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80d156d7fd
* qatar/master: qdm2: Use floating point synthesis filter. h264: correct border check. h264: fix loopfilter with threading at slice boundaries. Fix ff_mpa_synth_filter_fixed() prototype Rename costablegen.c ---> cos_tablegen.c. Collapse tableprint.c into tableprint.h. Simplify trig table rules Remove potentially unstable filenames from comments in generated files. Ignore generated tables and generated table generator programs. Simplify CLEANFILES make variable by using wildcards. Remove silly insults from avformat_version() Doxygen documentation. mpegaudiodsp: fix x86 and ppc makefiles configure: Adjust AVX assembler check. mpegaudio: remove unused version of SAME_HEADER_MASK mpegaudio: remove useless #undef at end of file asfdec: add missing #include for av_bswap32() mpegaudio: merge two #if CONFIG_FLOAT blocks mpegaudio: move some struct definitions from mpegaudio.h Move some mpegaudio functions to new mpegaudiodsp subsystem Conflicts: libavcodec/h264.c libavcodec/x86/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>
105 lines
3.3 KiB
C
105 lines
3.3 KiB
C
/*
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* Musepack decoder core
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* Copyright (c) 2006 Konstantin Shishkov
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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
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* Musepack decoder core
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* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
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* divided into 32 subbands.
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*/
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#include "avcodec.h"
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#include "get_bits.h"
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#include "dsputil.h"
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#include "mpegaudiodsp.h"
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#include "mpegaudio.h"
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#include "mpc.h"
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#include "mpcdata.h"
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void ff_mpc_init(void)
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{
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ff_mpa_synth_init_fixed(ff_mpa_synth_window_fixed);
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}
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/**
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* Process decoded Musepack data and produce PCM
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*/
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static void mpc_synth(MPCContext *c, int16_t *out, int channels)
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{
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int dither_state = 0;
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int i, ch;
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OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
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for(ch = 0; ch < channels; ch++){
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samples_ptr = samples + ch;
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for(i = 0; i < SAMPLES_PER_BAND; i++) {
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ff_mpa_synth_filter_fixed(&c->mpadsp,
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c->synth_buf[ch], &(c->synth_buf_offset[ch]),
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ff_mpa_synth_window_fixed, &dither_state,
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samples_ptr, channels,
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c->sb_samples[ch][i]);
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samples_ptr += 32 * channels;
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}
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}
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for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
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*out++=samples[i];
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}
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void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
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{
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int i, j, ch;
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Band *bands = c->bands;
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int off;
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float mul;
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/* dequantize */
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memset(c->sb_samples, 0, sizeof(c->sb_samples));
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off = 0;
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for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
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for(ch = 0; ch < 2; ch++){
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if(bands[i].res[ch]){
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j = 0;
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mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0]];
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for(; j < 12; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1]];
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for(; j < 24; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2]];
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for(; j < 36; j++)
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c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
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}
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}
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if(bands[i].msf){
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int t1, t2;
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for(j = 0; j < SAMPLES_PER_BAND; j++){
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t1 = c->sb_samples[0][j][i];
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t2 = c->sb_samples[1][j][i];
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c->sb_samples[0][j][i] = t1 + t2;
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c->sb_samples[1][j][i] = t1 - t2;
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}
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}
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}
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mpc_synth(c, data, channels);
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}
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