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e3b355c0be
It is not necessary at all. So remove it. This also breaks an inclusion cycle mem.h->avutil.h->common.h->mem.h. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
94 lines
3.1 KiB
C
94 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 <string.h>
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#include "libavutil/common.h"
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#include "mpegaudiodsp.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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