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https://github.com/FFmpeg/FFmpeg.git
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16bb8247b4
These arrays are used by the Musepack decoders, the MPEG audio decoders as well as qdm2 and up until now, these arrays might be initialized more than once, leading to potential data races as well as unnecessary initializations. Therefore this commit ensures that each array will only be initialized once. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
95 lines
3.1 KiB
C
95 lines
3.1 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 "libavutil/attributes.h"
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#include "avcodec.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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/**
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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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for(ch = 0; ch < channels; 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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out[ch] + 32 * i, 1,
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c->sb_samples[ch][i]);
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}
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}
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}
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void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
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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+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0] & 0xFF];
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for(; j < 12; j++)
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c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
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mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1] & 0xFF];
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for(; j < 24; j++)
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c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
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mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2] & 0xFF];
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for(; j < 36; j++)
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c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
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}
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}
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if(bands[i].msf){
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unsigned 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, out, channels);
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}
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