view oggparseogm.c @ 3062:9a56dcf7edfb libavformat

Fix issue357 Do what the spec says, insane or not: " Format 0 (uncompressed) and Format 3 (uncompressed little-endian) are similar. Both encode uncompressed audio samples. For 8-bit samples, the two formats are identical. For 16-bit samples, the two formats differ in byte ordering. In Format 0, 16-bit samples are encoded and decoded according to the native byte ordering of the platform on which the encoder and Flash Player, respectively, are running. In Format 3, 16-bit samples are always encoded in little-endian order (least significant byte first), and are byte-swapped if necessary in Flash Player before playback. Format 0 is clearly disadvantageous because it introduces a playback platform dependency. For 16-bit samples, Format 3 is highly preferable to Format 0 for SWF version 4 or later. "
author michael
date Sun, 24 Feb 2008 01:04:00 +0000
parents c3de842d7ff5
children 6f61c3b36632
line wrap: on
line source

/**
    Copyright (C) 2005  Michael Ahlberg, Måns Rullgård

    Permission is hereby granted, free of charge, to any person
    obtaining a copy of this software and associated documentation
    files (the "Software"), to deal in the Software without
    restriction, including without limitation the rights to use, copy,
    modify, merge, publish, distribute, sublicense, and/or sell copies
    of the Software, and to permit persons to whom the Software is
    furnished to do so, subject to the following conditions:

    The above copyright notice and this permission notice shall be
    included in all copies or substantial portions of the Software.

    THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
    EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
    MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
    NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
    HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
    WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
    OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
    DEALINGS IN THE SOFTWARE.
**/

#include <stdlib.h>
#include "avformat.h"
#include "bitstream.h"
#include "bytestream.h"
#include "intreadwrite.h"
#include "oggdec.h"
#include "riff.h"

static int
ogm_header(AVFormatContext *s, int idx)
{
    ogg_t *ogg = s->priv_data;
    ogg_stream_t *os = ogg->streams + idx;
    AVStream *st = s->streams[idx];
    const uint8_t *p = os->buf + os->pstart;
    uint64_t time_unit;
    uint64_t spu;
    uint32_t default_len;

    if(!(*p & 1))
        return 0;
    if(*p != 1)
        return 1;

    p++;

    if(*p == 'v'){
        int tag;
        st->codec->codec_type = CODEC_TYPE_VIDEO;
        p += 8;
        tag = bytestream_get_le32(&p);
        st->codec->codec_id = codec_get_id(codec_bmp_tags, tag);
        st->codec->codec_tag = tag;
    } else if (*p == 't') {
        st->codec->codec_type = CODEC_TYPE_SUBTITLE;
        st->codec->codec_id = CODEC_ID_TEXT;
        p += 12;
    } else {
        uint8_t acid[5];
        int cid;
        st->codec->codec_type = CODEC_TYPE_AUDIO;
        p += 8;
        bytestream_get_buffer(&p, acid, 4);
        acid[4] = 0;
        cid = strtol(acid, NULL, 16);
        st->codec->codec_id = codec_get_id(codec_wav_tags, cid);
    }

    p += 4;                     /* useless size field */

    time_unit   = bytestream_get_le64(&p);
    spu         = bytestream_get_le64(&p);
    default_len = bytestream_get_le32(&p);

    p += 8;                     /* buffersize + bits_per_sample */

    if(st->codec->codec_type == CODEC_TYPE_VIDEO){
        st->codec->width = bytestream_get_le32(&p);
        st->codec->height = bytestream_get_le32(&p);
        st->codec->time_base.den = spu * 10000000;
        st->codec->time_base.num = time_unit;
        st->time_base = st->codec->time_base;
    } else {
        st->codec->channels = bytestream_get_le16(&p);
        p += 2;                 /* block_align */
        st->codec->bit_rate = bytestream_get_le32(&p) * 8;
        st->codec->sample_rate = spu * 10000000 / time_unit;
        st->time_base.num = 1;
        st->time_base.den = st->codec->sample_rate;
    }

    return 1;
}

static int
ogm_dshow_header(AVFormatContext *s, int idx)
{
    ogg_t *ogg = s->priv_data;
    ogg_stream_t *os = ogg->streams + idx;
    AVStream *st = s->streams[idx];
    uint8_t *p = os->buf + os->pstart;
    uint32_t t;

    if(!(*p & 1))
        return 0;
    if(*p != 1)
        return 1;

    t = AV_RL32(p + 96);

    if(t == 0x05589f80){
        st->codec->codec_type = CODEC_TYPE_VIDEO;
        st->codec->codec_id = codec_get_id(codec_bmp_tags, AV_RL32(p + 68));
        st->codec->time_base.den = 10000000;
        st->codec->time_base.num = AV_RL64(p + 164);
        st->codec->width = AV_RL32(p + 176);
        st->codec->height = AV_RL32(p + 180);
    } else if(t == 0x05589f81){
        st->codec->codec_type = CODEC_TYPE_AUDIO;
        st->codec->codec_id = codec_get_id(codec_wav_tags, AV_RL16(p + 124));
        st->codec->channels = AV_RL16(p + 126);
        st->codec->sample_rate = AV_RL32(p + 128);
        st->codec->bit_rate = AV_RL32(p + 132) * 8;
    }

    return 1;
}

static int
ogm_packet(AVFormatContext *s, int idx)
{
    ogg_t *ogg = s->priv_data;
    ogg_stream_t *os = ogg->streams + idx;
    uint8_t *p = os->buf + os->pstart;
    int lb;

    if(*p & 8)
        os->pflags |= PKT_FLAG_KEY;

    lb = ((*p & 2) << 1) | ((*p >> 6) & 3);
    os->pstart += lb + 1;
    os->psize -= lb + 1;

    return 0;
}

ogg_codec_t ogm_video_codec = {
    .magic = "\001video",
    .magicsize = 6,
    .header = ogm_header,
    .packet = ogm_packet
};

ogg_codec_t ogm_audio_codec = {
    .magic = "\001audio",
    .magicsize = 6,
    .header = ogm_header,
    .packet = ogm_packet
};

ogg_codec_t ogm_text_codec = {
    .magic = "\001text",
    .magicsize = 5,
    .header = ogm_header,
    .packet = ogm_packet
};

ogg_codec_t ogm_old_codec = {
    .magic = "\001Direct Show Samples embedded in Ogg",
    .magicsize = 35,
    .header = ogm_dshow_header,
    .packet = ogm_packet
};