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/*
* Musepack decoder core
* Copyright (c) 2006 Konstantin Shishkov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Musepack decoder core
* MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
* divided into 32 subbands.
*/
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "mpegaudio.h"
#include "mpc.h"
#include "mpcdata.h"
void ff_mpc_init(void)
{
ff_mpa_synth_init(ff_mpa_synth_window);
}
/**
* Process decoded Musepack data and produce PCM
*/
static void mpc_synth(MPCContext *c, int16_t *out, int channels)
{
int dither_state = 0;
int i, ch;
OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;
for(ch = 0; ch < channels; ch++){
samples_ptr = samples + ch;
for(i = 0; i < SAMPLES_PER_BAND; i++) {
ff_mpa_synth_filter(c->synth_buf[ch], &(c->synth_buf_offset[ch]),
ff_mpa_synth_window, &dither_state,
samples_ptr, channels,
c->sb_samples[ch][i]);
samples_ptr += 32 * channels;
}
}
for(i = 0; i < MPC_FRAME_SIZE*channels; i++)
*out++=samples[i];
}
void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data, int channels)
{
int i, j, ch;
Band *bands = c->bands;
int off;
float mul;
/* dequantize */
memset(c->sb_samples, 0, sizeof(c->sb_samples));
off = 0;
for(i = 0; i <= maxband; i++, off += SAMPLES_PER_BAND){
for(ch = 0; ch < 2; ch++){
if(bands[i].res[ch]){
j = 0;
mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0]];
for(; j < 12; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1]];
for(; j < 24; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
mul = mpc_CC[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2]];
for(; j < 36; j++)
c->sb_samples[ch][j][i] = mul * c->Q[ch][j + off];
}
}
if(bands[i].msf){
int t1, t2;
for(j = 0; j < SAMPLES_PER_BAND; j++){
t1 = c->sb_samples[0][j][i];
t2 = c->sb_samples[1][j][i];
c->sb_samples[0][j][i] = t1 + t2;
c->sb_samples[1][j][i] = t1 - t2;
}
}
}
mpc_synth(c, data, channels);
}
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