view mpegaudiodecheader.c @ 5306:abc5c130b448 libavcodec

AC-3 decoder, soc revision 32, Jul 17 09:37:32 2006 UTC by cloud9 Latest commit. There is no error in parsing and or recovering transform coefficients. Double checked with ac3dec. Getting consistent results with the bit allocation routine and transform coefficients. The code is able to parse valid ac3 bitstreams without error from start to end. I have also implemented the imdct when block switching is not enabled. However, can anybody provide an insight into how to convert float samples to int16_t ? lrint is of no help cuz it produces output -1, 0 or 1 whereas the output should be between -32768 to 32767.

/*
 * 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;
}