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OKed sections of code from the SoC AAC decoder
Originally committed as revision 14626 to svn://svn.ffmpeg.org/ffmpeg/trunk
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Robert Swain
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241
libavcodec/aac.c
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241
libavcodec/aac.c
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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 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 "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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#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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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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newpcs->mono_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1;
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newpcs->stereo_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1;
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if (get_bits1(gb)) {
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newpcs->mixdown_coeff_index = get_bits(gb, 2);
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newpcs->pseudo_surround = get_bits1(gb);
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}
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program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_FRONT, gb, num_front);
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program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_SIDE, gb, num_side );
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program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_BACK, gb, num_back );
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program_config_element_parse_tags(NULL, newpcs->che_type[ID_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
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skip_bits_long(gb, 4 * num_assoc_data);
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program_config_element_parse_tags(newpcs->che_type[ID_CCE], newpcs->che_type[ID_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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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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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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// -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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#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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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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* 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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* @param pulse pointer to pulse data struct
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* @param icoef array of quantized spectral data
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*/
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static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
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int i, off = ics->swb_offset[pulse->start];
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for (i = 0; i < pulse->num_pulse; i++) {
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int ic;
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off += pulse->offset[i];
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ic = (icoef[off] - 1)>>31;
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icoef[off] += (pulse->amp[i]^ic) - ic;
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}
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}
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static av_cold int aac_decode_close(AVCodecContext * avccontext) {
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AACContext * ac = avccontext->priv_data;
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int i, j;
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for (i = 0; i < MAX_TAGID; i++) {
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for(j = 0; j < 4; j++)
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av_freep(&ac->che[j][i]);
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}
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ff_mdct_end(&ac->mdct);
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ff_mdct_end(&ac->mdct_small);
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av_freep(&ac->interleaved_output);
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return 0 ;
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}
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AVCodec aac_decoder = {
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"aac",
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CODEC_TYPE_AUDIO,
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CODEC_ID_AAC,
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sizeof(AACContext),
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aac_decode_init,
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NULL,
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aac_decode_close,
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aac_decode_frame,
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.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
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};
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