view aac_parser.c @ 5876:731ee5ad6bde libavcodec

Correct assignment of interlaced_frame; was being set on output frames, in display order, based on decoding information in decoding order. Now set properly, immediately upon completion of decode. Based on original patch from Reinhard Nissl, rnisssl % gmx , de Original Thread: [FFmpeg-devel] H.264 + PAFF: BBC HD recording shows extreme interlacing artefacts, Thu, 01 Nov 2007 22:43:09
author heydowns
date Mon, 05 Nov 2007 18:16:42 +0000
parents ced30500e2b1
children 48759bfbd073
line wrap: on
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/*
 * 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 */
    if(size < AAC_HEADER_SIZE)
        return 0;

    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,
};