ffmpeg/libavformat/oggparseogm.c
Michael Niedermayer af26185bdc Merge commit '163196562fe744149ef599d754c30c08a9898381' into release/1.1
* commit '163196562fe744149ef599d754c30c08a9898381':
  oggparseogm: Convert to use bytestream2
  rv34: Check the return value from ff_rv34_decode_init
  matroskadec: Verify realaudio codec parameters
  mace: Make sure that the channel count is set to a valid value
  svq3: Check for any negative return value from ff_h264_check_intra_pred_mode
  vp3: Check the framerate for validity
  cavsdec: Make sure a sequence header has been decoded before decoding pictures
  vocdec: Don't update codec parameters mid-stream
  sierravmd: Do sanity checking of frame sizes
  omadec: Properly check lengths before incrementing the position
  mpc8: Make sure the first stream exists before parsing the seek table

Conflicts:
	libavcodec/mace.c
	libavformat/oggparseogm.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-10-08 00:49:18 +02:00

205 lines
6.6 KiB
C

/**
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 "libavutil/avassert.h"
#include "libavutil/intreadwrite.h"
#include "libavcodec/get_bits.h"
#include "libavcodec/bytestream.h"
#include "avformat.h"
#include "internal.h"
#include "oggdec.h"
#include "riff.h"
static int
ogm_header(AVFormatContext *s, int idx)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *os = ogg->streams + idx;
AVStream *st = s->streams[idx];
GetByteContext p;
uint64_t time_unit;
uint64_t spu;
uint32_t size;
bytestream2_init(&p, os->buf + os->pstart, os->psize);
if (!(bytestream2_peek_byte(&p) & 1))
return 0;
if (bytestream2_peek_byte(&p) == 1) {
bytestream2_skip(&p, 1);
if (bytestream2_peek_byte(&p) == 'v'){
int tag;
st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
bytestream2_skip(&p, 8);
tag = bytestream2_get_le32(&p);
st->codec->codec_id = ff_codec_get_id(ff_codec_bmp_tags, tag);
st->codec->codec_tag = tag;
} else if (bytestream2_peek_byte(&p) == 't') {
st->codec->codec_type = AVMEDIA_TYPE_SUBTITLE;
st->codec->codec_id = AV_CODEC_ID_TEXT;
bytestream2_skip(&p, 12);
} else {
uint8_t acid[5] = { 0 };
int cid;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
bytestream2_skip(&p, 8);
bytestream2_get_buffer(&p, acid, 4);
acid[4] = 0;
cid = strtol(acid, NULL, 16);
st->codec->codec_id = ff_codec_get_id(ff_codec_wav_tags, cid);
// our parser completely breaks AAC in Ogg
if (st->codec->codec_id != AV_CODEC_ID_AAC)
st->need_parsing = AVSTREAM_PARSE_FULL;
}
size = bytestream2_get_le32(&p);
size = FFMIN(size, os->psize);
time_unit = bytestream2_get_le64(&p);
spu = bytestream2_get_le64(&p);
bytestream2_skip(&p, 4); /* default_len */
bytestream2_skip(&p, 8); /* buffersize + bits_per_sample */
if(st->codec->codec_type == AVMEDIA_TYPE_VIDEO){
st->codec->width = bytestream2_get_le32(&p);
st->codec->height = bytestream2_get_le32(&p);
avpriv_set_pts_info(st, 64, time_unit, spu * 10000000);
} else {
st->codec->channels = bytestream2_get_le16(&p);
bytestream2_skip(&p, 2); /* block_align */
st->codec->bit_rate = bytestream2_get_le32(&p) * 8;
st->codec->sample_rate = time_unit ? spu * 10000000 / time_unit : 0;
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
if (size >= 56 && st->codec->codec_id == AV_CODEC_ID_AAC) {
bytestream2_skip(&p, 4);
size -= 4;
}
if (size > 52) {
av_assert0(FF_INPUT_BUFFER_PADDING_SIZE <= 52);
size -= 52;
st->codec->extradata_size = size;
st->codec->extradata = av_malloc(size + FF_INPUT_BUFFER_PADDING_SIZE);
bytestream2_get_buffer(&p, st->codec->extradata, size);
}
}
} else if (bytestream2_peek_byte(&p) == 3) {
bytestream2_skip(&p, 7);
if (bytestream2_get_bytes_left(&p) > 1)
ff_vorbis_comment(s, &st->metadata, p.buffer, bytestream2_get_bytes_left(&p) - 1);
}
return 1;
}
static int
ogm_dshow_header(AVFormatContext *s, int idx)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *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 = AVMEDIA_TYPE_VIDEO;
st->codec->codec_id = ff_codec_get_id(ff_codec_bmp_tags, AV_RL32(p + 68));
avpriv_set_pts_info(st, 64, AV_RL64(p + 164), 10000000);
st->codec->width = AV_RL32(p + 176);
st->codec->height = AV_RL32(p + 180);
} else if(t == 0x05589f81){
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = ff_codec_get_id(ff_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)
{
struct ogg *ogg = s->priv_data;
struct ogg_stream *os = ogg->streams + idx;
uint8_t *p = os->buf + os->pstart;
int lb;
if(*p & 8)
os->pflags |= AV_PKT_FLAG_KEY;
lb = ((*p & 2) << 1) | ((*p >> 6) & 3);
os->pstart += lb + 1;
os->psize -= lb + 1;
while (lb--)
os->pduration += p[lb+1] << (lb*8);
return 0;
}
const struct ogg_codec ff_ogm_video_codec = {
.magic = "\001video",
.magicsize = 6,
.header = ogm_header,
.packet = ogm_packet,
.granule_is_start = 1,
.nb_header = 2,
};
const struct ogg_codec ff_ogm_audio_codec = {
.magic = "\001audio",
.magicsize = 6,
.header = ogm_header,
.packet = ogm_packet,
.granule_is_start = 1,
.nb_header = 2,
};
const struct ogg_codec ff_ogm_text_codec = {
.magic = "\001text",
.magicsize = 5,
.header = ogm_header,
.packet = ogm_packet,
.granule_is_start = 1,
.nb_header = 2,
};
const struct ogg_codec ff_ogm_old_codec = {
.magic = "\001Direct Show Samples embedded in Ogg",
.magicsize = 35,
.header = ogm_dshow_header,
.packet = ogm_packet,
.granule_is_start = 1,
.nb_header = 1,
};