ffmpeg / libavcodec / aac.c @ 9106a698
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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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* 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 libavcodec/aac.c
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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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* N (in progress) 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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* N (planned) 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 "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 "mpeg4audio.h" |
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#include "aac_parser.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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union float754 { float f; uint32_t i; }; |
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static VLC vlc_scalefactors;
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static VLC vlc_spectral[11]; |
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static ChannelElement* get_che(AACContext *ac, int type, int elem_id) { |
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static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 }; |
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if (ac->tag_che_map[type][elem_id]) {
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return ac->tag_che_map[type][elem_id];
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} |
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if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
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return NULL; |
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} |
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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[0] LFE[0]. If we seem to have
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encountered such a stream, transfer the LFE[0] element to SCE[1] */
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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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* 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 int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { |
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AVCodecContext *avctx = ac->avccontext; |
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int i, type, channels = 0; |
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if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) |
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return 0; /* no change */ |
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memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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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(che_pos[type][i]) {
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if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) |
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return AVERROR(ENOMEM);
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if(type != TYPE_CCE) {
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ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
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if(type == TYPE_CPE) {
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ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
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} |
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} |
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} else
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av_freep(&ac->che[type][i]); |
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} |
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} |
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if (channel_config) {
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memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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ac->tags_mapped = 0;
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} else {
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memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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ac->tags_mapped = 4*MAX_ELEM_ID;
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} |
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avctx->channels = channels; |
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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, enum ChannelPosition type, GetBitContext * gb, int n) { |
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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(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
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GetBitContext * gb) {
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int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
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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(sampling_index > 12) { |
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
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return -1; |
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} |
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ac->m4ac.sampling_index = sampling_index; |
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ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; |
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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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skip_bits_long(gb, 8 * get_bits(gb, 8)); |
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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 int set_default_channel_config(AACContext *ac, 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(ac->avccontext, 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) |
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if(channel_config == 4) |
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new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center |
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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 |
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if(channel_config == 7) |
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new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right |
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return 0; |
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} |
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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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* @return Returns error status. 0 - OK, !0 - error
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*/
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static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { |
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enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
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int extension_flag, ret;
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|
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if(get_bits1(gb)) { // frameLengthFlag |
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ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
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return -1; |
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} |
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|
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if (get_bits1(gb)) // dependsOnCoreCoder |
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skip_bits(gb, 14); // coreCoderDelay |
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extension_flag = get_bits1(gb); |
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|
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if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
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ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) |
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skip_bits(gb, 3); // layerNr |
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|
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memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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if (channel_config == 0) { |
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skip_bits(gb, 4); // element_instance_tag |
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if((ret = decode_pce(ac, 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(ac, new_che_pos, channel_config)))
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return ret;
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} |
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if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config)))
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return ret;
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|
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if (extension_flag) {
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switch (ac->m4ac.object_type) {
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case AOT_ER_BSAC:
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skip_bits(gb, 5); // numOfSubFrame |
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skip_bits(gb, 11); // layer_length |
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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 |
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* aacScalefactorDataResilienceFlag
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* aacSpectralDataResilienceFlag
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*/
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break;
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} |
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skip_bits1(gb); // extensionFlag3 (TBD in version 3)
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} |
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return 0; |
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} |
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|
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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 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. 0 - OK, !0 - error
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*/
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static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { |
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GetBitContext gb; |
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int i;
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init_get_bits(&gb, data, data_size * 8);
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|
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if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) |
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return -1; |
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if(ac->m4ac.sampling_index > 12) { |
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
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return -1; |
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} |
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|
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skip_bits_long(&gb, i); |
| 396 |
|
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switch (ac->m4ac.object_type) {
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case AOT_AAC_MAIN:
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case AOT_AAC_LC:
|
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if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
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return -1; |
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break;
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default:
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av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
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ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); |
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return -1; |
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} |
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return 0; |
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} |
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|
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/**
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* linear congruential pseudorandom number generator
|
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*
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* @param previous_val pointer to the current state of the generator
|
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*
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* @return Returns a 32-bit pseudorandom integer
|
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*/
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static av_always_inline int lcg_random(int previous_val) { |
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return previous_val * 1664525 + 1013904223; |
| 420 |
} |
| 421 |
|
| 422 |
static void reset_predict_state(PredictorState * ps) { |
| 423 |
ps->r0 = 0.0f; |
| 424 |
ps->r1 = 0.0f; |
| 425 |
ps->cor0 = 0.0f; |
| 426 |
ps->cor1 = 0.0f; |
| 427 |
ps->var0 = 1.0f; |
| 428 |
ps->var1 = 1.0f; |
| 429 |
} |
| 430 |
|
| 431 |
static void reset_all_predictors(PredictorState * ps) { |
| 432 |
int i;
|
| 433 |
for (i = 0; i < MAX_PREDICTORS; i++) |
| 434 |
reset_predict_state(&ps[i]); |
| 435 |
} |
| 436 |
|
| 437 |
static void reset_predictor_group(PredictorState * ps, int group_num) { |
| 438 |
int i;
|
| 439 |
for (i = group_num-1; i < MAX_PREDICTORS; i+=30) |
| 440 |
reset_predict_state(&ps[i]); |
| 441 |
} |
| 442 |
|
| 443 |
static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
| 444 |
AACContext * ac = avccontext->priv_data; |
| 445 |
int i;
|
| 446 |
|
| 447 |
ac->avccontext = avccontext; |
| 448 |
|
| 449 |
if (avccontext->extradata_size > 0) { |
| 450 |
if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
|
| 451 |
return -1; |
| 452 |
avccontext->sample_rate = ac->m4ac.sample_rate; |
| 453 |
} else if (avccontext->channels > 0) { |
| 454 |
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
| 455 |
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| 456 |
if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8))) |
| 457 |
return -1; |
| 458 |
if(output_configure(ac, ac->che_pos, new_che_pos, 1)) |
| 459 |
return -1; |
| 460 |
ac->m4ac.sample_rate = avccontext->sample_rate; |
| 461 |
} else {
|
| 462 |
ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0); |
| 463 |
return -1; |
| 464 |
} |
| 465 |
|
| 466 |
avccontext->sample_fmt = SAMPLE_FMT_S16; |
| 467 |
avccontext->frame_size = 1024;
|
| 468 |
|
| 469 |
AAC_INIT_VLC_STATIC( 0, 144); |
| 470 |
AAC_INIT_VLC_STATIC( 1, 114); |
| 471 |
AAC_INIT_VLC_STATIC( 2, 188); |
| 472 |
AAC_INIT_VLC_STATIC( 3, 180); |
| 473 |
AAC_INIT_VLC_STATIC( 4, 172); |
| 474 |
AAC_INIT_VLC_STATIC( 5, 140); |
| 475 |
AAC_INIT_VLC_STATIC( 6, 168); |
| 476 |
AAC_INIT_VLC_STATIC( 7, 114); |
| 477 |
AAC_INIT_VLC_STATIC( 8, 262); |
| 478 |
AAC_INIT_VLC_STATIC( 9, 248); |
| 479 |
AAC_INIT_VLC_STATIC(10, 384); |
| 480 |
|
| 481 |
dsputil_init(&ac->dsp, avccontext); |
| 482 |
|
| 483 |
ac->random_state = 0x1f2e3d4c;
|
| 484 |
|
| 485 |
// -1024 - Compensate wrong IMDCT method.
|
| 486 |
// 32768 - Required to scale values to the correct range for the bias method
|
| 487 |
// for float to int16 conversion.
|
| 488 |
|
| 489 |
if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
|
| 490 |
ac->add_bias = 385.0f; |
| 491 |
ac->sf_scale = 1. / (-1024. * 32768.); |
| 492 |
ac->sf_offset = 0;
|
| 493 |
} else {
|
| 494 |
ac->add_bias = 0.0f; |
| 495 |
ac->sf_scale = 1. / -1024.; |
| 496 |
ac->sf_offset = 60;
|
| 497 |
} |
| 498 |
|
| 499 |
#if !CONFIG_HARDCODED_TABLES
|
| 500 |
for (i = 0; i < 428; i++) |
| 501 |
ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
| 502 |
#endif /* CONFIG_HARDCODED_TABLES */ |
| 503 |
|
| 504 |
INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
|
| 505 |
ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
| 506 |
ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), |
| 507 |
352);
|
| 508 |
|
| 509 |
ff_mdct_init(&ac->mdct, 11, 1); |
| 510 |
ff_mdct_init(&ac->mdct_small, 8, 1); |
| 511 |
// window initialization
|
| 512 |
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); |
| 513 |
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); |
| 514 |
ff_sine_window_init(ff_sine_1024, 1024);
|
| 515 |
ff_sine_window_init(ff_sine_128, 128);
|
| 516 |
|
| 517 |
return 0; |
| 518 |
} |
| 519 |
|
| 520 |
/**
|
| 521 |
* Skip data_stream_element; reference: table 4.10.
|
| 522 |
*/
|
| 523 |
static void skip_data_stream_element(GetBitContext * gb) { |
| 524 |
int byte_align = get_bits1(gb);
|
| 525 |
int count = get_bits(gb, 8); |
| 526 |
if (count == 255) |
| 527 |
count += get_bits(gb, 8);
|
| 528 |
if (byte_align)
|
| 529 |
align_get_bits(gb); |
| 530 |
skip_bits_long(gb, 8 * count);
|
| 531 |
} |
| 532 |
|
| 533 |
static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { |
| 534 |
int sfb;
|
| 535 |
if (get_bits1(gb)) {
|
| 536 |
ics->predictor_reset_group = get_bits(gb, 5);
|
| 537 |
if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
| 538 |
av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
|
| 539 |
return -1; |
| 540 |
} |
| 541 |
} |
| 542 |
for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
| 543 |
ics->prediction_used[sfb] = get_bits1(gb); |
| 544 |
} |
| 545 |
return 0; |
| 546 |
} |
| 547 |
|
| 548 |
/**
|
| 549 |
* Decode Individual Channel Stream info; reference: table 4.6.
|
| 550 |
*
|
| 551 |
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
| 552 |
*/
|
| 553 |
static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { |
| 554 |
if (get_bits1(gb)) {
|
| 555 |
av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
|
| 556 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
| 557 |
return -1; |
| 558 |
} |
| 559 |
ics->window_sequence[1] = ics->window_sequence[0]; |
| 560 |
ics->window_sequence[0] = get_bits(gb, 2); |
| 561 |
ics->use_kb_window[1] = ics->use_kb_window[0]; |
| 562 |
ics->use_kb_window[0] = get_bits1(gb);
|
| 563 |
ics->num_window_groups = 1;
|
| 564 |
ics->group_len[0] = 1; |
| 565 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 566 |
int i;
|
| 567 |
ics->max_sfb = get_bits(gb, 4);
|
| 568 |
for (i = 0; i < 7; i++) { |
| 569 |
if (get_bits1(gb)) {
|
| 570 |
ics->group_len[ics->num_window_groups-1]++;
|
| 571 |
} else {
|
| 572 |
ics->num_window_groups++; |
| 573 |
ics->group_len[ics->num_window_groups-1] = 1; |
| 574 |
} |
| 575 |
} |
| 576 |
ics->num_windows = 8;
|
| 577 |
ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; |
| 578 |
ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; |
| 579 |
ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; |
| 580 |
ics->predictor_present = 0;
|
| 581 |
} else {
|
| 582 |
ics->max_sfb = get_bits(gb, 6);
|
| 583 |
ics->num_windows = 1;
|
| 584 |
ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; |
| 585 |
ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; |
| 586 |
ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; |
| 587 |
ics->predictor_present = get_bits1(gb); |
| 588 |
ics->predictor_reset_group = 0;
|
| 589 |
if (ics->predictor_present) {
|
| 590 |
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
| 591 |
if (decode_prediction(ac, ics, gb)) {
|
| 592 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
| 593 |
return -1; |
| 594 |
} |
| 595 |
} else if (ac->m4ac.object_type == AOT_AAC_LC) { |
| 596 |
av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
|
| 597 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
| 598 |
return -1; |
| 599 |
} else {
|
| 600 |
ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
| 601 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
| 602 |
return -1; |
| 603 |
} |
| 604 |
} |
| 605 |
} |
| 606 |
|
| 607 |
if(ics->max_sfb > ics->num_swb) {
|
| 608 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 609 |
"Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
|
| 610 |
ics->max_sfb, ics->num_swb); |
| 611 |
memset(ics, 0, sizeof(IndividualChannelStream)); |
| 612 |
return -1; |
| 613 |
} |
| 614 |
|
| 615 |
return 0; |
| 616 |
} |
| 617 |
|
| 618 |
/**
|
| 619 |
* Decode band types (section_data payload); reference: table 4.46.
|
| 620 |
*
|
| 621 |
* @param band_type array of the used band type
|
| 622 |
* @param band_type_run_end array of the last scalefactor band of a band type run
|
| 623 |
*
|
| 624 |
* @return Returns error status. 0 - OK, !0 - error
|
| 625 |
*/
|
| 626 |
static int decode_band_types(AACContext * ac, enum BandType band_type[120], |
| 627 |
int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
| 628 |
int g, idx = 0; |
| 629 |
const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
| 630 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 631 |
int k = 0; |
| 632 |
while (k < ics->max_sfb) {
|
| 633 |
uint8_t sect_len = k; |
| 634 |
int sect_len_incr;
|
| 635 |
int sect_band_type = get_bits(gb, 4); |
| 636 |
if (sect_band_type == 12) { |
| 637 |
av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
|
| 638 |
return -1; |
| 639 |
} |
| 640 |
while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
| 641 |
sect_len += sect_len_incr; |
| 642 |
sect_len += sect_len_incr; |
| 643 |
if (sect_len > ics->max_sfb) {
|
| 644 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 645 |
"Number of bands (%d) exceeds limit (%d).\n",
|
| 646 |
sect_len, ics->max_sfb); |
| 647 |
return -1; |
| 648 |
} |
| 649 |
for (; k < sect_len; k++) {
|
| 650 |
band_type [idx] = sect_band_type; |
| 651 |
band_type_run_end[idx++] = sect_len; |
| 652 |
} |
| 653 |
} |
| 654 |
} |
| 655 |
return 0; |
| 656 |
} |
| 657 |
|
| 658 |
/**
|
| 659 |
* Decode scalefactors; reference: table 4.47.
|
| 660 |
*
|
| 661 |
* @param global_gain first scalefactor value as scalefactors are differentially coded
|
| 662 |
* @param band_type array of the used band type
|
| 663 |
* @param band_type_run_end array of the last scalefactor band of a band type run
|
| 664 |
* @param sf array of scalefactors or intensity stereo positions
|
| 665 |
*
|
| 666 |
* @return Returns error status. 0 - OK, !0 - error
|
| 667 |
*/
|
| 668 |
static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
| 669 |
unsigned int global_gain, IndividualChannelStream * ics, |
| 670 |
enum BandType band_type[120], int band_type_run_end[120]) { |
| 671 |
const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
| 672 |
int g, i, idx = 0; |
| 673 |
int offset[3] = { global_gain, global_gain - 90, 100 }; |
| 674 |
int noise_flag = 1; |
| 675 |
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
| 676 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 677 |
for (i = 0; i < ics->max_sfb;) { |
| 678 |
int run_end = band_type_run_end[idx];
|
| 679 |
if (band_type[idx] == ZERO_BT) {
|
| 680 |
for(; i < run_end; i++, idx++)
|
| 681 |
sf[idx] = 0.;
|
| 682 |
}else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
| 683 |
for(; i < run_end; i++, idx++) {
|
| 684 |
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| 685 |
if(offset[2] > 255U) { |
| 686 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 687 |
"%s (%d) out of range.\n", sf_str[2], offset[2]); |
| 688 |
return -1; |
| 689 |
} |
| 690 |
sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
| 691 |
} |
| 692 |
}else if(band_type[idx] == NOISE_BT) { |
| 693 |
for(; i < run_end; i++, idx++) {
|
| 694 |
if(noise_flag-- > 0) |
| 695 |
offset[1] += get_bits(gb, 9) - 256; |
| 696 |
else
|
| 697 |
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| 698 |
if(offset[1] > 255U) { |
| 699 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 700 |
"%s (%d) out of range.\n", sf_str[1], offset[1]); |
| 701 |
return -1; |
| 702 |
} |
| 703 |
sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; |
| 704 |
} |
| 705 |
}else {
|
| 706 |
for(; i < run_end; i++, idx++) {
|
| 707 |
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| 708 |
if(offset[0] > 255U) { |
| 709 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 710 |
"%s (%d) out of range.\n", sf_str[0], offset[0]); |
| 711 |
return -1; |
| 712 |
} |
| 713 |
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
|
| 714 |
} |
| 715 |
} |
| 716 |
} |
| 717 |
} |
| 718 |
return 0; |
| 719 |
} |
| 720 |
|
| 721 |
/**
|
| 722 |
* Decode pulse data; reference: table 4.7.
|
| 723 |
*/
|
| 724 |
static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
| 725 |
int i, pulse_swb;
|
| 726 |
pulse->num_pulse = get_bits(gb, 2) + 1; |
| 727 |
pulse_swb = get_bits(gb, 6);
|
| 728 |
if (pulse_swb >= num_swb)
|
| 729 |
return -1; |
| 730 |
pulse->pos[0] = swb_offset[pulse_swb];
|
| 731 |
pulse->pos[0] += get_bits(gb, 5); |
| 732 |
if (pulse->pos[0] > 1023) |
| 733 |
return -1; |
| 734 |
pulse->amp[0] = get_bits(gb, 4); |
| 735 |
for (i = 1; i < pulse->num_pulse; i++) { |
| 736 |
pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; |
| 737 |
if (pulse->pos[i] > 1023) |
| 738 |
return -1; |
| 739 |
pulse->amp[i] = get_bits(gb, 4);
|
| 740 |
} |
| 741 |
return 0; |
| 742 |
} |
| 743 |
|
| 744 |
/**
|
| 745 |
* Decode Temporal Noise Shaping data; reference: table 4.48.
|
| 746 |
*
|
| 747 |
* @return Returns error status. 0 - OK, !0 - error
|
| 748 |
*/
|
| 749 |
static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, |
| 750 |
GetBitContext * gb, const IndividualChannelStream * ics) {
|
| 751 |
int w, filt, i, coef_len, coef_res, coef_compress;
|
| 752 |
const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; |
| 753 |
const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; |
| 754 |
for (w = 0; w < ics->num_windows; w++) { |
| 755 |
if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
| 756 |
coef_res = get_bits1(gb); |
| 757 |
|
| 758 |
for (filt = 0; filt < tns->n_filt[w]; filt++) { |
| 759 |
int tmp2_idx;
|
| 760 |
tns->length[w][filt] = get_bits(gb, 6 - 2*is8); |
| 761 |
|
| 762 |
if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
| 763 |
av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
|
| 764 |
tns->order[w][filt], tns_max_order); |
| 765 |
tns->order[w][filt] = 0;
|
| 766 |
return -1; |
| 767 |
} |
| 768 |
if (tns->order[w][filt]) {
|
| 769 |
tns->direction[w][filt] = get_bits1(gb); |
| 770 |
coef_compress = get_bits1(gb); |
| 771 |
coef_len = coef_res + 3 - coef_compress;
|
| 772 |
tmp2_idx = 2*coef_compress + coef_res;
|
| 773 |
|
| 774 |
for (i = 0; i < tns->order[w][filt]; i++) |
| 775 |
tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; |
| 776 |
} |
| 777 |
} |
| 778 |
} |
| 779 |
} |
| 780 |
return 0; |
| 781 |
} |
| 782 |
|
| 783 |
/**
|
| 784 |
* Decode Mid/Side data; reference: table 4.54.
|
| 785 |
*
|
| 786 |
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
| 787 |
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
| 788 |
* [3] reserved for scalable AAC
|
| 789 |
*/
|
| 790 |
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, |
| 791 |
int ms_present) {
|
| 792 |
int idx;
|
| 793 |
if (ms_present == 1) { |
| 794 |
for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) |
| 795 |
cpe->ms_mask[idx] = get_bits1(gb); |
| 796 |
} else if (ms_present == 2) { |
| 797 |
memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); |
| 798 |
} |
| 799 |
} |
| 800 |
|
| 801 |
/**
|
| 802 |
* Decode spectral data; reference: table 4.50.
|
| 803 |
* Dequantize and scale spectral data; reference: 4.6.3.3.
|
| 804 |
*
|
| 805 |
* @param coef array of dequantized, scaled spectral data
|
| 806 |
* @param sf array of scalefactors or intensity stereo positions
|
| 807 |
* @param pulse_present set if pulses are present
|
| 808 |
* @param pulse pointer to pulse data struct
|
| 809 |
* @param band_type array of the used band type
|
| 810 |
*
|
| 811 |
* @return Returns error status. 0 - OK, !0 - error
|
| 812 |
*/
|
| 813 |
static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], |
| 814 |
int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { |
| 815 |
int i, k, g, idx = 0; |
| 816 |
const int c = 1024/ics->num_windows; |
| 817 |
const uint16_t * offsets = ics->swb_offset;
|
| 818 |
float *coef_base = coef;
|
| 819 |
static const float sign_lookup[] = { 1.0f, -1.0f }; |
| 820 |
|
| 821 |
for (g = 0; g < ics->num_windows; g++) |
| 822 |
memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); |
| 823 |
|
| 824 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 825 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
| 826 |
const int cur_band_type = band_type[idx]; |
| 827 |
const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; |
| 828 |
const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); |
| 829 |
int group;
|
| 830 |
if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
|
| 831 |
for (group = 0; group < ics->group_len[g]; group++) { |
| 832 |
memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); |
| 833 |
} |
| 834 |
}else if (cur_band_type == NOISE_BT) { |
| 835 |
for (group = 0; group < ics->group_len[g]; group++) { |
| 836 |
float scale;
|
| 837 |
float band_energy = 0; |
| 838 |
for (k = offsets[i]; k < offsets[i+1]; k++) { |
| 839 |
ac->random_state = lcg_random(ac->random_state); |
| 840 |
coef[group*128+k] = ac->random_state;
|
| 841 |
band_energy += coef[group*128+k]*coef[group*128+k]; |
| 842 |
} |
| 843 |
scale = sf[idx] / sqrtf(band_energy); |
| 844 |
for (k = offsets[i]; k < offsets[i+1]; k++) { |
| 845 |
coef[group*128+k] *= scale;
|
| 846 |
} |
| 847 |
} |
| 848 |
}else {
|
| 849 |
for (group = 0; group < ics->group_len[g]; group++) { |
| 850 |
for (k = offsets[i]; k < offsets[i+1]; k += dim) { |
| 851 |
const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); |
| 852 |
const int coef_tmp_idx = (group << 7) + k; |
| 853 |
const float *vq_ptr; |
| 854 |
int j;
|
| 855 |
if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { |
| 856 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 857 |
"Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
|
| 858 |
cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); |
| 859 |
return -1; |
| 860 |
} |
| 861 |
vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
|
| 862 |
if (is_cb_unsigned) {
|
| 863 |
if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; |
| 864 |
if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; |
| 865 |
if (dim == 4) { |
| 866 |
if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; |
| 867 |
if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; |
| 868 |
} |
| 869 |
if (cur_band_type == ESC_BT) {
|
| 870 |
for (j = 0; j < 2; j++) { |
| 871 |
if (vq_ptr[j] == 64.0f) { |
| 872 |
int n = 4; |
| 873 |
/* The total length of escape_sequence must be < 22 bits according
|
| 874 |
to the specification (i.e. max is 11111111110xxxxxxxxxx). */
|
| 875 |
while (get_bits1(gb) && n < 15) n++; |
| 876 |
if(n == 15) { |
| 877 |
av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
|
| 878 |
return -1; |
| 879 |
} |
| 880 |
n = (1<<n) + get_bits(gb, n);
|
| 881 |
coef[coef_tmp_idx + j] *= cbrtf(n) * n; |
| 882 |
}else
|
| 883 |
coef[coef_tmp_idx + j] *= vq_ptr[j]; |
| 884 |
} |
| 885 |
}else
|
| 886 |
{
|
| 887 |
coef[coef_tmp_idx ] *= vq_ptr[0];
|
| 888 |
coef[coef_tmp_idx + 1] *= vq_ptr[1]; |
| 889 |
if (dim == 4) { |
| 890 |
coef[coef_tmp_idx + 2] *= vq_ptr[2]; |
| 891 |
coef[coef_tmp_idx + 3] *= vq_ptr[3]; |
| 892 |
} |
| 893 |
} |
| 894 |
}else {
|
| 895 |
coef[coef_tmp_idx ] = vq_ptr[0];
|
| 896 |
coef[coef_tmp_idx + 1] = vq_ptr[1]; |
| 897 |
if (dim == 4) { |
| 898 |
coef[coef_tmp_idx + 2] = vq_ptr[2]; |
| 899 |
coef[coef_tmp_idx + 3] = vq_ptr[3]; |
| 900 |
} |
| 901 |
} |
| 902 |
coef[coef_tmp_idx ] *= sf[idx]; |
| 903 |
coef[coef_tmp_idx + 1] *= sf[idx];
|
| 904 |
if (dim == 4) { |
| 905 |
coef[coef_tmp_idx + 2] *= sf[idx];
|
| 906 |
coef[coef_tmp_idx + 3] *= sf[idx];
|
| 907 |
} |
| 908 |
} |
| 909 |
} |
| 910 |
} |
| 911 |
} |
| 912 |
coef += ics->group_len[g]<<7;
|
| 913 |
} |
| 914 |
|
| 915 |
if (pulse_present) {
|
| 916 |
idx = 0;
|
| 917 |
for(i = 0; i < pulse->num_pulse; i++){ |
| 918 |
float co = coef_base[ pulse->pos[i] ];
|
| 919 |
while(offsets[idx + 1] <= pulse->pos[i]) |
| 920 |
idx++; |
| 921 |
if (band_type[idx] != NOISE_BT && sf[idx]) {
|
| 922 |
float ico = -pulse->amp[i];
|
| 923 |
if (co) {
|
| 924 |
co /= sf[idx]; |
| 925 |
ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
|
| 926 |
} |
| 927 |
coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; |
| 928 |
} |
| 929 |
} |
| 930 |
} |
| 931 |
return 0; |
| 932 |
} |
| 933 |
|
| 934 |
static av_always_inline float flt16_round(float pf) { |
| 935 |
union float754 tmp;
|
| 936 |
tmp.f = pf; |
| 937 |
tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
| 938 |
return tmp.f;
|
| 939 |
} |
| 940 |
|
| 941 |
static av_always_inline float flt16_even(float pf) { |
| 942 |
union float754 tmp;
|
| 943 |
tmp.f = pf; |
| 944 |
tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U>>16)) & 0xFFFF0000U; |
| 945 |
return tmp.f;
|
| 946 |
} |
| 947 |
|
| 948 |
static av_always_inline float flt16_trunc(float pf) { |
| 949 |
union float754 pun;
|
| 950 |
pun.f = pf; |
| 951 |
pun.i &= 0xFFFF0000U;
|
| 952 |
return pun.f;
|
| 953 |
} |
| 954 |
|
| 955 |
static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { |
| 956 |
const float a = 0.953125; // 61.0/64 |
| 957 |
const float alpha = 0.90625; // 29.0/32 |
| 958 |
float e0, e1;
|
| 959 |
float pv;
|
| 960 |
float k1, k2;
|
| 961 |
|
| 962 |
k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
| 963 |
k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
| 964 |
|
| 965 |
pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
| 966 |
if (output_enable)
|
| 967 |
*coef += pv * ac->sf_scale; |
| 968 |
|
| 969 |
e0 = *coef / ac->sf_scale; |
| 970 |
e1 = e0 - k1 * ps->r0; |
| 971 |
|
| 972 |
ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); |
| 973 |
ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); |
| 974 |
ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
| 975 |
ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
| 976 |
|
| 977 |
ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
| 978 |
ps->r0 = flt16_trunc(a * e0); |
| 979 |
} |
| 980 |
|
| 981 |
/**
|
| 982 |
* Apply AAC-Main style frequency domain prediction.
|
| 983 |
*/
|
| 984 |
static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { |
| 985 |
int sfb, k;
|
| 986 |
|
| 987 |
if (!sce->ics.predictor_initialized) {
|
| 988 |
reset_all_predictors(sce->predictor_state); |
| 989 |
sce->ics.predictor_initialized = 1;
|
| 990 |
} |
| 991 |
|
| 992 |
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
| 993 |
for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
| 994 |
for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
| 995 |
predict(ac, &sce->predictor_state[k], &sce->coeffs[k], |
| 996 |
sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
| 997 |
} |
| 998 |
} |
| 999 |
if (sce->ics.predictor_reset_group)
|
| 1000 |
reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
| 1001 |
} else
|
| 1002 |
reset_all_predictors(sce->predictor_state); |
| 1003 |
} |
| 1004 |
|
| 1005 |
/**
|
| 1006 |
* Decode an individual_channel_stream payload; reference: table 4.44.
|
| 1007 |
*
|
| 1008 |
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
| 1009 |
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
|
| 1010 |
*
|
| 1011 |
* @return Returns error status. 0 - OK, !0 - error
|
| 1012 |
*/
|
| 1013 |
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { |
| 1014 |
Pulse pulse; |
| 1015 |
TemporalNoiseShaping * tns = &sce->tns; |
| 1016 |
IndividualChannelStream * ics = &sce->ics; |
| 1017 |
float * out = sce->coeffs;
|
| 1018 |
int global_gain, pulse_present = 0; |
| 1019 |
|
| 1020 |
/* This assignment is to silence a GCC warning about the variable being used
|
| 1021 |
* uninitialized when in fact it always is.
|
| 1022 |
*/
|
| 1023 |
pulse.num_pulse = 0;
|
| 1024 |
|
| 1025 |
global_gain = get_bits(gb, 8);
|
| 1026 |
|
| 1027 |
if (!common_window && !scale_flag) {
|
| 1028 |
if (decode_ics_info(ac, ics, gb, 0) < 0) |
| 1029 |
return -1; |
| 1030 |
} |
| 1031 |
|
| 1032 |
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) |
| 1033 |
return -1; |
| 1034 |
if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) |
| 1035 |
return -1; |
| 1036 |
|
| 1037 |
pulse_present = 0;
|
| 1038 |
if (!scale_flag) {
|
| 1039 |
if ((pulse_present = get_bits1(gb))) {
|
| 1040 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 1041 |
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
|
| 1042 |
return -1; |
| 1043 |
} |
| 1044 |
if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
|
| 1045 |
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
|
| 1046 |
return -1; |
| 1047 |
} |
| 1048 |
} |
| 1049 |
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
|
| 1050 |
return -1; |
| 1051 |
if (get_bits1(gb)) {
|
| 1052 |
ff_log_missing_feature(ac->avccontext, "SSR", 1); |
| 1053 |
return -1; |
| 1054 |
} |
| 1055 |
} |
| 1056 |
|
| 1057 |
if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
| 1058 |
return -1; |
| 1059 |
|
| 1060 |
if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
|
| 1061 |
apply_prediction(ac, sce); |
| 1062 |
|
| 1063 |
return 0; |
| 1064 |
} |
| 1065 |
|
| 1066 |
/**
|
| 1067 |
* Mid/Side stereo decoding; reference: 4.6.8.1.3.
|
| 1068 |
*/
|
| 1069 |
static void apply_mid_side_stereo(ChannelElement * cpe) { |
| 1070 |
const IndividualChannelStream * ics = &cpe->ch[0].ics; |
| 1071 |
float *ch0 = cpe->ch[0].coeffs; |
| 1072 |
float *ch1 = cpe->ch[1].coeffs; |
| 1073 |
int g, i, k, group, idx = 0; |
| 1074 |
const uint16_t * offsets = ics->swb_offset;
|
| 1075 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 1076 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
| 1077 |
if (cpe->ms_mask[idx] &&
|
| 1078 |
cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { |
| 1079 |
for (group = 0; group < ics->group_len[g]; group++) { |
| 1080 |
for (k = offsets[i]; k < offsets[i+1]; k++) { |
| 1081 |
float tmp = ch0[group*128 + k] - ch1[group*128 + k]; |
| 1082 |
ch0[group*128 + k] += ch1[group*128 + k]; |
| 1083 |
ch1[group*128 + k] = tmp;
|
| 1084 |
} |
| 1085 |
} |
| 1086 |
} |
| 1087 |
} |
| 1088 |
ch0 += ics->group_len[g]*128;
|
| 1089 |
ch1 += ics->group_len[g]*128;
|
| 1090 |
} |
| 1091 |
} |
| 1092 |
|
| 1093 |
/**
|
| 1094 |
* intensity stereo decoding; reference: 4.6.8.2.3
|
| 1095 |
*
|
| 1096 |
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
| 1097 |
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
| 1098 |
* [3] reserved for scalable AAC
|
| 1099 |
*/
|
| 1100 |
static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { |
| 1101 |
const IndividualChannelStream * ics = &cpe->ch[1].ics; |
| 1102 |
SingleChannelElement * sce1 = &cpe->ch[1];
|
| 1103 |
float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; |
| 1104 |
const uint16_t * offsets = ics->swb_offset;
|
| 1105 |
int g, group, i, k, idx = 0; |
| 1106 |
int c;
|
| 1107 |
float scale;
|
| 1108 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 1109 |
for (i = 0; i < ics->max_sfb;) { |
| 1110 |
if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
|
| 1111 |
const int bt_run_end = sce1->band_type_run_end[idx]; |
| 1112 |
for (; i < bt_run_end; i++, idx++) {
|
| 1113 |
c = -1 + 2 * (sce1->band_type[idx] - 14); |
| 1114 |
if (ms_present)
|
| 1115 |
c *= 1 - 2 * cpe->ms_mask[idx]; |
| 1116 |
scale = c * sce1->sf[idx]; |
| 1117 |
for (group = 0; group < ics->group_len[g]; group++) |
| 1118 |
for (k = offsets[i]; k < offsets[i+1]; k++) |
| 1119 |
coef1[group*128 + k] = scale * coef0[group*128 + k]; |
| 1120 |
} |
| 1121 |
} else {
|
| 1122 |
int bt_run_end = sce1->band_type_run_end[idx];
|
| 1123 |
idx += bt_run_end - i; |
| 1124 |
i = bt_run_end; |
| 1125 |
} |
| 1126 |
} |
| 1127 |
coef0 += ics->group_len[g]*128;
|
| 1128 |
coef1 += ics->group_len[g]*128;
|
| 1129 |
} |
| 1130 |
} |
| 1131 |
|
| 1132 |
/**
|
| 1133 |
* Decode a channel_pair_element; reference: table 4.4.
|
| 1134 |
*
|
| 1135 |
* @param elem_id Identifies the instance of a syntax element.
|
| 1136 |
*
|
| 1137 |
* @return Returns error status. 0 - OK, !0 - error
|
| 1138 |
*/
|
| 1139 |
static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) { |
| 1140 |
int i, ret, common_window, ms_present = 0; |
| 1141 |
|
| 1142 |
common_window = get_bits1(gb); |
| 1143 |
if (common_window) {
|
| 1144 |
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) |
| 1145 |
return -1; |
| 1146 |
i = cpe->ch[1].ics.use_kb_window[0]; |
| 1147 |
cpe->ch[1].ics = cpe->ch[0].ics; |
| 1148 |
cpe->ch[1].ics.use_kb_window[1] = i; |
| 1149 |
ms_present = get_bits(gb, 2);
|
| 1150 |
if(ms_present == 3) { |
| 1151 |
av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
|
| 1152 |
return -1; |
| 1153 |
} else if(ms_present) |
| 1154 |
decode_mid_side_stereo(cpe, gb, ms_present); |
| 1155 |
} |
| 1156 |
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) |
| 1157 |
return ret;
|
| 1158 |
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) |
| 1159 |
return ret;
|
| 1160 |
|
| 1161 |
if (common_window) {
|
| 1162 |
if (ms_present)
|
| 1163 |
apply_mid_side_stereo(cpe); |
| 1164 |
if (ac->m4ac.object_type == AOT_AAC_MAIN) {
|
| 1165 |
apply_prediction(ac, &cpe->ch[0]);
|
| 1166 |
apply_prediction(ac, &cpe->ch[1]);
|
| 1167 |
} |
| 1168 |
} |
| 1169 |
|
| 1170 |
apply_intensity_stereo(cpe, ms_present); |
| 1171 |
return 0; |
| 1172 |
} |
| 1173 |
|
| 1174 |
/**
|
| 1175 |
* Decode coupling_channel_element; reference: table 4.8.
|
| 1176 |
*
|
| 1177 |
* @param elem_id Identifies the instance of a syntax element.
|
| 1178 |
*
|
| 1179 |
* @return Returns error status. 0 - OK, !0 - error
|
| 1180 |
*/
|
| 1181 |
static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { |
| 1182 |
int num_gain = 0; |
| 1183 |
int c, g, sfb, ret;
|
| 1184 |
int sign;
|
| 1185 |
float scale;
|
| 1186 |
SingleChannelElement * sce = &che->ch[0];
|
| 1187 |
ChannelCoupling * coup = &che->coup; |
| 1188 |
|
| 1189 |
coup->coupling_point = 2*get_bits1(gb);
|
| 1190 |
coup->num_coupled = get_bits(gb, 3);
|
| 1191 |
for (c = 0; c <= coup->num_coupled; c++) { |
| 1192 |
num_gain++; |
| 1193 |
coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; |
| 1194 |
coup->id_select[c] = get_bits(gb, 4);
|
| 1195 |
if (coup->type[c] == TYPE_CPE) {
|
| 1196 |
coup->ch_select[c] = get_bits(gb, 2);
|
| 1197 |
if (coup->ch_select[c] == 3) |
| 1198 |
num_gain++; |
| 1199 |
} else
|
| 1200 |
coup->ch_select[c] = 2;
|
| 1201 |
} |
| 1202 |
coup->coupling_point += get_bits1(gb) || (coup->coupling_point>>1);
|
| 1203 |
|
| 1204 |
sign = get_bits(gb, 1);
|
| 1205 |
scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
| 1206 |
|
| 1207 |
if ((ret = decode_ics(ac, sce, gb, 0, 0))) |
| 1208 |
return ret;
|
| 1209 |
|
| 1210 |
for (c = 0; c < num_gain; c++) { |
| 1211 |
int idx = 0; |
| 1212 |
int cge = 1; |
| 1213 |
int gain = 0; |
| 1214 |
float gain_cache = 1.; |
| 1215 |
if (c) {
|
| 1216 |
cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
|
| 1217 |
gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; |
| 1218 |
gain_cache = pow(scale, -gain); |
| 1219 |
} |
| 1220 |
if (coup->coupling_point == AFTER_IMDCT) {
|
| 1221 |
coup->gain[c][0] = gain_cache;
|
| 1222 |
} else {
|
| 1223 |
for (g = 0; g < sce->ics.num_window_groups; g++) { |
| 1224 |
for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { |
| 1225 |
if (sce->band_type[idx] != ZERO_BT) {
|
| 1226 |
if (!cge) {
|
| 1227 |
int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
| 1228 |
if (t) {
|
| 1229 |
int s = 1; |
| 1230 |
t = gain += t; |
| 1231 |
if (sign) {
|
| 1232 |
s -= 2 * (t & 0x1); |
| 1233 |
t >>= 1;
|
| 1234 |
} |
| 1235 |
gain_cache = pow(scale, -t) * s; |
| 1236 |
} |
| 1237 |
} |
| 1238 |
coup->gain[c][idx] = gain_cache; |
| 1239 |
} |
| 1240 |
} |
| 1241 |
} |
| 1242 |
} |
| 1243 |
} |
| 1244 |
return 0; |
| 1245 |
} |
| 1246 |
|
| 1247 |
/**
|
| 1248 |
* Decode Spectral Band Replication extension data; reference: table 4.55.
|
| 1249 |
*
|
| 1250 |
* @param crc flag indicating the presence of CRC checksum
|
| 1251 |
* @param cnt length of TYPE_FIL syntactic element in bytes
|
| 1252 |
*
|
| 1253 |
* @return Returns number of bytes consumed from the TYPE_FIL element.
|
| 1254 |
*/
|
| 1255 |
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
| 1256 |
// TODO : sbr_extension implementation
|
| 1257 |
ff_log_missing_feature(ac->avccontext, "SBR", 0); |
| 1258 |
skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
| 1259 |
return cnt;
|
| 1260 |
} |
| 1261 |
|
| 1262 |
/**
|
| 1263 |
* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
|
| 1264 |
*
|
| 1265 |
* @return Returns number of bytes consumed.
|
| 1266 |
*/
|
| 1267 |
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { |
| 1268 |
int i;
|
| 1269 |
int num_excl_chan = 0; |
| 1270 |
|
| 1271 |
do {
|
| 1272 |
for (i = 0; i < 7; i++) |
| 1273 |
che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); |
| 1274 |
} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); |
| 1275 |
|
| 1276 |
return num_excl_chan / 7; |
| 1277 |
} |
| 1278 |
|
| 1279 |
/**
|
| 1280 |
* Decode dynamic range information; reference: table 4.52.
|
| 1281 |
*
|
| 1282 |
* @param cnt length of TYPE_FIL syntactic element in bytes
|
| 1283 |
*
|
| 1284 |
* @return Returns number of bytes consumed.
|
| 1285 |
*/
|
| 1286 |
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { |
| 1287 |
int n = 1; |
| 1288 |
int drc_num_bands = 1; |
| 1289 |
int i;
|
| 1290 |
|
| 1291 |
/* pce_tag_present? */
|
| 1292 |
if(get_bits1(gb)) {
|
| 1293 |
che_drc->pce_instance_tag = get_bits(gb, 4);
|
| 1294 |
skip_bits(gb, 4); // tag_reserved_bits |
| 1295 |
n++; |
| 1296 |
} |
| 1297 |
|
| 1298 |
/* excluded_chns_present? */
|
| 1299 |
if(get_bits1(gb)) {
|
| 1300 |
n += decode_drc_channel_exclusions(che_drc, gb); |
| 1301 |
} |
| 1302 |
|
| 1303 |
/* drc_bands_present? */
|
| 1304 |
if (get_bits1(gb)) {
|
| 1305 |
che_drc->band_incr = get_bits(gb, 4);
|
| 1306 |
che_drc->interpolation_scheme = get_bits(gb, 4);
|
| 1307 |
n++; |
| 1308 |
drc_num_bands += che_drc->band_incr; |
| 1309 |
for (i = 0; i < drc_num_bands; i++) { |
| 1310 |
che_drc->band_top[i] = get_bits(gb, 8);
|
| 1311 |
n++; |
| 1312 |
} |
| 1313 |
} |
| 1314 |
|
| 1315 |
/* prog_ref_level_present? */
|
| 1316 |
if (get_bits1(gb)) {
|
| 1317 |
che_drc->prog_ref_level = get_bits(gb, 7);
|
| 1318 |
skip_bits1(gb); // prog_ref_level_reserved_bits
|
| 1319 |
n++; |
| 1320 |
} |
| 1321 |
|
| 1322 |
for (i = 0; i < drc_num_bands; i++) { |
| 1323 |
che_drc->dyn_rng_sgn[i] = get_bits1(gb); |
| 1324 |
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
|
| 1325 |
n++; |
| 1326 |
} |
| 1327 |
|
| 1328 |
return n;
|
| 1329 |
} |
| 1330 |
|
| 1331 |
/**
|
| 1332 |
* Decode extension data (incomplete); reference: table 4.51.
|
| 1333 |
*
|
| 1334 |
* @param cnt length of TYPE_FIL syntactic element in bytes
|
| 1335 |
*
|
| 1336 |
* @return Returns number of bytes consumed
|
| 1337 |
*/
|
| 1338 |
static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { |
| 1339 |
int crc_flag = 0; |
| 1340 |
int res = cnt;
|
| 1341 |
switch (get_bits(gb, 4)) { // extension type |
| 1342 |
case EXT_SBR_DATA_CRC:
|
| 1343 |
crc_flag++; |
| 1344 |
case EXT_SBR_DATA:
|
| 1345 |
res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
| 1346 |
break;
|
| 1347 |
case EXT_DYNAMIC_RANGE:
|
| 1348 |
res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
| 1349 |
break;
|
| 1350 |
case EXT_FILL:
|
| 1351 |
case EXT_FILL_DATA:
|
| 1352 |
case EXT_DATA_ELEMENT:
|
| 1353 |
default:
|
| 1354 |
skip_bits_long(gb, 8*cnt - 4); |
| 1355 |
break;
|
| 1356 |
}; |
| 1357 |
return res;
|
| 1358 |
} |
| 1359 |
|
| 1360 |
/**
|
| 1361 |
* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
|
| 1362 |
*
|
| 1363 |
* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
|
| 1364 |
* @param coef spectral coefficients
|
| 1365 |
*/
|
| 1366 |
static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
| 1367 |
const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
| 1368 |
int w, filt, m, i;
|
| 1369 |
int bottom, top, order, start, end, size, inc;
|
| 1370 |
float lpc[TNS_MAX_ORDER];
|
| 1371 |
|
| 1372 |
for (w = 0; w < ics->num_windows; w++) { |
| 1373 |
bottom = ics->num_swb; |
| 1374 |
for (filt = 0; filt < tns->n_filt[w]; filt++) { |
| 1375 |
top = bottom; |
| 1376 |
bottom = FFMAX(0, top - tns->length[w][filt]);
|
| 1377 |
order = tns->order[w][filt]; |
| 1378 |
if (order == 0) |
| 1379 |
continue;
|
| 1380 |
|
| 1381 |
// tns_decode_coef
|
| 1382 |
compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
| 1383 |
|
| 1384 |
start = ics->swb_offset[FFMIN(bottom, mmm)]; |
| 1385 |
end = ics->swb_offset[FFMIN( top, mmm)]; |
| 1386 |
if ((size = end - start) <= 0) |
| 1387 |
continue;
|
| 1388 |
if (tns->direction[w][filt]) {
|
| 1389 |
inc = -1; start = end - 1; |
| 1390 |
} else {
|
| 1391 |
inc = 1;
|
| 1392 |
} |
| 1393 |
start += w * 128;
|
| 1394 |
|
| 1395 |
// ar filter
|
| 1396 |
for (m = 0; m < size; m++, start += inc) |
| 1397 |
for (i = 1; i <= FFMIN(m, order); i++) |
| 1398 |
coef[start] -= coef[start - i*inc] * lpc[i-1];
|
| 1399 |
} |
| 1400 |
} |
| 1401 |
} |
| 1402 |
|
| 1403 |
/**
|
| 1404 |
* Conduct IMDCT and windowing.
|
| 1405 |
*/
|
| 1406 |
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { |
| 1407 |
IndividualChannelStream * ics = &sce->ics; |
| 1408 |
float * in = sce->coeffs;
|
| 1409 |
float * out = sce->ret;
|
| 1410 |
float * saved = sce->saved;
|
| 1411 |
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
| 1412 |
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
| 1413 |
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
| 1414 |
float * buf = ac->buf_mdct;
|
| 1415 |
float * temp = ac->temp;
|
| 1416 |
int i;
|
| 1417 |
|
| 1418 |
// imdct
|
| 1419 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 1420 |
if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) |
| 1421 |
av_log(ac->avccontext, AV_LOG_WARNING, |
| 1422 |
"Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
|
| 1423 |
"If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
|
| 1424 |
for (i = 0; i < 1024; i += 128) |
| 1425 |
ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
| 1426 |
} else
|
| 1427 |
ff_imdct_half(&ac->mdct, buf, in); |
| 1428 |
|
| 1429 |
/* window overlapping
|
| 1430 |
* NOTE: To simplify the overlapping code, all 'meaningless' short to long
|
| 1431 |
* and long to short transitions are considered to be short to short
|
| 1432 |
* transitions. This leaves just two cases (long to long and short to short)
|
| 1433 |
* with a little special sauce for EIGHT_SHORT_SEQUENCE.
|
| 1434 |
*/
|
| 1435 |
if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
| 1436 |
(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
| 1437 |
ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512);
|
| 1438 |
} else {
|
| 1439 |
for (i = 0; i < 448; i++) |
| 1440 |
out[i] = saved[i] + ac->add_bias; |
| 1441 |
|
| 1442 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 1443 |
ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
| 1444 |
ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); |
| 1445 |
ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); |
| 1446 |
ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); |
| 1447 |
ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
| 1448 |
memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
| 1449 |
} else {
|
| 1450 |
ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
| 1451 |
for (i = 576; i < 1024; i++) |
| 1452 |
out[i] = buf[i-512] + ac->add_bias;
|
| 1453 |
} |
| 1454 |
} |
| 1455 |
|
| 1456 |
// buffer update
|
| 1457 |
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
| 1458 |
for (i = 0; i < 64; i++) |
| 1459 |
saved[i] = temp[64 + i] - ac->add_bias;
|
| 1460 |
ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
| 1461 |
ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
| 1462 |
ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
| 1463 |
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
| 1464 |
} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
| 1465 |
memcpy( saved, buf + 512, 448 * sizeof(float)); |
| 1466 |
memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
| 1467 |
} else { // LONG_STOP or ONLY_LONG |
| 1468 |
memcpy( saved, buf + 512, 512 * sizeof(float)); |
| 1469 |
} |
| 1470 |
} |
| 1471 |
|
| 1472 |
/**
|
| 1473 |
* Apply dependent channel coupling (applied before IMDCT).
|
| 1474 |
*
|
| 1475 |
* @param index index into coupling gain array
|
| 1476 |
*/
|
| 1477 |
static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
| 1478 |
IndividualChannelStream * ics = &cce->ch[0].ics;
|
| 1479 |
const uint16_t * offsets = ics->swb_offset;
|
| 1480 |
float * dest = target->coeffs;
|
| 1481 |
const float * src = cce->ch[0].coeffs; |
| 1482 |
int g, i, group, k, idx = 0; |
| 1483 |
if(ac->m4ac.object_type == AOT_AAC_LTP) {
|
| 1484 |
av_log(ac->avccontext, AV_LOG_ERROR, |
| 1485 |
"Dependent coupling is not supported together with LTP\n");
|
| 1486 |
return;
|
| 1487 |
} |
| 1488 |
for (g = 0; g < ics->num_window_groups; g++) { |
| 1489 |
for (i = 0; i < ics->max_sfb; i++, idx++) { |
| 1490 |
if (cce->ch[0].band_type[idx] != ZERO_BT) { |
| 1491 |
const float gain = cce->coup.gain[index][idx]; |
| 1492 |
for (group = 0; group < ics->group_len[g]; group++) { |
| 1493 |
for (k = offsets[i]; k < offsets[i+1]; k++) { |
| 1494 |
// XXX dsputil-ize
|
| 1495 |
dest[group*128+k] += gain * src[group*128+k]; |
| 1496 |
} |
| 1497 |
} |
| 1498 |
} |
| 1499 |
} |
| 1500 |
dest += ics->group_len[g]*128;
|
| 1501 |
src += ics->group_len[g]*128;
|
| 1502 |
} |
| 1503 |
} |
| 1504 |
|
| 1505 |
/**
|
| 1506 |
* Apply independent channel coupling (applied after IMDCT).
|
| 1507 |
*
|
| 1508 |
* @param index index into coupling gain array
|
| 1509 |
*/
|
| 1510 |
static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
| 1511 |
int i;
|
| 1512 |
const float gain = cce->coup.gain[index][0]; |
| 1513 |
const float bias = ac->add_bias; |
| 1514 |
const float* src = cce->ch[0].ret; |
| 1515 |
float* dest = target->ret;
|
| 1516 |
|
| 1517 |
for (i = 0; i < 1024; i++) |
| 1518 |
dest[i] += gain * (src[i] - bias); |
| 1519 |
} |
| 1520 |
|
| 1521 |
/**
|
| 1522 |
* channel coupling transformation interface
|
| 1523 |
*
|
| 1524 |
* @param index index into coupling gain array
|
| 1525 |
* @param apply_coupling_method pointer to (in)dependent coupling function
|
| 1526 |
*/
|
| 1527 |
static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, |
| 1528 |
enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
| 1529 |
void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
| 1530 |
{
|
| 1531 |
int i, c;
|
| 1532 |
|
| 1533 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
| 1534 |
ChannelElement *cce = ac->che[TYPE_CCE][i]; |
| 1535 |
int index = 0; |
| 1536 |
|
| 1537 |
if (cce && cce->coup.coupling_point == coupling_point) {
|
| 1538 |
ChannelCoupling * coup = &cce->coup; |
| 1539 |
|
| 1540 |
for (c = 0; c <= coup->num_coupled; c++) { |
| 1541 |
if (coup->type[c] == type && coup->id_select[c] == elem_id) {
|
| 1542 |
if (coup->ch_select[c] != 1) { |
| 1543 |
apply_coupling_method(ac, &cc->ch[0], cce, index);
|
| 1544 |
if (coup->ch_select[c] != 0) |
| 1545 |
index++; |
| 1546 |
} |
| 1547 |
if (coup->ch_select[c] != 2) |
| 1548 |
apply_coupling_method(ac, &cc->ch[1], cce, index++);
|
| 1549 |
} else
|
| 1550 |
index += 1 + (coup->ch_select[c] == 3); |
| 1551 |
} |
| 1552 |
} |
| 1553 |
} |
| 1554 |
} |
| 1555 |
|
| 1556 |
/**
|
| 1557 |
* Convert spectral data to float samples, applying all supported tools as appropriate.
|
| 1558 |
*/
|
| 1559 |
static void spectral_to_sample(AACContext * ac) { |
| 1560 |
int i, type;
|
| 1561 |
for(type = 3; type >= 0; type--) { |
| 1562 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
| 1563 |
ChannelElement *che = ac->che[type][i]; |
| 1564 |
if(che) {
|
| 1565 |
if(type <= TYPE_CPE)
|
| 1566 |
apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
| 1567 |
if(che->ch[0].tns.present) |
| 1568 |
apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); |
| 1569 |
if(che->ch[1].tns.present) |
| 1570 |
apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); |
| 1571 |
if(type <= TYPE_CPE)
|
| 1572 |
apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
| 1573 |
if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
|
| 1574 |
imdct_and_windowing(ac, &che->ch[0]);
|
| 1575 |
if(type == TYPE_CPE)
|
| 1576 |
imdct_and_windowing(ac, &che->ch[1]);
|
| 1577 |
if(type <= TYPE_CCE)
|
| 1578 |
apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
| 1579 |
} |
| 1580 |
} |
| 1581 |
} |
| 1582 |
} |
| 1583 |
|
| 1584 |
static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) { |
| 1585 |
|
| 1586 |
int size;
|
| 1587 |
AACADTSHeaderInfo hdr_info; |
| 1588 |
|
| 1589 |
size = ff_aac_parse_header(gb, &hdr_info); |
| 1590 |
if (size > 0) { |
| 1591 |
if (hdr_info.chan_config)
|
| 1592 |
ac->m4ac.chan_config = hdr_info.chan_config; |
| 1593 |
ac->m4ac.sample_rate = hdr_info.sample_rate; |
| 1594 |
ac->m4ac.sampling_index = hdr_info.sampling_index; |
| 1595 |
ac->m4ac.object_type = hdr_info.object_type; |
| 1596 |
if (hdr_info.num_aac_frames == 1) { |
| 1597 |
if (!hdr_info.crc_absent)
|
| 1598 |
skip_bits(gb, 16);
|
| 1599 |
} else {
|
| 1600 |
ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); |
| 1601 |
return -1; |
| 1602 |
} |
| 1603 |
} |
| 1604 |
return size;
|
| 1605 |
} |
| 1606 |
|
| 1607 |
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, AVPacket *avpkt) { |
| 1608 |
const uint8_t *buf = avpkt->data;
|
| 1609 |
int buf_size = avpkt->size;
|
| 1610 |
AACContext * ac = avccontext->priv_data; |
| 1611 |
ChannelElement * che = NULL;
|
| 1612 |
GetBitContext gb; |
| 1613 |
enum RawDataBlockType elem_type;
|
| 1614 |
int err, elem_id, data_size_tmp;
|
| 1615 |
|
| 1616 |
init_get_bits(&gb, buf, buf_size*8);
|
| 1617 |
|
| 1618 |
if (show_bits(&gb, 12) == 0xfff) { |
| 1619 |
if ((err = parse_adts_frame_header(ac, &gb)) < 0) { |
| 1620 |
av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
|
| 1621 |
return -1; |
| 1622 |
} |
| 1623 |
if (ac->m4ac.sampling_index > 12) { |
| 1624 |
av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
|
| 1625 |
return -1; |
| 1626 |
} |
| 1627 |
} |
| 1628 |
|
| 1629 |
// parse
|
| 1630 |
while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { |
| 1631 |
elem_id = get_bits(&gb, 4);
|
| 1632 |
err = -1;
|
| 1633 |
|
| 1634 |
if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
|
| 1635 |
av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
|
| 1636 |
return -1; |
| 1637 |
} |
| 1638 |
|
| 1639 |
switch (elem_type) {
|
| 1640 |
|
| 1641 |
case TYPE_SCE:
|
| 1642 |
err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
| 1643 |
break;
|
| 1644 |
|
| 1645 |
case TYPE_CPE:
|
| 1646 |
err = decode_cpe(ac, &gb, che); |
| 1647 |
break;
|
| 1648 |
|
| 1649 |
case TYPE_CCE:
|
| 1650 |
err = decode_cce(ac, &gb, che); |
| 1651 |
break;
|
| 1652 |
|
| 1653 |
case TYPE_LFE:
|
| 1654 |
err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
| 1655 |
break;
|
| 1656 |
|
| 1657 |
case TYPE_DSE:
|
| 1658 |
skip_data_stream_element(&gb); |
| 1659 |
err = 0;
|
| 1660 |
break;
|
| 1661 |
|
| 1662 |
case TYPE_PCE:
|
| 1663 |
{
|
| 1664 |
enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
| 1665 |
memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
| 1666 |
if((err = decode_pce(ac, new_che_pos, &gb)))
|
| 1667 |
break;
|
| 1668 |
err = output_configure(ac, ac->che_pos, new_che_pos, 0);
|
| 1669 |
break;
|
| 1670 |
} |
| 1671 |
|
| 1672 |
case TYPE_FIL:
|
| 1673 |
if (elem_id == 15) |
| 1674 |
elem_id += get_bits(&gb, 8) - 1; |
| 1675 |
while (elem_id > 0) |
| 1676 |
elem_id -= decode_extension_payload(ac, &gb, elem_id); |
| 1677 |
err = 0; /* FIXME */ |
| 1678 |
break;
|
| 1679 |
|
| 1680 |
default:
|
| 1681 |
err = -1; /* should not happen, but keeps compiler happy */ |
| 1682 |
break;
|
| 1683 |
} |
| 1684 |
|
| 1685 |
if(err)
|
| 1686 |
return err;
|
| 1687 |
} |
| 1688 |
|
| 1689 |
spectral_to_sample(ac); |
| 1690 |
|
| 1691 |
if (!ac->is_saved) {
|
| 1692 |
ac->is_saved = 1;
|
| 1693 |
*data_size = 0;
|
| 1694 |
return buf_size;
|
| 1695 |
} |
| 1696 |
|
| 1697 |
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); |
| 1698 |
if(*data_size < data_size_tmp) {
|
| 1699 |
av_log(avccontext, AV_LOG_ERROR, |
| 1700 |
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
|
| 1701 |
*data_size, data_size_tmp); |
| 1702 |
return -1; |
| 1703 |
} |
| 1704 |
*data_size = data_size_tmp; |
| 1705 |
|
| 1706 |
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); |
| 1707 |
|
| 1708 |
return buf_size;
|
| 1709 |
} |
| 1710 |
|
| 1711 |
static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
| 1712 |
AACContext * ac = avccontext->priv_data; |
| 1713 |
int i, type;
|
| 1714 |
|
| 1715 |
for (i = 0; i < MAX_ELEM_ID; i++) { |
| 1716 |
for(type = 0; type < 4; type++) |
| 1717 |
av_freep(&ac->che[type][i]); |
| 1718 |
} |
| 1719 |
|
| 1720 |
ff_mdct_end(&ac->mdct); |
| 1721 |
ff_mdct_end(&ac->mdct_small); |
| 1722 |
return 0 ; |
| 1723 |
} |
| 1724 |
|
| 1725 |
AVCodec aac_decoder = {
|
| 1726 |
"aac",
|
| 1727 |
CODEC_TYPE_AUDIO, |
| 1728 |
CODEC_ID_AAC, |
| 1729 |
sizeof(AACContext),
|
| 1730 |
aac_decode_init, |
| 1731 |
NULL,
|
| 1732 |
aac_decode_close, |
| 1733 |
aac_decode_frame, |
| 1734 |
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
|
| 1735 |
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
|
| 1736 |
}; |