/*

 * 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 <string.h>

#include "libavutil/common.h"

#include "mpegaudiodsp.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);

}