view aac_parser.c @ 10273:38147f0f94cc libavcodec

Specify maximum sample rate of MLP by defining the factor relative to 48000 instead of directly. This makes clear that the code assumes the maximum sample rate to be a multiple of 48000 and also removes the division from the MAX_BLOCKSIZE macros, which causes an issue with the Solaris assembler where "/" is a comment marker unless the --divide option is used.

/*
 * 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 "aac_parser.h"
#include "get_bits.h"
#include "mpeg4audio.h"

int ff_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
{
    int size, rdb, ch, sr;
    int aot, crc_abs;

    if(get_bits(gbc, 12) != 0xfff)
        return AAC_AC3_PARSE_ERROR_SYNC;

    skip_bits1(gbc);             /* id */
    skip_bits(gbc, 2);           /* layer */
    crc_abs = get_bits1(gbc);    /* protection_absent */
    aot     = get_bits(gbc, 2);  /* profile_objecttype */
    sr      = get_bits(gbc, 4);  /* sample_frequency_index */
    if(!ff_mpeg4audio_sample_rates[sr])
        return AAC_AC3_PARSE_ERROR_SAMPLE_RATE;
    skip_bits1(gbc);             /* private_bit */
    ch      = get_bits(gbc, 3);  /* channel_configuration */

    skip_bits1(gbc);             /* original/copy */
    skip_bits1(gbc);             /* home */

    /* adts_variable_header */
    skip_bits1(gbc);             /* copyright_identification_bit */
    skip_bits1(gbc);             /* copyright_identification_start */
    size    = get_bits(gbc, 13); /* aac_frame_length */
    if(size < AAC_ADTS_HEADER_SIZE)
        return AAC_AC3_PARSE_ERROR_FRAME_SIZE;

    skip_bits(gbc, 11);          /* adts_buffer_fullness */
    rdb = get_bits(gbc, 2);      /* number_of_raw_data_blocks_in_frame */

    hdr->object_type    = aot + 1;
    hdr->chan_config    = ch;
    hdr->crc_absent     = crc_abs;
    hdr->num_aac_frames = rdb + 1;
    hdr->sampling_index = sr;
    hdr->sample_rate    = ff_mpeg4audio_sample_rates[sr];
    hdr->samples        = (rdb + 1) * 1024;
    hdr->bit_rate       = size * 8 * hdr->sample_rate / hdr->samples;

    return size;
}

static int aac_sync(uint64_t state, AACAC3ParseContext *hdr_info,
        int *need_next_header, int *new_frame_start)
{
    GetBitContext bits;
    AACADTSHeaderInfo hdr;
    int size;
    union {
        uint64_t u64;
        uint8_t  u8[8];
    } tmp;

    tmp.u64 = be2me_64(state);
    init_get_bits(&bits, tmp.u8+8-AAC_ADTS_HEADER_SIZE, AAC_ADTS_HEADER_SIZE * 8);

    if ((size = ff_aac_parse_header(&bits, &hdr)) < 0)
        return 0;
    *need_next_header = 0;
    *new_frame_start  = 1;
    hdr_info->sample_rate = hdr.sample_rate;
    hdr_info->channels    = ff_mpeg4audio_channels[hdr.chan_config];
    hdr_info->samples     = hdr.samples;
    hdr_info->bit_rate    = hdr.bit_rate;
    return size;
}

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


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