view mpc.c @ 10320:44f31c1c9acc libavcodec

Make sure that dv encoder initializes all encoded packet data. The specification does not say which value to use for unused parts, so fill all unused bytes with 0xff, which is consistent with what DV usually uses for reserved or unused parts.

/*
 * 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 libavcodec/mpc.c 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"

static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);

void ff_mpc_init(void)
{
    ff_mpa_synth_init(mpa_window);
}

/**
 * Process decoded Musepack data and produce PCM
 */
static void mpc_synth(MPCContext *c, int16_t *out)
{
    int dither_state = 0;
    int i, ch;
    OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE], *samples_ptr;

    for(ch = 0;  ch < 2; 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]),
                                mpa_window, &dither_state,
                                samples_ptr, 2,
                                c->sb_samples[ch][i]);
            samples_ptr += 64;
        }
    }
    for(i = 0; i < MPC_FRAME_SIZE*2; i++)
        *out++=samples[i];
}

void ff_mpc_dequantize_and_synth(MPCContext * c, int maxband, void *data)
{
    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);
}