annotate aac_parser.c @ 5319:40af705cef7e libavcodec

AC-3 decoder, soc revision 69, Aug 31 07:12:56 2006 UTC by cloud9 Fix the bugs: 1. The quality of output because of incorrect windowing coefficients. New code for window generation. 2. Dynrng values were reset where dynrng value is present in the first block, but not in the subsequent block.

/*
 * Audio and Video frame extraction
 * Copyright (c) 2003 Fabrice Bellard.
 * Copyright (c) 2003 Michael Niedermayer.
 *
 * 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
 */

#include "parser.h"
#include "aac_ac3_parser.h"
#include "bitstream.h"


#define AAC_HEADER_SIZE 7


static const int aac_sample_rates[16] = {
    96000, 88200, 64000, 48000, 44100, 32000,
    24000, 22050, 16000, 12000, 11025, 8000, 7350
};

static const int aac_channels[8] = {
    0, 1, 2, 3, 4, 5, 6, 8
};


static int aac_sync(const uint8_t *buf, int *channels, int *sample_rate,
                    int *bit_rate, int *samples)
{
    GetBitContext bits;
    int size, rdb, ch, sr;

    init_get_bits(&bits, buf, AAC_HEADER_SIZE * 8);

    if(get_bits(&bits, 12) != 0xfff)
        return 0;

    skip_bits1(&bits);          /* id */
    skip_bits(&bits, 2);        /* layer */
    skip_bits1(&bits);          /* protection_absent */
    skip_bits(&bits, 2);        /* profile_objecttype */
    sr = get_bits(&bits, 4);    /* sample_frequency_index */
    if(!aac_sample_rates[sr])
        return 0;
    skip_bits1(&bits);          /* private_bit */
    ch = get_bits(&bits, 3);    /* channel_configuration */
    if(!aac_channels[ch])
        return 0;
    skip_bits1(&bits);          /* original/copy */
    skip_bits1(&bits);          /* home */

    /* adts_variable_header */
    skip_bits1(&bits);          /* copyright_identification_bit */
    skip_bits1(&bits);          /* copyright_identification_start */
    size = get_bits(&bits, 13); /* aac_frame_length */
    skip_bits(&bits, 11);       /* adts_buffer_fullness */
    rdb = get_bits(&bits, 2);   /* number_of_raw_data_blocks_in_frame */

    *channels = aac_channels[ch];
    *sample_rate = aac_sample_rates[sr];
    *samples = (rdb + 1) * 1024;
    *bit_rate = size * 8 * *sample_rate / *samples;

    return size;
}

static int aac_parse_init(AVCodecParserContext *s1)
{
    AACAC3ParseContext *s = s1->priv_data;
    s->inbuf_ptr = s->inbuf;
    s->header_size = AAC_HEADER_SIZE;
    s->sync = aac_sync;
    return 0;
}


AVCodecParser aac_parser = {
    { CODEC_ID_AAC },
    sizeof(AACAC3ParseContext),
    aac_parse_init,
    ff_aac_ac3_parse,
    NULL,
};