view mpegaudiodecheader.c @ 5310:9aa9197034d7 libavcodec

AC-3 decoder, soc revision 40, Aug 9 00:10:14 2006 UTC by cloud9 More code cleanup. Window is now runtime generated. Fixed the bugs in rematrixing routine and in Decoding AC3 Bitstreams when coupling is in use. Still struggling to find out what affects the quality of the produced sound. Can anybody have a look at the imdct routines do_imdct_256 and do_imdct_512 and tell me whether it is the correctly implemented as described in standard.

/*
 * MPEG Audio header decoder
 * Copyright (c) 2001, 2002 Fabrice Bellard.
 *
 * 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 mpegaudiodecheader.c
 * MPEG Audio header decoder.
 */

//#define DEBUG
#include "avcodec.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"


int ff_mpegaudio_decode_header(MPADecodeContext *s, uint32_t header)
{
    int sample_rate, frame_size, mpeg25, padding;
    int sample_rate_index, bitrate_index;
    if (header & (1<<20)) {
        s->lsf = (header & (1<<19)) ? 0 : 1;
        mpeg25 = 0;
    } else {
        s->lsf = 1;
        mpeg25 = 1;
    }

    s->layer = 4 - ((header >> 17) & 3);
    /* extract frequency */
    sample_rate_index = (header >> 10) & 3;
    sample_rate = ff_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
    sample_rate_index += 3 * (s->lsf + mpeg25);
    s->sample_rate_index = sample_rate_index;
    s->error_protection = ((header >> 16) & 1) ^ 1;
    s->sample_rate = sample_rate;

    bitrate_index = (header >> 12) & 0xf;
    padding = (header >> 9) & 1;
    //extension = (header >> 8) & 1;
    s->mode = (header >> 6) & 3;
    s->mode_ext = (header >> 4) & 3;
    //copyright = (header >> 3) & 1;
    //original = (header >> 2) & 1;
    //emphasis = header & 3;

    if (s->mode == MPA_MONO)
        s->nb_channels = 1;
    else
        s->nb_channels = 2;

    if (bitrate_index != 0) {
        frame_size = ff_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];
        s->bit_rate = frame_size * 1000;
        switch(s->layer) {
        case 1:
            frame_size = (frame_size * 12000) / sample_rate;
            frame_size = (frame_size + padding) * 4;
            break;
        case 2:
            frame_size = (frame_size * 144000) / sample_rate;
            frame_size += padding;
            break;
        default:
        case 3:
            frame_size = (frame_size * 144000) / (sample_rate << s->lsf);
            frame_size += padding;
            break;
        }
        s->frame_size = frame_size;
    } else {
        /* if no frame size computed, signal it */
        return 1;
    }

#if defined(DEBUG)
    dprintf(s->avctx, "layer%d, %d Hz, %d kbits/s, ",
           s->layer, s->sample_rate, s->bit_rate);
    if (s->nb_channels == 2) {
        if (s->layer == 3) {
            if (s->mode_ext & MODE_EXT_MS_STEREO)
                dprintf(s->avctx, "ms-");
            if (s->mode_ext & MODE_EXT_I_STEREO)
                dprintf(s->avctx, "i-");
        }
        dprintf(s->avctx, "stereo");
    } else {
        dprintf(s->avctx, "mono");
    }
    dprintf(s->avctx, "\n");
#endif
    return 0;
}