mirror of https://git.ffmpeg.org/ffmpeg.git
844 lines
29 KiB
C
844 lines
29 KiB
C
/*
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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 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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* N 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 "bitstream.h"
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#include "dsputil.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 <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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#ifndef CONFIG_HARDCODED_TABLES
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static float ff_aac_ivquant_tab[IVQUANT_SIZE];
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static float ff_aac_pow2sf_tab[316];
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#endif /* CONFIG_HARDCODED_TABLES */
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static VLC vlc_scalefactors;
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static VLC vlc_spectral[11];
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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;
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skip_bits(gb, 2); // object_type
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ac->m4ac.sampling_index = get_bits(gb, 4);
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if(ac->m4ac.sampling_index > 11) {
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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.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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return -1;
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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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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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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)))
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return ret;
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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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* 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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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 > 11) {
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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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skip_bits_long(&gb, i);
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switch (ac->m4ac.object_type) {
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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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static av_cold int aac_decode_init(AVCodecContext * avccontext) {
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AACContext * ac = avccontext->priv_data;
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int i;
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ac->avccontext = avccontext;
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if (avccontext->extradata_size <= 0 ||
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decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
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return -1;
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avccontext->sample_fmt = SAMPLE_FMT_S16;
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avccontext->sample_rate = ac->m4ac.sample_rate;
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avccontext->frame_size = 1024;
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AAC_INIT_VLC_STATIC( 0, 144);
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AAC_INIT_VLC_STATIC( 1, 114);
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AAC_INIT_VLC_STATIC( 2, 188);
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AAC_INIT_VLC_STATIC( 3, 180);
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AAC_INIT_VLC_STATIC( 4, 172);
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AAC_INIT_VLC_STATIC( 5, 140);
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AAC_INIT_VLC_STATIC( 6, 168);
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AAC_INIT_VLC_STATIC( 7, 114);
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AAC_INIT_VLC_STATIC( 8, 262);
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AAC_INIT_VLC_STATIC( 9, 248);
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AAC_INIT_VLC_STATIC(10, 384);
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dsputil_init(&ac->dsp, avccontext);
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ac->random_state = 0x1f2e3d4c;
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// -1024 - Compensate wrong IMDCT method.
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// 32768 - Required to scale values to the correct range for the bias method
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// for float to int16 conversion.
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if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
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ac->add_bias = 385.0f;
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ac->sf_scale = 1. / (-1024. * 32768.);
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ac->sf_offset = 0;
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} else {
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ac->add_bias = 0.0f;
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ac->sf_scale = 1. / -1024.;
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ac->sf_offset = 60;
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}
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#ifndef CONFIG_HARDCODED_TABLES
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for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
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ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i;
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for (i = 0; i < 316; i++)
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ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
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#endif /* CONFIG_HARDCODED_TABLES */
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INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
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ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
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ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
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352);
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ff_mdct_init(&ac->mdct, 11, 1);
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ff_mdct_init(&ac->mdct_small, 8, 1);
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return 0;
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}
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/**
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* Skip data_stream_element; reference: table 4.10.
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*/
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static void skip_data_stream_element(GetBitContext * gb) {
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int byte_align = get_bits1(gb);
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int count = get_bits(gb, 8);
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if (count == 255)
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count += get_bits(gb, 8);
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if (byte_align)
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align_get_bits(gb);
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skip_bits_long(gb, 8 * count);
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}
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/**
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* Decode Individual Channel Stream info; reference: table 4.6.
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*
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* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
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*/
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static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
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if (get_bits1(gb)) {
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av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
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memset(ics, 0, sizeof(IndividualChannelStream));
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return -1;
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}
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ics->window_sequence[1] = ics->window_sequence[0];
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ics->window_sequence[0] = get_bits(gb, 2);
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ics->use_kb_window[1] = ics->use_kb_window[0];
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ics->use_kb_window[0] = get_bits1(gb);
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ics->num_window_groups = 1;
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ics->group_len[0] = 1;
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return 0;
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}
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/**
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* inverse quantization
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*
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* @param a quantized value to be dequantized
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* @return Returns dequantized value.
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*/
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static inline float ivquant(int a) {
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if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1)
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return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1];
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else
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return cbrtf(fabsf(a)) * a;
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}
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/**
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* Decode band types (section_data payload); reference: table 4.46.
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*
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* @param band_type array of the used band type
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* @param band_type_run_end array of the last scalefactor band of a band type run
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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_band_types(AACContext * ac, enum BandType band_type[120],
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int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
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int g, idx = 0;
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const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
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for (g = 0; g < ics->num_window_groups; g++) {
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int k = 0;
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while (k < ics->max_sfb) {
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uint8_t sect_len = k;
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int sect_len_incr;
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int sect_band_type = get_bits(gb, 4);
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if (sect_band_type == 12) {
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av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
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return -1;
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}
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while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
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sect_len += sect_len_incr;
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sect_len += sect_len_incr;
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if (sect_len > ics->max_sfb) {
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av_log(ac->avccontext, AV_LOG_ERROR,
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"Number of bands (%d) exceeds limit (%d).\n",
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sect_len, ics->max_sfb);
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return -1;
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}
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}
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}
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return 0;
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}
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/**
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* Decode scalefactors; reference: table 4.47.
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*
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* @param mix_gain channel gain (Not used by AAC bitstream.)
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* @param global_gain first scalefactor value as scalefactors are differentially coded
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* @param band_type array of the used band type
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* @param band_type_run_end array of the last scalefactor band of a band type run
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* @param sf array of scalefactors or intensity stereo positions
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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_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
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float mix_gain, unsigned int global_gain, IndividualChannelStream * ics,
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enum BandType band_type[120], int band_type_run_end[120]) {
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const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
|
|
int g, i, idx = 0;
|
|
int offset[3] = { global_gain, global_gain - 90, 100 };
|
|
int noise_flag = 1;
|
|
static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
|
|
ics->intensity_present = 0;
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb;) {
|
|
int run_end = band_type_run_end[idx];
|
|
if (band_type[idx] == ZERO_BT) {
|
|
for(; i < run_end; i++, idx++)
|
|
sf[idx] = 0.;
|
|
}else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
|
|
ics->intensity_present = 1;
|
|
for(; i < run_end; i++, idx++) {
|
|
offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
if(offset[2] > 255U) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR,
|
|
"%s (%d) out of range.\n", sf_str[2], offset[2]);
|
|
return -1;
|
|
}
|
|
sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
|
|
sf[idx] *= mix_gain;
|
|
}
|
|
}else if(band_type[idx] == NOISE_BT) {
|
|
for(; i < run_end; i++, idx++) {
|
|
if(noise_flag-- > 0)
|
|
offset[1] += get_bits(gb, 9) - 256;
|
|
else
|
|
offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
if(offset[1] > 255U) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR,
|
|
"%s (%d) out of range.\n", sf_str[1], offset[1]);
|
|
return -1;
|
|
}
|
|
sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
|
|
sf[idx] *= mix_gain;
|
|
}
|
|
}else {
|
|
for(; i < run_end; i++, idx++) {
|
|
offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
|
|
if(offset[0] > 255U) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR,
|
|
"%s (%d) out of range.\n", sf_str[0], offset[0]);
|
|
return -1;
|
|
}
|
|
sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
|
|
sf[idx] *= mix_gain;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode pulse data; reference: table 4.7.
|
|
*/
|
|
static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
|
|
int i;
|
|
pulse->num_pulse = get_bits(gb, 2) + 1;
|
|
pulse->start = get_bits(gb, 6);
|
|
for (i = 0; i < pulse->num_pulse; i++) {
|
|
pulse->offset[i] = get_bits(gb, 5);
|
|
pulse->amp [i] = get_bits(gb, 4);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Decode Mid/Side data; reference: table 4.54.
|
|
*
|
|
* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
|
|
* [1] mask is decoded from bitstream; [2] mask is all 1s;
|
|
* [3] reserved for scalable AAC
|
|
*/
|
|
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
|
|
int ms_present) {
|
|
|
|
/**
|
|
* Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
|
|
*
|
|
* @param pulse pointer to pulse data struct
|
|
* @param icoef array of quantized spectral data
|
|
*/
|
|
static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
|
|
int i, off = ics->swb_offset[pulse->start];
|
|
for (i = 0; i < pulse->num_pulse; i++) {
|
|
int ic;
|
|
off += pulse->offset[i];
|
|
ic = (icoef[off] - 1)>>31;
|
|
icoef[off] += (pulse->amp[i]^ic) - ic;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Decode an individual_channel_stream payload; reference: table 4.44.
|
|
*
|
|
* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
|
|
* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
|
|
int icoeffs[1024];
|
|
Pulse pulse;
|
|
TemporalNoiseShaping * tns = &sce->tns;
|
|
IndividualChannelStream * ics = &sce->ics;
|
|
float * out = sce->coeffs;
|
|
int global_gain, pulse_present = 0;
|
|
|
|
/* These two assignments are to silence some GCC warnings about the
|
|
* variables being used uninitialised when in fact they always are.
|
|
*/
|
|
pulse.num_pulse = 0;
|
|
pulse.start = 0;
|
|
|
|
global_gain = get_bits(gb, 8);
|
|
|
|
if (!common_window && !scale_flag) {
|
|
if (decode_ics_info(ac, ics, gb, 0) < 0)
|
|
return -1;
|
|
}
|
|
|
|
if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
|
|
return -1;
|
|
if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
|
|
return -1;
|
|
|
|
pulse_present = 0;
|
|
if (!scale_flag) {
|
|
if ((pulse_present = get_bits1(gb))) {
|
|
if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
|
|
return -1;
|
|
}
|
|
decode_pulses(&pulse, gb);
|
|
}
|
|
if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
|
|
return -1;
|
|
if (get_bits1(gb)) {
|
|
av_log_missing_feature(ac->avccontext, "SSR", 1);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
|
|
return -1;
|
|
if (pulse_present)
|
|
add_pulses(icoeffs, &pulse, ics);
|
|
dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode a channel_pair_element; reference: table 4.4.
|
|
*
|
|
* @param elem_id Identifies the instance of a syntax element.
|
|
*
|
|
* @return Returns error status. 0 - OK, !0 - error
|
|
*/
|
|
static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
|
|
int i, ret, common_window, ms_present = 0;
|
|
ChannelElement * cpe;
|
|
|
|
cpe = ac->che[TYPE_CPE][elem_id];
|
|
common_window = get_bits1(gb);
|
|
if (common_window) {
|
|
if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
|
|
return -1;
|
|
i = cpe->ch[1].ics.use_kb_window[0];
|
|
cpe->ch[1].ics = cpe->ch[0].ics;
|
|
cpe->ch[1].ics.use_kb_window[1] = i;
|
|
ms_present = get_bits(gb, 2);
|
|
if(ms_present == 3) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
|
|
return -1;
|
|
} else if(ms_present)
|
|
decode_mid_side_stereo(cpe, gb, ms_present);
|
|
}
|
|
if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
|
|
return ret;
|
|
if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
|
|
return ret;
|
|
|
|
if (common_window && ms_present)
|
|
apply_mid_side_stereo(cpe);
|
|
|
|
if (cpe->ch[1].ics.intensity_present)
|
|
apply_intensity_stereo(cpe, ms_present);
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Decode Spectral Band Replication extension data; reference: table 4.55.
|
|
*
|
|
* @param crc flag indicating the presence of CRC checksum
|
|
* @param cnt length of TYPE_FIL syntactic element in bytes
|
|
*
|
|
* @return Returns number of bytes consumed from the TYPE_FIL element.
|
|
*/
|
|
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
|
|
// TODO : sbr_extension implementation
|
|
av_log(ac->avccontext, AV_LOG_DEBUG, "aac: SBR not yet supported.\n");
|
|
skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
|
|
return cnt;
|
|
}
|
|
|
|
/**
|
|
* Decode dynamic range information; reference: table 4.52.
|
|
*
|
|
* @param cnt length of TYPE_FIL syntactic element in bytes
|
|
*
|
|
* @return Returns number of bytes consumed.
|
|
*/
|
|
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
|
|
int n = 1;
|
|
int drc_num_bands = 1;
|
|
int i;
|
|
|
|
/* pce_tag_present? */
|
|
if(get_bits1(gb)) {
|
|
che_drc->pce_instance_tag = get_bits(gb, 4);
|
|
skip_bits(gb, 4); // tag_reserved_bits
|
|
n++;
|
|
}
|
|
|
|
/* excluded_chns_present? */
|
|
if(get_bits1(gb)) {
|
|
n += decode_drc_channel_exclusions(che_drc, gb);
|
|
}
|
|
|
|
/* drc_bands_present? */
|
|
if (get_bits1(gb)) {
|
|
che_drc->band_incr = get_bits(gb, 4);
|
|
che_drc->interpolation_scheme = get_bits(gb, 4);
|
|
n++;
|
|
drc_num_bands += che_drc->band_incr;
|
|
for (i = 0; i < drc_num_bands; i++) {
|
|
che_drc->band_top[i] = get_bits(gb, 8);
|
|
n++;
|
|
}
|
|
}
|
|
|
|
/* prog_ref_level_present? */
|
|
if (get_bits1(gb)) {
|
|
che_drc->prog_ref_level = get_bits(gb, 7);
|
|
skip_bits1(gb); // prog_ref_level_reserved_bits
|
|
n++;
|
|
}
|
|
|
|
for (i = 0; i < drc_num_bands; i++) {
|
|
che_drc->dyn_rng_sgn[i] = get_bits1(gb);
|
|
che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
|
|
n++;
|
|
}
|
|
|
|
return n;
|
|
}
|
|
|
|
/**
|
|
* Decode extension data (incomplete); reference: table 4.51.
|
|
*
|
|
* @param cnt length of TYPE_FIL syntactic element in bytes
|
|
*
|
|
* @return Returns number of bytes consumed
|
|
*/
|
|
static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
|
|
int crc_flag = 0;
|
|
int res = cnt;
|
|
switch (get_bits(gb, 4)) { // extension type
|
|
case EXT_SBR_DATA_CRC:
|
|
crc_flag++;
|
|
case EXT_SBR_DATA:
|
|
res = decode_sbr_extension(ac, gb, crc_flag, cnt);
|
|
break;
|
|
case EXT_DYNAMIC_RANGE:
|
|
res = decode_dynamic_range(&ac->che_drc, gb, cnt);
|
|
break;
|
|
case EXT_FILL:
|
|
case EXT_FILL_DATA:
|
|
case EXT_DATA_ELEMENT:
|
|
default:
|
|
skip_bits_long(gb, 8*cnt - 4);
|
|
break;
|
|
};
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* Conduct IMDCT and windowing.
|
|
*/
|
|
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
|
|
IndividualChannelStream * ics = &sce->ics;
|
|
float * in = sce->coeffs;
|
|
float * out = sce->ret;
|
|
float * saved = sce->saved;
|
|
const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
|
|
const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
|
|
const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
|
|
const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
|
|
float * buf = ac->buf_mdct;
|
|
int i;
|
|
|
|
/**
|
|
* Apply dependent channel coupling (applied before IMDCT).
|
|
*
|
|
* @param index index into coupling gain array
|
|
*/
|
|
static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
|
|
IndividualChannelStream * ics = &cc->ch[0].ics;
|
|
const uint16_t * offsets = ics->swb_offset;
|
|
float * dest = sce->coeffs;
|
|
const float * src = cc->ch[0].coeffs;
|
|
int g, i, group, k, idx = 0;
|
|
if(ac->m4ac.object_type == AOT_AAC_LTP) {
|
|
av_log(ac->avccontext, AV_LOG_ERROR,
|
|
"Dependent coupling is not supported together with LTP\n");
|
|
return;
|
|
}
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
|
for (i = 0; i < ics->max_sfb; i++, idx++) {
|
|
if (cc->ch[0].band_type[idx] != ZERO_BT) {
|
|
float gain = cc->coup.gain[index][idx] * sce->mixing_gain;
|
|
for (group = 0; group < ics->group_len[g]; group++) {
|
|
for (k = offsets[i]; k < offsets[i+1]; k++) {
|
|
// XXX dsputil-ize
|
|
dest[group*128+k] += gain * src[group*128+k];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
dest += ics->group_len[g]*128;
|
|
src += ics->group_len[g]*128;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Apply independent channel coupling (applied after IMDCT).
|
|
*
|
|
* @param index index into coupling gain array
|
|
*/
|
|
static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
|
|
int i;
|
|
float gain = cc->coup.gain[index][0] * sce->mixing_gain;
|
|
for (i = 0; i < 1024; i++)
|
|
sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
|
|
}
|
|
|
|
if (!ac->is_saved) {
|
|
ac->is_saved = 1;
|
|
*data_size = 0;
|
|
return 0;
|
|
}
|
|
|
|
data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
|
|
if(*data_size < data_size_tmp) {
|
|
av_log(avccontext, AV_LOG_ERROR,
|
|
"Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
|
|
*data_size, data_size_tmp);
|
|
return -1;
|
|
}
|
|
*data_size = data_size_tmp;
|
|
|
|
ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
|
|
|
|
return buf_size;
|
|
}
|
|
|
|
static av_cold int aac_decode_close(AVCodecContext * avccontext) {
|
|
AACContext * ac = avccontext->priv_data;
|
|
int i, j;
|
|
|
|
for (i = 0; i < MAX_ELEM_ID; i++) {
|
|
for(j = 0; j < 4; j++)
|
|
av_freep(&ac->che[j][i]);
|
|
}
|
|
|
|
ff_mdct_end(&ac->mdct);
|
|
ff_mdct_end(&ac->mdct_small);
|
|
return 0 ;
|
|
}
|
|
|
|
AVCodec aac_decoder = {
|
|
"aac",
|
|
CODEC_TYPE_AUDIO,
|
|
CODEC_ID_AAC,
|
|
sizeof(AACContext),
|
|
aac_decode_init,
|
|
NULL,
|
|
aac_decode_close,
|
|
aac_decode_frame,
|
|
.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
|
|
.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
|
|
};
|