ffmpeg/libavformat/oggparseogm.c
Michael Niedermayer 268098d8b2 Merge remote-tracking branch 'qatar/master'
* qatar/master: (29 commits)
  amrwb: remove duplicate arguments from extrapolate_isf().
  amrwb: error out early if mode is invalid.
  h264: change underread for 10bit QPEL to overread.
  matroska: check buffer size for RM-style byte reordering.
  vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
  vp8: change int stride to ptrdiff_t stride.
  wma: fix invalid buffer size assumptions causing random overreads.
  Windows Media Audio Lossless decoder
  rv10/20: Fix slice overflow with checked bitstream reader.
  h263dec: Disallow width/height changing with frame threads.
  rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
  rmdec: Honor .RMF tag size rather than assuming 18.
  g722: Fix the QMF scaling
  r3d: don't set codec timebase.
  electronicarts: set timebase for tgv video.
  electronicarts: parse the framerate for cmv video.
  ogg: don't set codec timebase
  electronicarts: don't set codec timebase
  avs: don't set codec timebase
  wavpack: Fix an integer overflow
  ...

Conflicts:
	libavcodec/arm/vp8dsp_init_arm.c
	libavcodec/fraps.c
	libavcodec/h264.c
	libavcodec/mpeg4videodec.c
	libavcodec/mpegvideo.c
	libavcodec/msmpeg4.c
	libavcodec/pnmdec.c
	libavcodec/qpeg.c
	libavcodec/rawenc.c
	libavcodec/ulti.c
	libavcodec/vcr1.c
	libavcodec/version.h
	libavcodec/wmalosslessdec.c
	libavformat/electronicarts.c
	libswscale/ppc/yuv2rgb_altivec.c
	tests/ref/acodec/g722
	tests/ref/fate/ea-cmv

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-03 00:23:10 +01:00

199 lines
6.2 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];
const uint8_t *p = os->buf + os->pstart;
uint64_t time_unit;
uint64_t spu;
uint32_t size;
if(!(*p & 1))
return 0;
if(*p == 1) {
p++;
if(*p == 'v'){
int tag;
st->codec->codec_type = AVMEDIA_TYPE_VIDEO;
p += 8;
tag = bytestream_get_le32(&p);
st->codec->codec_id = ff_codec_get_id(ff_codec_bmp_tags, tag);
st->codec->codec_tag = tag;
} else if (*p == 't') {
st->codec->codec_type = AVMEDIA_TYPE_SUBTITLE;
st->codec->codec_id = CODEC_ID_TEXT;
p += 12;
} else {
uint8_t acid[5];
int cid;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
p += 8;
bytestream_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 != CODEC_ID_AAC)
st->need_parsing = AVSTREAM_PARSE_FULL;
}
size = bytestream_get_le32(&p);
size = FFMIN(size, os->psize);
time_unit = bytestream_get_le64(&p);
spu = bytestream_get_le64(&p);
p += 4; /* default_len */
p += 8; /* buffersize + bits_per_sample */
if(st->codec->codec_type == AVMEDIA_TYPE_VIDEO){
st->codec->width = bytestream_get_le32(&p);
st->codec->height = bytestream_get_le32(&p);
avpriv_set_pts_info(st, 64, spu * 10000000, time_unit);
} 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;
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
if (size >= 56 && st->codec->codec_id == CODEC_ID_AAC) {
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);
bytestream_get_buffer(&p, st->codec->extradata, size);
}
}
} else if (*p == 3) {
if (os->psize > 8)
ff_vorbis_comment(s, &st->metadata, p+7, os->psize-8);
}
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,
};
const struct ogg_codec ff_ogm_audio_codec = {
.magic = "\001audio",
.magicsize = 6,
.header = ogm_header,
.packet = ogm_packet,
.granule_is_start = 1,
};
const struct ogg_codec ff_ogm_text_codec = {
.magic = "\001text",
.magicsize = 5,
.header = ogm_header,
.packet = ogm_packet,
.granule_is_start = 1,
};
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,
};