Mercurial > libavcodec.hg
view aac.c @ 7504:c75c3adf11bc libavcodec
Cosmetics: alignment
author | vitor |
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date | Wed, 06 Aug 2008 02:49:22 +0000 |
parents | d6012be599d3 |
children | a3f7ffdb676d |
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/* * AAC decoder * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /** * @file aac.c * AAC decoder * @author Oded Shimon ( ods15 ods15 dyndns org ) * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) */ /* * supported tools * * Support? Name * N (code in SoC repo) gain control * Y block switching * Y window shapes - standard * N window shapes - Low Delay * Y filterbank - standard * N (code in SoC repo) filterbank - Scalable Sample Rate * Y Temporal Noise Shaping * N (code in SoC repo) Long Term Prediction * Y intensity stereo * Y channel coupling * N frequency domain prediction * Y Perceptual Noise Substitution * Y Mid/Side stereo * N Scalable Inverse AAC Quantization * N Frequency Selective Switch * N upsampling filter * Y quantization & coding - AAC * N quantization & coding - TwinVQ * N quantization & coding - BSAC * N AAC Error Resilience tools * N Error Resilience payload syntax * N Error Protection tool * N CELP * N Silence Compression * N HVXC * N HVXC 4kbits/s VR * N Structured Audio tools * N Structured Audio Sample Bank Format * N MIDI * N Harmonic and Individual Lines plus Noise * N Text-To-Speech Interface * N (in progress) Spectral Band Replication * Y (not in this code) Layer-1 * Y (not in this code) Layer-2 * Y (not in this code) Layer-3 * N SinuSoidal Coding (Transient, Sinusoid, Noise) * N (planned) Parametric Stereo * N Direct Stream Transfer * * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. * - HE AAC v2 comprises LC AAC with Spectral Band Replication and Parametric Stereo. */ #include "avcodec.h" #include "bitstream.h" #include "dsputil.h" #include "aac.h" #include "aactab.h" #include "mpeg4audio.h" #include <assert.h> #include <errno.h> #include <math.h> #include <string.h> #ifndef CONFIG_HARDCODED_TABLES static float ff_aac_ivquant_tab[IVQUANT_SIZE]; #endif /* CONFIG_HARDCODED_TABLES */ static VLC vlc_scalefactors; static VLC vlc_spectral[11]; num_front = get_bits(gb, 4); num_side = get_bits(gb, 4); num_back = get_bits(gb, 4); num_lfe = get_bits(gb, 2); num_assoc_data = get_bits(gb, 3); num_cc = get_bits(gb, 4); newpcs->mono_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1; newpcs->stereo_mixdown_tag = get_bits1(gb) ? get_bits(gb, 4) : -1; if (get_bits1(gb)) { newpcs->mixdown_coeff_index = get_bits(gb, 2); newpcs->pseudo_surround = get_bits1(gb); } program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_FRONT, gb, num_front); program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_SIDE, gb, num_side ); program_config_element_parse_tags(newpcs->che_type[ID_CPE], newpcs->che_type[ID_SCE], AAC_CHANNEL_BACK, gb, num_back ); program_config_element_parse_tags(NULL, newpcs->che_type[ID_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); skip_bits_long(gb, 4 * num_assoc_data); program_config_element_parse_tags(newpcs->che_type[ID_CCE], newpcs->che_type[ID_CCE], AAC_CHANNEL_CC, gb, num_cc ); align_get_bits(gb); /* comment field, first byte is length */ skip_bits_long(gb, 8 * get_bits(gb, 8)); static av_cold int aac_decode_init(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i; ac->avccontext = avccontext; avccontext->sample_rate = ac->m4ac.sample_rate; avccontext->frame_size = 1024; AAC_INIT_VLC_STATIC( 0, 144); AAC_INIT_VLC_STATIC( 1, 114); AAC_INIT_VLC_STATIC( 2, 188); AAC_INIT_VLC_STATIC( 3, 180); AAC_INIT_VLC_STATIC( 4, 172); AAC_INIT_VLC_STATIC( 5, 140); AAC_INIT_VLC_STATIC( 6, 168); AAC_INIT_VLC_STATIC( 7, 114); AAC_INIT_VLC_STATIC( 8, 262); AAC_INIT_VLC_STATIC( 9, 248); AAC_INIT_VLC_STATIC(10, 384); dsputil_init(&ac->dsp, avccontext); // -1024 - Compensate wrong IMDCT method. // 32768 - Required to scale values to the correct range for the bias method // for float to int16 conversion. if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { ac->add_bias = 385.0f; ac->sf_scale = 1. / (-1024. * 32768.); ac->sf_offset = 0; } else { ac->add_bias = 0.0f; ac->sf_scale = 1. / -1024.; ac->sf_offset = 60; } #ifndef CONFIG_HARDCODED_TABLES for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++) ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i; #endif /* CONFIG_HARDCODED_TABLES */ INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), 352); ff_mdct_init(&ac->mdct, 11, 1); ff_mdct_init(&ac->mdct_small, 8, 1); return 0; } int byte_align = get_bits1(gb); int count = get_bits(gb, 8); if (count == 255) count += get_bits(gb, 8); if (byte_align) align_get_bits(gb); skip_bits_long(gb, 8 * count); } /** * inverse quantization * * @param a quantized value to be dequantized * @return Returns dequantized value. */ static inline float ivquant(int a) { if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1) return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1]; else return cbrtf(fabsf(a)) * a; } * @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; } } static av_cold int aac_decode_close(AVCodecContext * avccontext) { AACContext * ac = avccontext->priv_data; int i, j; for (i = 0; i < MAX_TAGID; i++) { for(j = 0; j < 4; j++) av_freep(&ac->che[j][i]); } ff_mdct_end(&ac->mdct); ff_mdct_end(&ac->mdct_small); av_freep(&ac->interleaved_output); 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"), };