Mercurial > libavcodec.hg
changeset 7501:d6012be599d3 libavcodec
OKed sections of code from the SoC AAC decoder
| author | superdump |
|---|---|
| date | Tue, 05 Aug 2008 19:32:01 +0000 |
| parents | 0499a257d17f |
| children | d5c528384f13 |
| files | aac.c |
| diffstat | 1 files changed, 241 insertions(+), 0 deletions(-) [+] |
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--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/aac.c Tue Aug 05 19:32:01 2008 +0000 @@ -0,0 +1,241 @@ +/* + * 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"), +};