annotate mpc.c @ 11864:7204cb7dd601 libavcodec

Quant changes only once per MB so move the corresponding scale factor assignment out of the block decoding loop. Indeo4 doesn't use any scale table but the quant level itself as scale. Therefore access scale table only if its pointer != NULL.

/*
 * 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 "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 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]),
                                ff_mpa_synth_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);
}