 |
ics->max_sfb = get_bits(gb, 4);
|
653 |
for (i = 0; i < 7; i++) { |
654 |
if (get_bits1(gb)) {
|
655 |
ics->group_len[ics->num_window_groups - 1]++;
|
656 |
} else {
|
657 |
ics->num_window_groups++; |
658 |
ics->group_len[ics->num_window_groups - 1] = 1; |
659 |
} |
660 |
} |
661 |
ics->num_windows = 8;
|
662 |
ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; |
663 |
ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; |
664 |
ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; |
665 |
ics->predictor_present = 0;
|
666 |
} else {
|
667 |
ics->max_sfb = get_bits(gb, 6);
|
668 |
ics->num_windows = 1;
|
669 |
ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; |
670 |
ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; |
671 |
ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; |
672 |
ics->predictor_present = get_bits1(gb); |
673 |
ics->predictor_reset_group = 0;
|
674 |
if (ics->predictor_present) {
|
675 |
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
676 |
if (decode_prediction(ac, ics, gb)) {
|
677 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
678 |
return -1; |
679 |
} |
680 |
} else if (ac->m4ac.object_type == AOT_AAC_LC) { |
681 |
av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
|
682 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
683 |
return -1; |
684 |
} else {
|
685 |
av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1); |
686 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
687 |
return -1; |
688 |
} |
689 |
} |
690 |
} |
691 |
|
692 |
if (ics->max_sfb > ics->num_swb) {
|
693 |
av_log(ac->avctx, AV_LOG_ERROR, |
694 |
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
|
695 |
ics->max_sfb, ics->num_swb); |
696 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
697 |
return -1; |
698 |
} |
699 |
|
700 |
return 0; |
701 |
} |
702 |
|
703 |
/**
|
704 |
* Decode band types (section_data payload); reference: table 4.46.
|
705 |
*
|
706 |
* @param band_type array of the used band type
|
707 |
* @param band_type_run_end array of the last scalefactor band of a band type run
|
708 |
*
|
709 |
* @return Returns error status. 0 - OK, !0 - error
|
710 |
*/
|
711 |
static int decode_band_types(AACContext *ac, enum BandType band_type[120], |
712 |
int band_type_run_end[120], GetBitContext *gb, |
713 |
IndividualChannelStream *ics) |
714 |
{ |
715 |
int g, idx = 0; |
716 |
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
717 |
for (g = 0; g < ics->num_window_groups; g++) { |
718 |
int k = 0; |
719 |
while (k < ics->max_sfb) {
|
720 |
uint8_t sect_end = k; |
721 |
int sect_len_incr;
|
722 |
int sect_band_type = get_bits(gb, 4); |
723 |
if (sect_band_type == 12) { |
724 |
av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
|
725 |
return -1; |
726 |
} |
727 |
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) |
728 |
sect_end += sect_len_incr; |
729 |
sect_end += sect_len_incr; |
730 |
if (get_bits_left(gb) < 0) { |
731 |
av_log(ac->avctx, AV_LOG_ERROR, overread_err); |
732 |
return -1; |
733 |
} |
734 |
if (sect_end > ics->max_sfb) {
|
735 |
av_log(ac->avctx, AV_LOG_ERROR, |
736 |
"Number of bands (%d) exceeds limit (%d).\n",
|
737 |
sect_end, ics->max_sfb); |
738 |
return -1; |
739 |
} |
740 |
for (; k < sect_end; k++) {
|
741 |
band_type [idx] = sect_band_type; |
742 |
band_type_run_end[idx++] = sect_end; |
743 |
} |
744 |
} |
745 |
} |
746 |
return 0; |
747 |
} |
748 |
|
749 |
/**
|
750 |
* Decode scalefactors; reference: table 4.47.
|
751 |
*
|
752 |
* @param global_gain first scalefactor value as scalefactors are differentially coded
|
753 |
* @param band_type array of the used band type
|
754 |
* @param band_type_run_end array of the last scalefactor band of a band type run
|
755 |
* @param sf array of scalefactors or intensity stereo positions
|
756 |
*
|
757 |
* @return Returns error status. 0 - OK, !0 - error
|
758 |
*/
|
759 |
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, |
760 |
unsigned int global_gain, |
761 |
IndividualChannelStream *ics, |
762 |
enum BandType band_type[120], |
763 |
int band_type_run_end[120]) |
764 |
{ |
765 |
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
766 |
int g, i, idx = 0; |
767 |
int offset[3] = { global_gain, global_gain - 90, 100 }; |
768 |
int noise_flag = 1; |
769 |
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
770 |
for (g = 0; g < ics->num_window_groups; g++) { |
771 |
for (i = 0; i < ics->max_sfb;) { |
772 |
int run_end = band_type_run_end[idx];
|
773 |
if (band_type[idx] == ZERO_BT) {
|
774 |
for (; i < run_end; i++, idx++)
|
775 |
sf[idx] = 0.;
|
776 |
} else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
777 |
for (; i < run_end; i++, idx++) {
|
778 |
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
779 |
if (offset[2] > 255U) { |
780 |
av_log(ac->avctx, AV_LOG_ERROR, |
781 |
"%s (%d) out of range.\n", sf_str[2], offset[2]); |
782 |
return -1; |
783 |
} |
784 |
sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
785 |
} |
786 |
} else if (band_type[idx] == NOISE_BT) { |
787 |
for (; i < run_end; i++, idx++) {
|
788 |
if (noise_flag-- > 0) |
789 |
offset[1] += get_bits(gb, 9) - 256; |
790 |
else
|
791 |
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
792 |
if (offset[1] > 255U) { |
793 |
av_log(ac->avctx, AV_LOG_ERROR, |
794 |
"%s (%d) out of range.\n", sf_str[1], offset[1]); |
795 |
return -1; |
796 |
} |
797 |
sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; |
798 |
} |
799 |
} else {
|
800 |
for (; i < run_end; i++, idx++) {
|
801 |
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
802 |
if (offset[0] > 255U) { |
803 |
av_log(ac->avctx, AV_LOG_ERROR, |
804 |
"%s (%d) out of range.\n", sf_str[0], offset[0]); |
805 |
return -1; |
806 |
} |
807 |
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
|
808 |
} |
809 |
} |
810 |
} |
811 |
} |
812 |
return 0; |
813 |
} |
814 |
|
815 |
/**
|
816 |
* Decode pulse data; reference: table 4.7.
|
817 |
*/
|
818 |
static int decode_pulses(Pulse *pulse, GetBitContext *gb, |
819 |
const uint16_t *swb_offset, int num_swb) |
820 |
{ |
821 |
int i, pulse_swb;
|
822 |
pulse->num_pulse = get_bits(gb, 2) + 1; |
823 |
pulse_swb = get_bits(gb, 6);
|
824 |
if (pulse_swb >= num_swb)
|
825 |
return -1; |
826 |
pulse->pos[0] = swb_offset[pulse_swb];
|
827 |
pulse->pos[0] += get_bits(gb, 5); |
828 |
if (pulse->pos[0] > 1023) |
829 |
return -1; |
830 |
pulse->amp[0] = get_bits(gb, 4); |
831 |
for (i = 1; i < pulse->num_pulse; i++) { |
832 |
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; |
833 |
if (pulse->pos[i] > 1023) |
834 |
return -1; |
835 |
pulse->amp[i] = get_bits(gb, 4);
|
836 |
} |
837 |
return 0; |
838 |
} |
839 |
|
840 |
/**
|
841 |
* Decode Temporal Noise Shaping data; reference: table 4.48.
|
842 |
*
|
843 |
* @return Returns error status. 0 - OK, !0 - error
|
844 |
*/
|
845 |
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, |
846 |
GetBitContext *gb, const IndividualChannelStream *ics)
|
847 |
{ |
848 |
int w, filt, i, coef_len, coef_res, coef_compress;
|
849 |
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; |
850 |
const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; |
851 |
for (w = 0; w < ics->num_windows; w++) { |
852 |
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
853 |
coef_res = get_bits1(gb); |
854 |
|
855 |
for (filt = 0; filt < tns->n_filt[w]; filt++) { |
856 |
int tmp2_idx;
|
857 |
tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); |
858 |
|
859 |
if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { |
860 |
av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
|
861 |
tns->order[w][filt], tns_max_order); |
862 |
tns->order[w][filt] = 0;
|
863 |
return -1; |
864 |
} |
865 |
if (tns->order[w][filt]) {
|
866 |
tns->direction[w][filt] = get_bits1(gb); |
867 |
coef_compress = get_bits1(gb); |
868 |
coef_len = coef_res + 3 - coef_compress;
|
869 |
tmp2_idx = 2 * coef_compress + coef_res;
|
870 |
|
871 |
for (i = 0; i < tns->order[w][filt]; i++) |
872 |
tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; |
873 |
} |
874 |
} |
875 |
} |
876 |
} |
877 |
return 0; |
878 |
} |
879 |
|
880 |
/**
|
881 |
* Decode Mid/Side data; reference: table 4.54.
|
882 |
*
|
883 |
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
884 |
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
885 |
* [3] reserved for scalable AAC
|
886 |
*/
|
887 |
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, |
888 |
int ms_present)
|
889 |
{ |
890 |
int idx;
|
891 |
if (ms_present == 1) { |
892 |
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) |
893 |
cpe->ms_mask[idx] = get_bits1(gb); |
894 |
} else if (ms_present == 2) { |
895 |
memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); |
896 |
} |
897 |
} |
898 |
|
899 |
#ifndef VMUL2
|
900 |
static inline float *VMUL2(float *dst, const float *v, unsigned idx, |
901 |
const float *scale) |
902 |
{ |
903 |
float s = *scale;
|
904 |
*dst++ = v[idx & 15] * s;
|
905 |
*dst++ = v[idx>>4 & 15] * s; |
906 |
return dst;
|
907 |
} |
908 |
#endif
|
909 |
|
910 |
#ifndef VMUL4
|
911 |
static inline float *VMUL4(float *dst, const float *v, unsigned idx, |
912 |
const float *scale) |
913 |
{ |
914 |
float s = *scale;
|
915 |
*dst++ = v[idx & 3] * s;
|
916 |
*dst++ = v[idx>>2 & 3] * s; |
917 |
*dst++ = v[idx>>4 & 3] * s; |
918 |
*dst++ = v[idx>>6 & 3] * s; |
919 |
return dst;
|
920 |
} |
921 |
#endif
|
922 |
|
923 |
#ifndef VMUL2S
|
924 |
static inline float *VMUL2S(float *dst, const float *v, unsigned idx, |
925 |
unsigned sign, const float *scale) |
926 |
{ |
927 |
union float754 s0, s1;
|
928 |
|
929 |
s0.f = s1.f = *scale; |
930 |
s0.i ^= sign >> 1 << 31; |
931 |
s1.i ^= sign << 31;
|
932 |
|
933 |
*dst++ = v[idx & 15] * s0.f;
|
934 |
*dst++ = v[idx>>4 & 15] * s1.f; |
935 |
|
936 |
return dst;
|
937 |
} |
938 |
#endif
|
939 |
|
940 |
#ifndef VMUL4S
|
941 |
static inline float *VMUL4S(float *dst, const float *v, unsigned idx, |
942 |
unsigned sign, const float *scale) |
943 |
{ |
944 |
unsigned nz = idx >> 12; |
945 |
union float754 s = { .f = *scale };
|
946 |
union float754 t;
|
947 |
|
948 |
t.i = s.i ^ (sign & 1<<31); |
949 |
*dst++ = v[idx & 3] * t.f;
|
950 |
|
951 |
sign <<= nz & 1; nz >>= 1; |
952 |
t.i = s.i ^ (sign & 1<<31); |
953 |
*dst++ = v[idx>>2 & 3] * t.f; |
954 |
|
955 |
sign <<= nz & 1; nz >>= 1; |
956 |
t.i = s.i ^ (sign & 1<<31); |
957 |
*dst++ = v[idx>>4 & 3] * t.f; |
958 |
|
959 |
sign <<= nz & 1; nz >>= 1; |
960 |
t.i = s.i ^ (sign & 1<<31); |
961 |
*dst++ = v[idx>>6 & 3] * t.f; |
962 |
|
963 |
return dst;
|
964 |
} |
965 |
#endif
|
966 |
|
967 |
/**
|
968 |
* Decode spectral data; reference: table 4.50.
|
969 |
* Dequantize and scale spectral data; reference: 4.6.3.3.
|
970 |
*
|
971 |
* @param coef array of dequantized, scaled spectral data
|
972 |
* @param sf array of scalefactors or intensity stereo positions
|
973 |
* @param pulse_present set if pulses are present
|
974 |
* @param pulse pointer to pulse data struct
|
975 |
* @param band_type array of the used band type
|
976 |
*
|
977 |
* @return Returns error status. 0 - OK, !0 - error
|
978 |
*/
|
979 |
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], |
980 |
GetBitContext *gb, const float sf[120], |
981 |
int pulse_present, const Pulse *pulse, |
982 |
const IndividualChannelStream *ics,
|
983 |
enum BandType band_type[120]) |
984 |
{ |
985 |
int i, k, g, idx = 0; |
986 |
const int c = 1024 / ics->num_windows; |
987 |
const uint16_t *offsets = ics->swb_offset;
|
988 |
float *coef_base = coef;
|
989 |
|
990 |
for (g = 0; g < ics->num_windows; g++) |
991 |
memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); |
992 |
|
993 |
for (g = 0; g < ics->num_window_groups; g++) { |
994 |
unsigned g_len = ics->group_len[g];
|
995 |
|
996 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
997 |
const unsigned cbt_m1 = band_type[idx] - 1; |
998 |
float *cfo = coef + offsets[i];
|
999 |
int off_len = offsets[i + 1] - offsets[i]; |
1000 |
int group;
|
1001 |
|
1002 |
if (cbt_m1 >= INTENSITY_BT2 - 1) { |
1003 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1004 |
memset(cfo, 0, off_len * sizeof(float)); |
1005 |
} |
1006 |
} else if (cbt_m1 == NOISE_BT - 1) { |
1007 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1008 |
float scale;
|
1009 |
float band_energy;
|
1010 |
|
1011 |
for (k = 0; k < off_len; k++) { |
1012 |
ac->random_state = lcg_random(ac->random_state); |
1013 |
cfo[k] = ac->random_state; |
1014 |
} |
1015 |
|
1016 |
band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len); |
1017 |
scale = sf[idx] / sqrtf(band_energy); |
1018 |
ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len); |
1019 |
} |
1020 |
} else {
|
1021 |
const float *vq = ff_aac_codebook_vector_vals[cbt_m1]; |
1022 |
const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
|
1023 |
VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
|
1024 |
OPEN_READER(re, gb); |
1025 |
|
1026 |
switch (cbt_m1 >> 1) { |
1027 |
case 0: |
1028 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1029 |
float *cf = cfo;
|
1030 |
int len = off_len;
|
1031 |
|
1032 |
do {
|
1033 |
int code;
|
1034 |
unsigned cb_idx;
|
1035 |
|
1036 |
UPDATE_CACHE(re, gb); |
1037 |
GET_VLC(code, re, gb, vlc_tab, 8, 2); |
1038 |
cb_idx = cb_vector_idx[code]; |
1039 |
cf = VMUL4(cf, vq, cb_idx, sf + idx); |
1040 |
} while (len -= 4); |
1041 |
} |
1042 |
break;
|
1043 |
|
1044 |
case 1: |
1045 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1046 |
float *cf = cfo;
|
1047 |
int len = off_len;
|
1048 |
|
1049 |
do {
|
1050 |
int code;
|
1051 |
unsigned nnz;
|
1052 |
unsigned cb_idx;
|
1053 |
uint32_t bits; |
1054 |
|
1055 |
UPDATE_CACHE(re, gb); |
1056 |
GET_VLC(code, re, gb, vlc_tab, 8, 2); |
1057 |
cb_idx = cb_vector_idx[code]; |
1058 |
nnz = cb_idx >> 8 & 15; |
1059 |
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
|
1060 |
LAST_SKIP_BITS(re, gb, nnz); |
1061 |
cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx); |
1062 |
} while (len -= 4); |
1063 |
} |
1064 |
break;
|
1065 |
|
1066 |
case 2: |
1067 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1068 |
float *cf = cfo;
|
1069 |
int len = off_len;
|
1070 |
|
1071 |
do {
|
1072 |
int code;
|
1073 |
unsigned cb_idx;
|
1074 |
|
1075 |
UPDATE_CACHE(re, gb); |
1076 |
GET_VLC(code, re, gb, vlc_tab, 8, 2); |
1077 |
cb_idx = cb_vector_idx[code]; |
1078 |
cf = VMUL2(cf, vq, cb_idx, sf + idx); |
1079 |
} while (len -= 2); |
1080 |
} |
1081 |
break;
|
1082 |
|
1083 |
case 3: |
1084 |
case 4: |
1085 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1086 |
float *cf = cfo;
|
1087 |
int len = off_len;
|
1088 |
|
1089 |
do {
|
1090 |
int code;
|
1091 |
unsigned nnz;
|
1092 |
unsigned cb_idx;
|
1093 |
unsigned sign;
|
1094 |
|
1095 |
UPDATE_CACHE(re, gb); |
1096 |
GET_VLC(code, re, gb, vlc_tab, 8, 2); |
1097 |
cb_idx = cb_vector_idx[code]; |
1098 |
nnz = cb_idx >> 8 & 15; |
1099 |
sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
|
1100 |
LAST_SKIP_BITS(re, gb, nnz); |
1101 |
cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx); |
1102 |
} while (len -= 2); |
1103 |
} |
1104 |
break;
|
1105 |
|
1106 |
default:
|
1107 |
for (group = 0; group < g_len; group++, cfo+=128) { |
1108 |
float *cf = cfo;
|
1109 |
uint32_t *icf = (uint32_t *) cf; |
1110 |
int len = off_len;
|
1111 |
|
1112 |
do {
|
1113 |
int code;
|
1114 |
unsigned nzt, nnz;
|
1115 |
unsigned cb_idx;
|
1116 |
uint32_t bits; |
1117 |
int j;
|
1118 |
|
1119 |
UPDATE_CACHE(re, gb); |
1120 |
GET_VLC(code, re, gb, vlc_tab, 8, 2); |
1121 |
|
1122 |
if (!code) {
|
1123 |
*icf++ = 0;
|
1124 |
*icf++ = 0;
|
1125 |
continue;
|
1126 |
} |
1127 |
|
1128 |
cb_idx = cb_vector_idx[code]; |
1129 |
nnz = cb_idx >> 12;
|
1130 |
nzt = cb_idx >> 8;
|
1131 |
bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
|
1132 |
LAST_SKIP_BITS(re, gb, nnz); |
1133 |
|
1134 |
for (j = 0; j < 2; j++) { |
1135 |
if (nzt & 1<<j) { |
1136 |
uint32_t b; |
1137 |
int n;
|
1138 |
/* The total length of escape_sequence must be < 22 bits according
|
1139 |
to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
|
1140 |
UPDATE_CACHE(re, gb); |
1141 |
b = GET_CACHE(re, gb); |
1142 |
b = 31 - av_log2(~b);
|
1143 |
|
1144 |
if (b > 8) { |
1145 |
av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
|
1146 |
return -1; |
1147 |
} |
1148 |
|
1149 |
SKIP_BITS(re, gb, b + 1);
|
1150 |
b += 4;
|
1151 |
n = (1 << b) + SHOW_UBITS(re, gb, b);
|
1152 |
LAST_SKIP_BITS(re, gb, b); |
1153 |
*icf++ = cbrt_tab[n] | (bits & 1<<31); |
1154 |
bits <<= 1;
|
1155 |
} else {
|
1156 |
unsigned v = ((const uint32_t*)vq)[cb_idx & 15]; |
1157 |
*icf++ = (bits & 1<<31) | v; |
1158 |
bits <<= !!v; |
1159 |
} |
1160 |
cb_idx >>= 4;
|
1161 |
} |
1162 |
} while (len -= 2); |
1163 |
|
1164 |
ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len); |
1165 |
} |
1166 |
} |
1167 |
|
1168 |
CLOSE_READER(re, gb); |
1169 |
} |
1170 |
} |
1171 |
coef += g_len << 7;
|
1172 |
} |
1173 |
|
1174 |
if (pulse_present) {
|
1175 |
idx = 0;
|
1176 |
for (i = 0; i < pulse->num_pulse; i++) { |
1177 |
float co = coef_base[ pulse->pos[i] ];
|
1178 |
while (offsets[idx + 1] <= pulse->pos[i]) |
1179 |
idx++; |
1180 |
if (band_type[idx] != NOISE_BT && sf[idx]) {
|
1181 |
float ico = -pulse->amp[i];
|
1182 |
if (co) {
|
1183 |
co /= sf[idx]; |
1184 |
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
|
1185 |
} |
1186 |
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; |
1187 |
} |
1188 |
} |
1189 |
} |
1190 |
return 0; |
1191 |
} |
1192 |
|
1193 |
static av_always_inline float flt16_round(float pf) |
1194 |
{ |
1195 |
union float754 tmp;
|
1196 |
tmp.f = pf; |
1197 |
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
1198 |
return tmp.f;
|
1199 |
} |
1200 |
|
1201 |
static av_always_inline float flt16_even(float pf) |
1202 |
{ |
1203 |
union float754 tmp;
|
1204 |
tmp.f = pf; |
1205 |
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; |
1206 |
return tmp.f;
|
1207 |
} |
1208 |
|
1209 |
static av_always_inline float flt16_trunc(float pf) |
1210 |
{ |
1211 |
union float754 pun;
|
1212 |
pun.f = pf; |
1213 |
pun.i &= 0xFFFF0000U;
|
1214 |
return pun.f;
|
1215 |
} |
1216 |
|
1217 |
static av_always_inline void predict(PredictorState *ps, float *coef, |
1218 |
float sf_scale, float inv_sf_scale, |
1219 |
int output_enable)
|
1220 |
{ |
1221 |
const float a = 0.953125; // 61.0 / 64 |
1222 |
const float alpha = 0.90625; // 29.0 / 32 |
1223 |
float e0, e1;
|
1224 |
float pv;
|
1225 |
float k1, k2;
|
1226 |
float r0 = ps->r0, r1 = ps->r1;
|
1227 |
float cor0 = ps->cor0, cor1 = ps->cor1;
|
1228 |
float var0 = ps->var0, var1 = ps->var1;
|
1229 |
|
1230 |
k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; |
1231 |
k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; |
1232 |
|
1233 |
pv = flt16_round(k1 * r0 + k2 * r1); |
1234 |
if (output_enable)
|
1235 |
*coef += pv * sf_scale; |
1236 |
|
1237 |
e0 = *coef * inv_sf_scale; |
1238 |
e1 = e0 - k1 * r0; |
1239 |
|
1240 |
ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); |
1241 |
ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); |
1242 |
ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); |
1243 |
ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); |
1244 |
|
1245 |
ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); |
1246 |
ps->r0 = flt16_trunc(a * e0); |
1247 |
} |
1248 |
|
1249 |
/**
|
1250 |
* Apply AAC-Main style frequency domain prediction.
|
1251 |
*/
|
1252 |
static void apply_prediction(AACContext *ac, SingleChannelElement *sce) |
1253 |
{ |
1254 |
int sfb, k;
|
1255 |
float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale; |
1256 |
|
1257 |
if (!sce->ics.predictor_initialized) {
|
1258 |
reset_all_predictors(sce->predictor_state); |
1259 |
sce->ics.predictor_initialized = 1;
|
1260 |
} |
1261 |
|
1262 |
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
1263 |
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
1264 |
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
1265 |
predict(&sce->predictor_state[k], &sce->coeffs[k], |
1266 |
sf_scale, inv_sf_scale, |
1267 |
sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
1268 |
} |
1269 |
} |
1270 |
if (sce->ics.predictor_reset_group)
|
1271 |
reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
1272 |
} else
|
1273 |
reset_all_predictors(sce->predictor_state); |
1274 |
} |
1275 |
|
1276 |
/**
|
1277 |
* Decode an individual_channel_stream payload; reference: table 4.44.
|
1278 |
*
|
1279 |
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
1280 |
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
|
1281 |
*
|
1282 |
* @return Returns error status. 0 - OK, !0 - error
|
1283 |
*/
|
1284 |
static int decode_ics(AACContext *ac, SingleChannelElement *sce, |
1285 |
GetBitContext *gb, int common_window, int scale_flag) |
1286 |
{ |
1287 |
Pulse pulse; |
1288 |
TemporalNoiseShaping *tns = &sce->tns; |
1289 |
IndividualChannelStream *ics = &sce->ics; |
1290 |
float *out = sce->coeffs;
|
1291 |
int global_gain, pulse_present = 0; |
1292 |
|
1293 |
/* This assignment is to silence a GCC warning about the variable being used
|
1294 |
* uninitialized when in fact it always is.
|
1295 |
*/
|
1296 |
pulse.num_pulse = 0;
|
1297 |
|
1298 |
global_gain = get_bits(gb, 8);
|
1299 |
|
1300 |
if (!common_window && !scale_flag) {
|
1301 |
if (decode_ics_info(ac, ics, gb, 0) < 0) |
1302 |
return -1; |
1303 |
} |
1304 |
|
1305 |
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) |
1306 |
return -1; |
1307 |
if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) |
1308 |
return -1; |
1309 |
|
1310 |
pulse_present = 0;
|
1311 |
if (!scale_flag) {
|
1312 |
if ((pulse_present = get_bits1(gb))) {
|
1313 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1314 |
av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
|
1315 |
return -1; |
1316 |
} |
1317 |
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
|
1318 |
av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
|
1319 |
return -1; |
1320 |
} |
1321 |
} |
1322 |
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
|
1323 |
return -1; |
1324 |
if (get_bits1(gb)) {
|
1325 |
av_log_missing_feature(ac->avctx, "SSR", 1); |
1326 |
return -1; |
1327 |
} |
1328 |
} |
1329 |
|
1330 |
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
1331 |
return -1; |
1332 |
|
1333 |
if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
|
1334 |
apply_prediction(ac, sce); |
1335 |
|
1336 |
return 0; |
1337 |
} |
1338 |
|
1339 |
/**
|
1340 |
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
|
1341 |
*/
|
1342 |
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) |
1343 |
{ |
1344 |
const IndividualChannelStream *ics = &cpe->ch[0].ics; |
1345 |
float *ch0 = cpe->ch[0].coeffs; |
1346 |
float *ch1 = cpe->ch[1].coeffs; |
1347 |
int g, i, group, idx = 0; |
1348 |
const uint16_t *offsets = ics->swb_offset;
|
1349 |
for (g = 0; g < ics->num_window_groups; g++) { |
1350 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
1351 |
if (cpe->ms_mask[idx] &&
|
1352 |
cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { |
1353 |
for (group = 0; group < ics->group_len[g]; group++) { |
1354 |
ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
|
1355 |
ch1 + group * 128 + offsets[i],
|
1356 |
offsets[i+1] - offsets[i]);
|
1357 |
} |
1358 |
} |
1359 |
} |
1360 |
ch0 += ics->group_len[g] * 128;
|
1361 |
ch1 += ics->group_len[g] * 128;
|
1362 |
} |
1363 |
} |
1364 |
|
1365 |
/**
|
1366 |
* intensity stereo decoding; reference: 4.6.8.2.3
|
1367 |
*
|
1368 |
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
1369 |
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
1370 |
* [3] reserved for scalable AAC
|
1371 |
*/
|
1372 |
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present) |
1373 |
{ |
1374 |
const IndividualChannelStream *ics = &cpe->ch[1].ics; |
1375 |
SingleChannelElement *sce1 = &cpe->ch[1];
|
1376 |
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; |
1377 |
const uint16_t *offsets = ics->swb_offset;
|
1378 |
int g, group, i, k, idx = 0; |
1379 |
int c;
|
1380 |
float scale;
|
1381 |
for (g = 0; g < ics->num_window_groups; g++) { |
1382 |
for (i = 0; i < ics->max_sfb;) { |
1383 |
if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
|
1384 |
const int bt_run_end = sce1->band_type_run_end[idx]; |
1385 |
for (; i < bt_run_end; i++, idx++) {
|
1386 |
c = -1 + 2 * (sce1->band_type[idx] - 14); |
1387 |
if (ms_present)
|
1388 |
c *= 1 - 2 * cpe->ms_mask[idx]; |
1389 |
scale = c * sce1->sf[idx]; |
1390 |
for (group = 0; group < ics->group_len[g]; group++) |
1391 |
for (k = offsets[i]; k < offsets[i + 1]; k++) |
1392 |
coef1[group * 128 + k] = scale * coef0[group * 128 + k]; |
1393 |
} |
1394 |
} else {
|
1395 |
int bt_run_end = sce1->band_type_run_end[idx];
|
1396 |
idx += bt_run_end - i; |
1397 |
i = bt_run_end; |
1398 |
} |
1399 |
} |
1400 |
coef0 += ics->group_len[g] * 128;
|
1401 |
coef1 += ics->group_len[g] * 128;
|
1402 |
} |
1403 |
} |
1404 |
|
1405 |
/**
|
1406 |
* Decode a channel_pair_element; reference: table 4.4.
|
1407 |
*
|
1408 |
* @return Returns error status. 0 - OK, !0 - error
|
1409 |
*/
|
1410 |
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) |
1411 |
{ |
1412 |
int i, ret, common_window, ms_present = 0; |
1413 |
|
1414 |
common_window = get_bits1(gb); |
1415 |
if (common_window) {
|
1416 |
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) |
1417 |
return -1; |
1418 |
i = cpe->ch[1].ics.use_kb_window[0]; |
1419 |
cpe->ch[1].ics = cpe->ch[0].ics; |
1420 |
cpe->ch[1].ics.use_kb_window[1] = i; |
1421 |
ms_present = get_bits(gb, 2);
|
1422 |
if (ms_present == 3) { |
1423 |
av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
|
1424 |
return -1; |
1425 |
} else if (ms_present) |
1426 |
decode_mid_side_stereo(cpe, gb, ms_present); |
1427 |
} |
1428 |
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) |
1429 |
return ret;
|
1430 |
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) |
1431 |
return ret;
|
1432 |
|
1433 |
if (common_window) {
|
1434 |
if (ms_present)
|
1435 |
apply_mid_side_stereo(ac, cpe); |
1436 |
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
1437 |
apply_prediction(ac, &cpe->ch[0]);
|
1438 |
apply_prediction(ac, &cpe->ch[1]);
|
1439 |
} |
1440 |
} |
1441 |
|
1442 |
apply_intensity_stereo(cpe, ms_present); |
1443 |
return 0; |
1444 |
} |
1445 |
|
1446 |
static const float cce_scale[] = { |
1447 |
1.09050773266525765921, //2^(1/8) |
1448 |
1.18920711500272106672, //2^(1/4) |
1449 |
M_SQRT2, |
1450 |
2,
|
1451 |
}; |
1452 |
|
1453 |
/**
|
1454 |
* Decode coupling_channel_element; reference: table 4.8.
|
1455 |
*
|
1456 |
* @return Returns error status. 0 - OK, !0 - error
|
1457 |
*/
|
1458 |
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) |
1459 |
{ |
1460 |
int num_gain = 0; |
1461 |
int c, g, sfb, ret;
|
1462 |
int sign;
|
1463 |
float scale;
|
1464 |
SingleChannelElement *sce = &che->ch[0];
|
1465 |
ChannelCoupling *coup = &che->coup; |
1466 |
|
1467 |
coup->coupling_point = 2 * get_bits1(gb);
|
1468 |
coup->num_coupled = get_bits(gb, 3);
|
1469 |
for (c = 0; c <= coup->num_coupled; c++) { |
1470 |
num_gain++; |
1471 |
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; |
1472 |
coup->id_select[c] = get_bits(gb, 4);
|
1473 |
if (coup->type[c] == TYPE_CPE) {
|
1474 |
coup->ch_select[c] = get_bits(gb, 2);
|
1475 |
if (coup->ch_select[c] == 3) |
1476 |
num_gain++; |
1477 |
} else
|
1478 |
coup->ch_select[c] = 2;
|
1479 |
} |
1480 |
coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
|
1481 |
|
1482 |
sign = get_bits(gb, 1);
|
1483 |
scale = cce_scale[get_bits(gb, 2)];
|
1484 |
|
1485 |
if ((ret = decode_ics(ac, sce, gb, 0, 0))) |
1486 |
return ret;
|
1487 |
|
1488 |
for (c = 0; c < num_gain; c++) { |
1489 |
int idx = 0; |
1490 |
int cge = 1; |
1491 |
int gain = 0; |
1492 |
float gain_cache = 1.; |
1493 |
if (c) {
|
1494 |
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
|
1495 |
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; |
1496 |
gain_cache = powf(scale, -gain); |
1497 |
} |
1498 |
if (coup->coupling_point == AFTER_IMDCT) {
|
1499 |
coup->gain[c][0] = gain_cache;
|
1500 |
} else {
|
1501 |
for (g = 0; g < sce->ics.num_window_groups; g++) { |
1502 |
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { |
1503 |
if (sce->band_type[idx] != ZERO_BT) {
|
1504 |
if (!cge) {
|
1505 |
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
1506 |
if (t) {
|
1507 |
int s = 1; |
1508 |
t = gain += t; |
1509 |
if (sign) {
|
1510 |
s -= 2 * (t & 0x1); |
1511 |
t >>= 1;
|
1512 |
} |
1513 |
gain_cache = powf(scale, -t) * s; |
1514 |
} |
1515 |
} |
1516 |
coup->gain[c][idx] = gain_cache; |
1517 |
} |
1518 |
} |
1519 |
} |
1520 |
} |
1521 |
} |
1522 |
return 0; |
1523 |
} |
1524 |
|
1525 |
/**
|
1526 |
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
|
1527 |
*
|
1528 |
* @return Returns number of bytes consumed.
|
1529 |
*/
|
1530 |
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, |
1531 |
GetBitContext *gb) |
1532 |
{ |
1533 |
int i;
|
1534 |
int num_excl_chan = 0; |
1535 |
|
1536 |
do {
|
1537 |
for (i = 0; i < 7; i++) |
1538 |
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); |
1539 |
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); |
1540 |
|
1541 |
return num_excl_chan / 7; |
1542 |
} |
1543 |
|
1544 |
/**
|
1545 |
* Decode dynamic range information; reference: table 4.52.
|
1546 |
*
|
1547 |
* @param cnt length of TYPE_FIL syntactic element in bytes
|
1548 |
*
|
1549 |
* @return Returns number of bytes consumed.
|
1550 |
*/
|
1551 |
static int decode_dynamic_range(DynamicRangeControl *che_drc, |
1552 |
GetBitContext *gb, int cnt)
|
1553 |
{ |
1554 |
int n = 1; |
1555 |
int drc_num_bands = 1; |
1556 |
int i;
|
1557 |
|
1558 |
/* pce_tag_present? */
|
1559 |
if (get_bits1(gb)) {
|
1560 |
che_drc->pce_instance_tag = get_bits(gb, 4);
|
1561 |
skip_bits(gb, 4); // tag_reserved_bits |
1562 |
n++; |
1563 |
} |
1564 |
|
1565 |
/* excluded_chns_present? */
|
1566 |
if (get_bits1(gb)) {
|
1567 |
n += decode_drc_channel_exclusions(che_drc, gb); |
1568 |
} |
1569 |
|
1570 |
/* drc_bands_present? */
|
1571 |
if (get_bits1(gb)) {
|
1572 |
che_drc->band_incr = get_bits(gb, 4);
|
1573 |
che_drc->interpolation_scheme = get_bits(gb, 4);
|
1574 |
n++; |
1575 |
drc_num_bands += che_drc->band_incr; |
1576 |
for (i = 0; i < drc_num_bands; i++) { |
1577 |
che_drc->band_top[i] = get_bits(gb, 8);
|
1578 |
n++; |
1579 |
} |
1580 |
} |
1581 |
|
1582 |
/* prog_ref_level_present? */
|
1583 |
if (get_bits1(gb)) {
|
1584 |
che_drc->prog_ref_level = get_bits(gb, 7);
|
1585 |
skip_bits1(gb); // prog_ref_level_reserved_bits
|
1586 |
n++; |
1587 |
} |
1588 |
|
1589 |
for (i = 0; i < drc_num_bands; i++) { |
1590 |
che_drc->dyn_rng_sgn[i] = get_bits1(gb); |
1591 |
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
|
1592 |
n++; |
1593 |
} |
1594 |
|
1595 |
return n;
|
1596 |
} |
1597 |
|
1598 |
/**
|
1599 |
* Decode extension data (incomplete); reference: table 4.51.
|
1600 |
*
|
1601 |
* @param cnt length of TYPE_FIL syntactic element in bytes
|
1602 |
*
|
1603 |
* @return Returns number of bytes consumed
|
1604 |
*/
|
1605 |
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, |
1606 |
ChannelElement *che, enum RawDataBlockType elem_type)
|
1607 |
{ |
1608 |
int crc_flag = 0; |
1609 |
int res = cnt;
|
1610 |
switch (get_bits(gb, 4)) { // extension type |
1611 |
case EXT_SBR_DATA_CRC:
|
1612 |
crc_flag++; |
1613 |
case EXT_SBR_DATA:
|
1614 |
if (!che) {
|
1615 |
av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
|
1616 |
return res;
|
1617 |
} else if (!ac->m4ac.sbr) { |
1618 |
av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
|
1619 |
skip_bits_long(gb, 8 * cnt - 4); |
1620 |
return res;
|
1621 |
} else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) { |
1622 |
av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
|
1623 |
skip_bits_long(gb, 8 * cnt - 4); |
1624 |
return res;
|
1625 |
} else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) { |
1626 |
ac->m4ac.sbr = 1;
|
1627 |
ac->m4ac.ps = 1;
|
1628 |
output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured); |
1629 |
} else {
|
1630 |
ac->m4ac.sbr = 1;
|
1631 |
} |
1632 |
res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type); |
1633 |
break;
|
1634 |
case EXT_DYNAMIC_RANGE:
|
1635 |
res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
1636 |
break;
|
1637 |
case EXT_FILL:
|
1638 |
case EXT_FILL_DATA:
|
1639 |
case EXT_DATA_ELEMENT:
|
1640 |
default:
|
1641 |
skip_bits_long(gb, 8 * cnt - 4); |
1642 |
break;
|
1643 |
}; |
1644 |
return res;
|
1645 |
} |
1646 |
|
1647 |
/**
|
1648 |
* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
|
1649 |
*
|
1650 |
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
|
1651 |
* @param coef spectral coefficients
|
1652 |
*/
|
1653 |
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, |
1654 |
IndividualChannelStream *ics, int decode)
|
1655 |
{ |
1656 |
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
1657 |
int w, filt, m, i;
|
1658 |
int bottom, top, order, start, end, size, inc;
|
1659 |
float lpc[TNS_MAX_ORDER];
|
1660 |
|
1661 |
for (w = 0; w < ics->num_windows; w++) { |
1662 |
bottom = ics->num_swb; |
1663 |
for (filt = 0; filt < tns->n_filt[w]; filt++) { |
1664 |
top = bottom; |
1665 |
bottom = FFMAX(0, top - tns->length[w][filt]);
|
1666 |
order = tns->order[w][filt]; |
1667 |
if (order == 0) |
1668 |
continue;
|
1669 |
|
1670 |
// tns_decode_coef
|
1671 |
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
1672 |
|
1673 |
start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1674 |
end = ics->swb_offset[FFMIN( top, mmm)]; |
1675 |
if ((size = end - start) <= 0) |
1676 |
continue;
|
1677 |
if (tns->direction[w][filt]) {
|
1678 |
inc = -1;
|
1679 |
start = end - 1;
|
1680 |
} else {
|
1681 |
inc = 1;
|
1682 |
} |
1683 |
start += w * 128;
|
1684 |
|
1685 |
// ar filter
|
1686 |
for (m = 0; m < size; m++, start += inc) |
1687 |
for (i = 1; i <= FFMIN(m, order); i++) |
1688 |
coef[start] -= coef[start - i * inc] * lpc[i - 1];
|
1689 |
} |
1690 |
} |
1691 |
} |
1692 |
|
1693 |
/**
|
1694 |
* Conduct IMDCT and windowing.
|
1695 |
*/
|
1696 |
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce) |
1697 |
{ |
1698 |
IndividualChannelStream *ics = &sce->ics; |
1699 |
float *in = sce->coeffs;
|
1700 |
float *out = sce->ret;
|
1701 |
float *saved = sce->saved;
|
1702 |
const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1703 |
const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
1704 |
const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
1705 |
float *buf = ac->buf_mdct;
|
1706 |
float *temp = ac->temp;
|
1707 |
int i;
|
1708 |
|
1709 |
// imdct
|
1710 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1711 |
for (i = 0; i < 1024; i += 128) |
1712 |
ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
1713 |
} else
|
1714 |
ff_imdct_half(&ac->mdct, buf, in); |
1715 |
|
1716 |
/* window overlapping
|
1717 |
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
|
1718 |
* and long to short transitions are considered to be short to short
|
1719 |
* transitions. This leaves just two cases (long to long and short to short)
|
1720 |
* with a little special sauce for EIGHT_SHORT_SEQUENCE.
|
1721 |
*/
|
1722 |
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
1723 |
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
1724 |
ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, 0, 512); |
1725 |
} else {
|
1726 |
for (i = 0; i < 448; i++) |
1727 |
out[i] = saved[i]; |
1728 |
|
1729 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1730 |
ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 0, 64); |
1731 |
ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 0, 64); |
1732 |
ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 0, 64); |
1733 |
ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 0, 64); |
1734 |
ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 0, 64); |
1735 |
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
1736 |
} else {
|
1737 |
ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 0, 64); |
1738 |
for (i = 576; i < 1024; i++) |
1739 |
out[i] = buf[i-512];
|
1740 |
} |
1741 |
} |
1742 |
|
1743 |
// buffer update
|
1744 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1745 |
for (i = 0; i < 64; i++) |
1746 |
saved[i] = temp[64 + i];
|
1747 |
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
1748 |
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
1749 |
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
1750 |
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
1751 |
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
1752 |
memcpy( saved, buf + 512, 448 * sizeof(float)); |
1753 |
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
1754 |
} else { // LONG_STOP or ONLY_LONG |
1755 |
memcpy( saved, buf + 512, 512 * sizeof(float)); |
1756 |
} |
1757 |
} |
1758 |
|
1759 |
/**
|
1760 |
* Apply dependent channel coupling (applied before IMDCT).
|
1761 |
*
|
1762 |
* @param index index into coupling gain array
|
1763 |
*/
|
1764 |
static void apply_dependent_coupling(AACContext *ac, |
1765 |
SingleChannelElement *target, |
1766 |
ChannelElement *cce, int index)
|
1767 |
{ |
1768 |
IndividualChannelStream *ics = &cce->ch[0].ics;
|
1769 |
const uint16_t *offsets = ics->swb_offset;
|
1770 |
float *dest = target->coeffs;
|
1771 |
const float *src = cce->ch[0].coeffs; |
1772 |
int g, i, group, k, idx = 0; |
1773 |
if (ac->m4ac.object_type == AOT_AAC_LTP) {
|
1774 |
av_log(ac->avctx, AV_LOG_ERROR, |
1775 |
"Dependent coupling is not supported together with LTP\n");
|
1776 |
return;
|
1777 |
} |
1778 |
for (g = 0; g < ics->num_window_groups; g++) { |
1779 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
1780 |
if (cce->ch[0].band_type[idx] != ZERO_BT) { |
1781 |
const float gain = cce->coup.gain[index][idx]; |
1782 |
for (group = 0; group < ics->group_len[g]; group++) { |
1783 |
for (k = offsets[i]; k < offsets[i + 1]; k++) { |
1784 |
// XXX dsputil-ize
|
1785 |
dest[group * 128 + k] += gain * src[group * 128 + k]; |
1786 |
} |
1787 |
} |
1788 |
} |
1789 |
} |
1790 |
dest += ics->group_len[g] * 128;
|
1791 |
src += ics->group_len[g] * 128;
|
1792 |
} |
1793 |
} |
1794 |
|
1795 |
/**
|
1796 |
* Apply independent channel coupling (applied after IMDCT).
|
1797 |
*
|
1798 |
* @param index index into coupling gain array
|
1799 |
*/
|
1800 |
static void apply_independent_coupling(AACContext *ac, |
1801 |
SingleChannelElement *target, |
1802 |
ChannelElement *cce, int index)
|
1803 |
{ |
1804 |
int i;
|
1805 |
const float gain = cce->coup.gain[index][0]; |
1806 |
const float *src = cce->ch[0].ret; |
1807 |
float *dest = target->ret;
|
1808 |
const int len = 1024 << (ac->m4ac.sbr == 1); |
1809 |
|
1810 |
for (i = 0; i < len; i++) |
1811 |
dest[i] += gain * src[i]; |
1812 |
} |
1813 |
|
1814 |
/**
|
1815 |
* channel coupling transformation interface
|
1816 |
*
|
1817 |
* @param apply_coupling_method pointer to (in)dependent coupling function
|
1818 |
*/
|
1819 |
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, |
1820 |
enum RawDataBlockType type, int elem_id, |
1821 |
enum CouplingPoint coupling_point,
|
1822 |
void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) |
1823 |
{ |
1824 |
int i, c;
|
1825 |
|
1826 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
1827 |
ChannelElement *cce = ac->che[TYPE_CCE][i]; |
1828 |
int index = 0; |
1829 |
|
1830 |
if (cce && cce->coup.coupling_point == coupling_point) {
|
1831 |
ChannelCoupling *coup = &cce->coup; |
1832 |
|
1833 |
for (c = 0; c <= coup->num_coupled; c++) { |
1834 |
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
|
1835 |
if (coup->ch_select[c] != 1) { |
1836 |
apply_coupling_method(ac, &cc->ch[0], cce, index);
|
1837 |
if (coup->ch_select[c] != 0) |
1838 |
index++; |
1839 |
} |
1840 |
if (coup->ch_select[c] != 2) |
1841 |
apply_coupling_method(ac, &cc->ch[1], cce, index++);
|
1842 |
} else
|
1843 |
index += 1 + (coup->ch_select[c] == 3); |
1844 |
} |
1845 |
} |
1846 |
} |
1847 |
} |
1848 |
|
1849 |
/**
|
1850 |
* Convert spectral data to float samples, applying all supported tools as appropriate.
|
1851 |
*/
|
1852 |
static void spectral_to_sample(AACContext *ac) |
1853 |
{ |
1854 |
int i, type;
|
1855 |
for (type = 3; type >= 0; type--) { |
1856 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
1857 |
ChannelElement *che = ac->che[type][i]; |
1858 |
if (che) {
|
1859 |
if (type <= TYPE_CPE)
|
1860 |
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
1861 |
if (che->ch[0].tns.present) |
1862 |
apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); |
1863 |
if (che->ch[1].tns.present) |
1864 |
apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); |
1865 |
if (type <= TYPE_CPE)
|
1866 |
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
1867 |
if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
|
1868 |
imdct_and_windowing(ac, &che->ch[0]);
|
1869 |
if (type == TYPE_CPE) {
|
1870 |
imdct_and_windowing(ac, &che->ch[1]);
|
1871 |
} |
1872 |
if (ac->m4ac.sbr > 0) { |
1873 |
ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret); |
1874 |
} |
1875 |
} |
1876 |
if (type <= TYPE_CCE)
|
1877 |
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
1878 |
} |
1879 |
} |
1880 |
} |
1881 |
} |
1882 |
|
1883 |
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) |
1884 |
{ |
1885 |
int size;
|
1886 |
AACADTSHeaderInfo hdr_info; |
1887 |
|
1888 |
size = ff_aac_parse_header(gb, &hdr_info); |
1889 |
if (size > 0) { |
1890 |
if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
|
1891 |
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
1892 |
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
1893 |
ac->m4ac.chan_config = hdr_info.chan_config; |
1894 |
if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
|
1895 |
return -7; |
1896 |
if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
|
1897 |
return -7; |
1898 |
} else if (ac->output_configured != OC_LOCKED) { |
1899 |
ac->output_configured = OC_NONE; |
1900 |
} |
1901 |
if (ac->output_configured != OC_LOCKED) {
|
1902 |
ac->m4ac.sbr = -1;
|
1903 |
ac->m4ac.ps = -1;
|
1904 |
} |
1905 |
ac->m4ac.sample_rate = hdr_info.sample_rate; |
1906 |
ac->m4ac.sampling_index = hdr_info.sampling_index; |
1907 |
ac->m4ac.object_type = hdr_info.object_type; |
1908 |
if (!ac->avctx->sample_rate)
|
1909 |
ac->avctx->sample_rate = hdr_info.sample_rate; |
1910 |
if (hdr_info.num_aac_frames == 1) { |
1911 |
if (!hdr_info.crc_absent)
|
1912 |
skip_bits(gb, 16);
|
1913 |
} else {
|
1914 |
av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0); |
1915 |
return -1; |
1916 |
} |
1917 |
} |
1918 |
return size;
|
1919 |
} |
1920 |
|
1921 |
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, |
1922 |
int *data_size, GetBitContext *gb)
|
1923 |
{ |
1924 |
AACContext *ac = avctx->priv_data; |
1925 |
ChannelElement *che = NULL, *che_prev = NULL; |
1926 |
enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
|
1927 |
int err, elem_id, data_size_tmp;
|
1928 |
int samples = 0, multiplier; |
1929 |
|
1930 |
if (show_bits(gb, 12) == 0xfff) { |
1931 |
if (parse_adts_frame_header(ac, gb) < 0) { |
1932 |
av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
|
1933 |
return -1; |
1934 |
} |
1935 |
if (ac->m4ac.sampling_index > 12) { |
1936 |
av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
|
1937 |
return -1; |
1938 |
} |
1939 |
} |
1940 |
|
1941 |
ac->tags_mapped = 0;
|
1942 |
// parse
|
1943 |
while ((elem_type = get_bits(gb, 3)) != TYPE_END) { |
1944 |
elem_id = get_bits(gb, 4);
|
1945 |
|
1946 |
if (elem_type < TYPE_DSE) {
|
1947 |
if (!(che=get_che(ac, elem_type, elem_id))) {
|
1948 |
av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
|
1949 |
elem_type, elem_id); |
1950 |
return -1; |
1951 |
} |
1952 |
samples = 1024;
|
1953 |
} |
1954 |
|
1955 |
switch (elem_type) {
|
1956 |
|
1957 |
case TYPE_SCE:
|
1958 |
err = decode_ics(ac, &che->ch[0], gb, 0, 0); |
1959 |
break;
|
1960 |
|
1961 |
case TYPE_CPE:
|
1962 |
err = decode_cpe(ac, gb, che); |
1963 |
break;
|
1964 |
|
1965 |
case TYPE_CCE:
|
1966 |
err = decode_cce(ac, gb, che); |
1967 |
break;
|
1968 |
|
1969 |
case TYPE_LFE:
|
1970 |
err = decode_ics(ac, &che->ch[0], gb, 0, 0); |
1971 |
break;
|
1972 |
|
1973 |
case TYPE_DSE:
|
1974 |
err = skip_data_stream_element(ac, gb); |
1975 |
break;
|
1976 |
|
1977 |
case TYPE_PCE: {
|
1978 |
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
1979 |
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
1980 |
if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
|
1981 |
break;
|
1982 |
if (ac->output_configured > OC_TRIAL_PCE)
|
1983 |
av_log(avctx, AV_LOG_ERROR, |
1984 |
"Not evaluating a further program_config_element as this construct is dubious at best.\n");
|
1985 |
else
|
1986 |
err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
|
1987 |
break;
|
1988 |
} |
1989 |
|
1990 |
case TYPE_FIL:
|
1991 |
if (elem_id == 15) |
1992 |
elem_id += get_bits(gb, 8) - 1; |
1993 |
if (get_bits_left(gb) < 8 * elem_id) { |
1994 |
av_log(avctx, AV_LOG_ERROR, overread_err); |
1995 |
return -1; |
1996 |
} |
1997 |
while (elem_id > 0) |
1998 |
elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev); |
1999 |
err = 0; /* FIXME */ |
2000 |
break;
|
2001 |
|
2002 |
default:
|
2003 |
err = -1; /* should not happen, but keeps compiler happy */ |
2004 |
break;
|
2005 |
} |
2006 |
|
2007 |
che_prev = che; |
2008 |
elem_type_prev = elem_type; |
2009 |
|
2010 |
if (err)
|
2011 |
return err;
|
2012 |
|
2013 |
if (get_bits_left(gb) < 3) { |
2014 |
av_log(avctx, AV_LOG_ERROR, overread_err); |
2015 |
return -1; |
2016 |
} |
2017 |
} |
2018 |
|
2019 |
spectral_to_sample(ac); |
2020 |
|
2021 |
multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0; |
2022 |
samples <<= multiplier; |
2023 |
if (ac->output_configured < OC_LOCKED) {
|
2024 |
avctx->sample_rate = ac->m4ac.sample_rate << multiplier; |
2025 |
avctx->frame_size = samples; |
2026 |
} |
2027 |
|
2028 |
data_size_tmp = samples * avctx->channels * sizeof(int16_t);
|
2029 |
if (*data_size < data_size_tmp) {
|
2030 |
av_log(avctx, AV_LOG_ERROR, |
2031 |
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
|
2032 |
*data_size, data_size_tmp); |
2033 |
return -1; |
2034 |
} |
2035 |
*data_size = data_size_tmp; |
2036 |
|
2037 |
if (samples)
|
2038 |
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels); |
2039 |
|
2040 |
if (ac->output_configured)
|
2041 |
ac->output_configured = OC_LOCKED; |
2042 |
|
2043 |
return 0; |
2044 |
} |
2045 |
|
2046 |
static int aac_decode_frame(AVCodecContext *avctx, void *data, |
2047 |
int *data_size, AVPacket *avpkt)
|
2048 |
{ |
2049 |
const uint8_t *buf = avpkt->data;
|
2050 |
int buf_size = avpkt->size;
|
2051 |
GetBitContext gb; |
2052 |
int buf_consumed;
|
2053 |
int buf_offset;
|
2054 |
int err;
|
2055 |
|
2056 |
init_get_bits(&gb, buf, buf_size * 8);
|
2057 |
|
2058 |
if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0) |
2059 |
return err;
|
2060 |
|
2061 |
buf_consumed = (get_bits_count(&gb) + 7) >> 3; |
2062 |
for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
|
2063 |
if (buf[buf_offset])
|
2064 |
break;
|
2065 |
|
2066 |
return buf_size > buf_offset ? buf_consumed : buf_size;
|
2067 |
} |
2068 |
|
2069 |
static av_cold int aac_decode_close(AVCodecContext *avctx) |
2070 |
{ |
2071 |
AACContext *ac = avctx->priv_data; |
2072 |
int i, type;
|
2073 |
|
2074 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
2075 |
for (type = 0; type < 4; type++) { |
2076 |
if (ac->che[type][i])
|
2077 |
ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr); |
2078 |
av_freep(&ac->che[type][i]); |
2079 |
} |
2080 |
} |
2081 |
|
2082 |
ff_mdct_end(&ac->mdct); |
2083 |
ff_mdct_end(&ac->mdct_small); |
2084 |
return 0; |
2085 |
} |
2086 |
|
2087 |
|
2088 |
#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word |
2089 |
|
2090 |
struct LATMContext {
|
2091 |
AACContext aac_ctx; ///< containing AACContext
|
2092 |
int initialized; ///< initilized after a valid extradata was seen |
2093 |
|
2094 |
// parser data
|
2095 |
int audio_mux_version_A; ///< LATM syntax version |
2096 |
int frame_length_type; ///< 0/1 variable/fixed frame length |
2097 |
int frame_length; ///< frame length for fixed frame length |
2098 |
}; |
2099 |
|
2100 |
static inline uint32_t latm_get_value(GetBitContext *b) |
2101 |
{ |
2102 |
int length = get_bits(b, 2); |
2103 |
|
2104 |
return get_bits_long(b, (length+1)*8); |
2105 |
} |
2106 |
|
2107 |
static int latm_decode_audio_specific_config(struct LATMContext *latmctx, |
2108 |
GetBitContext *gb) |
2109 |
{ |
2110 |
AVCodecContext *avctx = latmctx->aac_ctx.avctx; |
2111 |
MPEG4AudioConfig m4ac; |
2112 |
int config_start_bit = get_bits_count(gb);
|
2113 |
int bits_consumed, esize;
|
2114 |
|
2115 |
if (config_start_bit % 8) { |
2116 |
av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
|
2117 |
"config not byte aligned.\n", 1); |
2118 |
return AVERROR_INVALIDDATA;
|
2119 |
} else {
|
2120 |
bits_consumed = |
2121 |
decode_audio_specific_config(NULL, avctx, &m4ac,
|
2122 |
gb->buffer + (config_start_bit / 8),
|
2123 |
get_bits_left(gb) / 8);
|
2124 |
|
2125 |
if (bits_consumed < 0) |
2126 |
return AVERROR_INVALIDDATA;
|
2127 |
|
2128 |
esize = (bits_consumed+7) / 8; |
2129 |
|
2130 |
if (avctx->extradata_size <= esize) {
|
2131 |
av_free(avctx->extradata); |
2132 |
avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE); |
2133 |
if (!avctx->extradata)
|
2134 |
return AVERROR(ENOMEM);
|
2135 |
} |
2136 |
|
2137 |
avctx->extradata_size = esize; |
2138 |
memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
|
2139 |
memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
|
2140 |
|
2141 |
skip_bits_long(gb, bits_consumed); |
2142 |
} |
2143 |
|
2144 |
return bits_consumed;
|
2145 |
} |
2146 |
|
2147 |
static int read_stream_mux_config(struct LATMContext *latmctx, |
2148 |
GetBitContext *gb) |
2149 |
{ |
2150 |
int ret, audio_mux_version = get_bits(gb, 1); |
2151 |
|
2152 |
latmctx->audio_mux_version_A = 0;
|
2153 |
if (audio_mux_version)
|
2154 |
latmctx->audio_mux_version_A = get_bits(gb, 1);
|
2155 |
|
2156 |
if (!latmctx->audio_mux_version_A) {
|
2157 |
|
2158 |
if (audio_mux_version)
|
2159 |
latm_get_value(gb); // taraFullness
|
2160 |
|
2161 |
skip_bits(gb, 1); // allStreamSameTimeFraming |
2162 |
skip_bits(gb, 6); // numSubFrames |
2163 |
// numPrograms
|
2164 |
if (get_bits(gb, 4)) { // numPrograms |
2165 |
av_log_missing_feature(latmctx->aac_ctx.avctx, |
2166 |
"multiple programs are not supported\n", 1); |
2167 |
return AVERROR_PATCHWELCOME;
|
2168 |
} |
2169 |
|
2170 |
// for each program (which there is only on in DVB)
|
2171 |
|
2172 |
// for each layer (which there is only on in DVB)
|
2173 |
if (get_bits(gb, 3)) { // numLayer |
2174 |
av_log_missing_feature(latmctx->aac_ctx.avctx, |
2175 |
"multiple layers are not supported\n", 1); |
2176 |
return AVERROR_PATCHWELCOME;
|
2177 |
} |
2178 |
|
2179 |
// for all but first stream: use_same_config = get_bits(gb, 1);
|
2180 |
if (!audio_mux_version) {
|
2181 |
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0) |
2182 |
return ret;
|
2183 |
} else {
|
2184 |
int ascLen = latm_get_value(gb);
|
2185 |
if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0) |
2186 |
return ret;
|
2187 |
ascLen -= ret; |
2188 |
skip_bits_long(gb, ascLen); |
2189 |
} |
2190 |
|
2191 |
latmctx->frame_length_type = get_bits(gb, 3);
|
2192 |
switch (latmctx->frame_length_type) {
|
2193 |
case 0: |
2194 |
skip_bits(gb, 8); // latmBufferFullness |
2195 |
break;
|
2196 |
case 1: |
2197 |
latmctx->frame_length = get_bits(gb, 9);
|
2198 |
break;
|
2199 |
case 3: |
2200 |
case 4: |
2201 |
case 5: |
2202 |
skip_bits(gb, 6); // CELP frame length table index |
2203 |
break;
|
2204 |
case 6: |
2205 |
case 7: |
2206 |
skip_bits(gb, 1); // HVXC frame length table index |
2207 |
break;
|
2208 |
} |
2209 |
|
2210 |
if (get_bits(gb, 1)) { // other data |
2211 |
if (audio_mux_version) {
|
2212 |
latm_get_value(gb); // other_data_bits
|
2213 |
} else {
|
2214 |
int esc;
|
2215 |
do {
|
2216 |
esc = get_bits(gb, 1);
|
2217 |
skip_bits(gb, 8);
|
2218 |
} while (esc);
|
2219 |
} |
2220 |
} |
2221 |
|
2222 |
if (get_bits(gb, 1)) // crc present |
2223 |
skip_bits(gb, 8); // config_crc |
2224 |
} |
2225 |
|
2226 |
return 0; |
2227 |
} |
2228 |
|
2229 |
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb) |
2230 |
{ |
2231 |
uint8_t tmp; |
2232 |
|
2233 |
if (ctx->frame_length_type == 0) { |
2234 |
int mux_slot_length = 0; |
2235 |
do {
|
2236 |
tmp = get_bits(gb, 8);
|
2237 |
mux_slot_length += tmp; |
2238 |
} while (tmp == 255); |
2239 |
return mux_slot_length;
|
2240 |
} else if (ctx->frame_length_type == 1) { |
2241 |
return ctx->frame_length;
|
2242 |
} else if (ctx->frame_length_type == 3 || |
2243 |
ctx->frame_length_type == 5 ||
|
2244 |
ctx->frame_length_type == 7) {
|
2245 |
skip_bits(gb, 2); // mux_slot_length_coded |
2246 |
} |
2247 |
return 0; |
2248 |
} |
2249 |
|
2250 |
static int read_audio_mux_element(struct LATMContext *latmctx, |
2251 |
GetBitContext *gb) |
2252 |
{ |
2253 |
int err;
|
2254 |
uint8_t use_same_mux = get_bits(gb, 1);
|
2255 |
if (!use_same_mux) {
|
2256 |
if ((err = read_stream_mux_config(latmctx, gb)) < 0) |
2257 |
return err;
|
2258 |
} else if (!latmctx->aac_ctx.avctx->extradata) { |
2259 |
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG, |
2260 |
"no decoder config found\n");
|
2261 |
return AVERROR(EAGAIN);
|
2262 |
} |
2263 |
if (latmctx->audio_mux_version_A == 0) { |
2264 |
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
|
2265 |
if (mux_slot_length_bytes * 8 > get_bits_left(gb)) { |
2266 |
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
|
2267 |
return AVERROR_INVALIDDATA;
|
2268 |
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) { |
2269 |
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, |
2270 |
"frame length mismatch %d << %d\n",
|
2271 |
mux_slot_length_bytes * 8, get_bits_left(gb));
|
2272 |
return AVERROR_INVALIDDATA;
|
2273 |
} |
2274 |
} |
2275 |
return 0; |
2276 |
} |
2277 |
|
2278 |
|
2279 |
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size, |
2280 |
AVPacket *avpkt) |
2281 |
{ |
2282 |
struct LATMContext *latmctx = avctx->priv_data;
|
2283 |
int muxlength, err;
|
2284 |
GetBitContext gb; |
2285 |
|
2286 |
if (avpkt->size == 0) |
2287 |
return 0; |
2288 |
|
2289 |
init_get_bits(&gb, avpkt->data, avpkt->size * 8);
|
2290 |
|
2291 |
// check for LOAS sync word
|
2292 |
if (get_bits(&gb, 11) != LOAS_SYNC_WORD) |
2293 |
return AVERROR_INVALIDDATA;
|
2294 |
|
2295 |
muxlength = get_bits(&gb, 13) + 3; |
2296 |
// not enough data, the parser should have sorted this
|
2297 |
if (muxlength > avpkt->size)
|
2298 |
return AVERROR_INVALIDDATA;
|
2299 |
|
2300 |
if ((err = read_audio_mux_element(latmctx, &gb)) < 0) |
2301 |
return err;
|
2302 |
|
2303 |
if (!latmctx->initialized) {
|
2304 |
if (!avctx->extradata) {
|
2305 |
*out_size = 0;
|
2306 |
return avpkt->size;
|
2307 |
} else {
|
2308 |
if ((err = aac_decode_init(avctx)) < 0) |
2309 |
return err;
|
2310 |
latmctx->initialized = 1;
|
2311 |
} |
2312 |
} |
2313 |
|
2314 |
if (show_bits(&gb, 12) == 0xfff) { |
2315 |
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, |
2316 |
"ADTS header detected, probably as result of configuration "
|
2317 |
"misparsing\n");
|
2318 |
return AVERROR_INVALIDDATA;
|
2319 |
} |
2320 |
|
2321 |
if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0) |
2322 |
return err;
|
2323 |
|
2324 |
return muxlength;
|
2325 |
} |
2326 |
|
2327 |
av_cold static int latm_decode_init(AVCodecContext *avctx) |
2328 |
{ |
2329 |
struct LATMContext *latmctx = avctx->priv_data;
|
2330 |
int ret;
|
2331 |
|
2332 |
ret = aac_decode_init(avctx); |
2333 |
|
2334 |
if (avctx->extradata_size > 0) { |
2335 |
latmctx->initialized = !ret; |
2336 |
} else {
|
2337 |
latmctx->initialized = 0;
|
2338 |
} |
2339 |
|
2340 |
return ret;
|
2341 |
} |
2342 |
|
2343 |
|
2344 |
AVCodec ff_aac_decoder = { |
2345 |
"aac",
|
2346 |
AVMEDIA_TYPE_AUDIO, |
2347 |
CODEC_ID_AAC, |
2348 |
sizeof(AACContext),
|
2349 |
aac_decode_init, |
2350 |
NULL,
|
2351 |
aac_decode_close, |
2352 |
aac_decode_frame, |
2353 |
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
|
2354 |
.sample_fmts = (const enum AVSampleFormat[]) { |
2355 |
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE |
2356 |
}, |
2357 |
.channel_layouts = aac_channel_layout, |
2358 |
}; |
2359 |
|
2360 |
/*
|
2361 |
Note: This decoder filter is intended to decode LATM streams transferred
|
2362 |
in MPEG transport streams which only contain one program.
|
2363 |
To do a more complex LATM demuxing a separate LATM demuxer should be used.
|
2364 |
*/
|
2365 |
AVCodec ff_aac_latm_decoder = { |
2366 |
.name = "aac_latm",
|
2367 |
.type = CODEC_TYPE_AUDIO, |
2368 |
.id = CODEC_ID_AAC_LATM, |
2369 |
.priv_data_size = sizeof(struct LATMContext), |
2370 |
.init = latm_decode_init, |
2371 |
.close = aac_decode_close, |
2372 |
.decode = latm_decode_frame, |
2373 |
.long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
|
2374 |
.sample_fmts = (const enum AVSampleFormat[]) { |
2375 |
AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE |
2376 |
}, |
2377 |
.channel_layouts = aac_channel_layout, |
2378 |
}; |