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241 lines
7.9 KiB
241 lines
7.9 KiB
/* |
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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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