libavcodec/mpc.c

/*
 * Musepack decoder core
 * Copyright (c) 2006 Konstantin Shishkov
 *
 * 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
 * Musepack decoder core
 * MPEG Audio Layer 1/2 -like codec with frames of 1152 samples
 * divided into 32 subbands.
 */

#include "libavutil/attributes.h"
#include "avcodec.h"
#include "mpegaudiodsp.h"
#include "mpegaudio.h"

#include "mpc.h"
#include "mpcdata.h"

/**
 * 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;

    for(ch = 0;  ch < channels; ch++){
        for(i = 0; i < SAMPLES_PER_BAND; i++) {
            ff_mpa_synth_filter_fixed(&c->mpadsp,
                                c->synth_buf[ch], &(c->synth_buf_offset[ch]),
                                ff_mpa_synth_window_fixed, &dither_state,
                                out[ch] + 32 * i, 1,
                                c->sb_samples[ch][i]);
        }
    }
}

void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, int16_t **out,
                                 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+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][0] & 0xFF];
                for(; j < 12; j++)
                    c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
                mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][1] & 0xFF];
                for(; j < 24; j++)
                    c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
                mul = (mpc_CC+1)[bands[i].res[ch]] * mpc_SCF[bands[i].scf_idx[ch][2] & 0xFF];
                for(; j < 36; j++)
                    c->sb_samples[ch][j][i] = av_clipf(mul * c->Q[ch][j + off], INT32_MIN, INT32_MAX);
            }
        }
        if(bands[i].msf){
            unsigned 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, out, channels);
}