According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
This avoids initializing a stream with dummy values or when the file does not
contain audio.
Also set duration for audio packets, using the sample rate as the time base.
Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Prefix the functions/tables brktimegm, pcm_read_seek,
dv_offset_reset, voc_get_packet, codec_movaudio_tags,
codec_movvideo_tags.
After this, lavf has no global symbols without the proper prefix.
Signed-off-by: Martin Storsjö <martin@martin.st>
Keep the old protocol name around for backwards compatibility
until the next bump.
Deprecate the method of implicitly assuming the nested protocol.
For applehttp://server/path, it might have felt logical, but
supporting hls://server/path isn't quite as intuitive. Therefore
only support hls+http://server/path from now on.
Using this protocol at all is discouraged, since the hls demuxer
is more complete and fits into the architecture better. There
have been cases where the protocol implementation worked better
than the demuxer, but this should no longer be the case.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this demuxer was created, there didn't seem to be any
consensus of a common short name for this protocol. Now
the consensus seems to be to call it hls.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows easily differentiating between both implementations within the build
system and combining the native implementation for plain RTMP with librtmp for
the RTMPE, RTMPS, RTMPT, RTMPTE protocol variants.
If muxing into mpegts, 4 byte startcodes for the first NAL
of an access unit is required. Thus it is simplest for the
RTP depacketizer to just use 4 byte startcodes everywhere.
Signed-off-by: Martin Storsjö <martin@martin.st>
When segmenting the output from the mpegts muxer, one can
now set this option when cutting to a new segment, to make sure
the next segment starts with PAT/PMT/SDT.
Signed-off-by: Martin Storsjö <martin@martin.st>
The last entry is the total size of the essence container.
Previously a TemporalOffset error would be logged, even though
segments like these are expected.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
These are common with audio atoms. Without this the demuxer would read two
bytes at a time for a mono 16-bit file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Specially crafted files can lead the parsing code to take too long.
We fix a lot of these problems by not allowing local tags to extend
past the end of the set and not allowing other KLVs to be read past
the end of themselves.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
This can happen if an index table segment has a very large IndexStartPosition.
zzuf3.mxf is an example of such a file.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Only the OPAtom demuxing logic is guaranteed to have index tables,
meaning OP1a files that lack an index would cause SIGSEGV.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Calculates based on total file size and wavetaillength from the header.
Falls back to multiplying finalframeblocks by 8 instead of 4 so that it will
at least be overestimating for 24-bit. Currently it can underestimate the
final packet size, leading to decoding errors.
Right now those muxers use the default timebase in all cases(1/90000).
This patch avoid unnecessary rescaling and makes the printed timestamps
more readable.
Also, extend the printed information to include the timebases and packet
pts/duration and align the columns.
Obviously changes the results of all fate tests which use those two
muxers.
Normally, the actual payload data contains sequence headers, too,
and the parser can extract this and set it as extradata. However,
the data in the dvc1 atom is the "official" extradata for the file.
This is required for proper stream copy of vc1 from ismv to ismv.
Signed-off-by: Martin Storsjö <martin@martin.st>
Do not use AVStream's duration for dts generation since it contains in
some cases the duration of the whole file instead of duration of the
samples in the moov. This happens if the mdhd holds the duration of the
whole file but has no entries or a zero duration in its stts.
dv: Fix null pointer dereference due to ach=0
Fixes part2 of CVE-2011-3929
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
dv: check stype
Fixes part1 of CVE-2011-3929
Possibly fixes part of CVE-2011-3936
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Roman Shaposhnik <roman@shaposhnik.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().
It also allows to set codec private options using av_opt_set_* before
opening the codec.
It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().
This makes the first packet of a track fragment run to get
the keyframe flag set properly if sample_degradation_priority
is nonzero.
This makes the keyframes flag be set properly for ismv files
created by Microsoft.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, we've only passed the key string on to the recursive
amf_parse_object for the mixedarray type, not for 'object'. By
passing the key string on, the recursive amf_parse_object can
store the amf objects as metadata.
This kind of data was seen in data from XSplit Broadcaster, received
over RTMP via Wowza. This patch allows reading this metadata.
Signed-off-by: Martin Storsjö <martin@martin.st>
Check results for av_malloc() and fix an overflow in one call.
Related to CVE-2011-3940.
Based in part on work from Michael Niedermayer.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Fixes CVE-2011-3940 (Out of bounds read resulting in out of bounds write)
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
(cherry picked from commit 5c011706bc)
Signed-off-by: Alex Converse <alex.converse@gmail.com>
-vbsf doesn't exist anymore. It got renamed to -bsf somewhere along the
line. Update print statement accordingly.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Current demuxer recognizes several colorspace formats that begin with 'C420'
but does not yet recognize plain 'C420'. GStreamer's y4menc component
generates .y4m files with a 'C420' colorspace. This new comparison is
placed after the other 'C420' checks so that it doesn't interfere with
them.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In this mode, no seeks will be done except for within moov/moof
fragments, which should fit within the AVIOContext buffer.
This allows pushing live smooth streaming format data to
a live publishing point on IIS over http.
Signed-off-by: Martin Storsjö <martin@martin.st>
If no data was seen for a stream decoder are returning 0 when fed with
empty packets for flushing. We can stop flushing when the decoder does
not return delayed delayed frames anymore. Changes try_decode_frame()
return value to got_picture or negative error.
CC: libav-stable@libav.org
The field frame_size isn't written to the output anywhere except
than in mov.
This facilitates stream copy from formats that don't set frame_size.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add some space around operators and wrap a comment
that extends past the 80 char "limit"/guideline.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can also optionally split the file into individual fragments,
which allows it to be served from any web server without any
server side support.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes calculation of trackDuration if the MOVIentry array
is cleared. This is required by the fragmentation support in the
next patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
It is a really bad idea to assign a video codec id
when we have set codec_type to audio and vice versa.
Prevents detection of mp2 in mxf as mpeg2video.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Specifically, this means parsing as before until we run into essence.
At that point we seek to the footer and parse until EOF. After that we start
seeking backward to the previous partition and parse that until we run into
essence or the next partition. This procedure is repeated until we encounter
the last partition we parsed in the forward direction.
The end result of all this is that large essence containers are not needlessly
parsed. This speeds up parsing large files a lot.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This fixes 0001GL.MXF.V1.mxf_opatom.mxf and 0001GL00.MXF.A1.mxf_opatom.mxf
getting two streams each due to both using the same SourcePackageID.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
Based on patch from Tomas Härdin <tomas.hardin@codemill.se>
and work by Georg Lippitsch <georg.lippitsch@gmx.at>
Changed av_calloc to av_mallocz and added overflow checks.
This fixes reading of partition packs. The code stops reading after the
operational pattern and should skip the array of essence container
labels that follow.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.
This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.
Signed-off-by: Martin Storsjö <martin@martin.st>
This check isn't relevant in the way the code currently works.
Also change a case of if (x == 0) into if (!x).
Signed-off-by: Martin Storsjö <martin@martin.st>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
Originally, sizeof(struct MOVIentry) was 48, after the reordering,
it is 40 in my build configuration.
When writing really long mov/mp4 files, this can make a difference
- this saves a bit over 2 MB of memory per hour of video (down to
10.3 MB per hour from 12.3 MB per hour initially) for a video with
75 packets per second - 25 fps + 50 audio packets (which is the
case for AMR audio).
Signed-off-by: Martin Storsjö <martin@martin.st>
The H.264 decoder needs SPS and PPS for initialization during
multi-threaded decoding. When probed single-threaded SPS and PPS are
copied to extradata and are available for proper initialization of
the decoder before the first frame is decoded.
This isn't used in practice anywhere within libav at the moment,
but change it for consistency until it is removed.
URL_RDONLY/WRONLY were fixed in commit 5b81e29593 (after the
values that actually were used were changed at the major bump,
in commit cbea3ac8), but this flag was unintentionally left unfixed.
Signed-off-by: Martin Storsjö <martin@martin.st>
If the creation time is stored in the file as a zero, the
mov demuxer skips exporting the creation time. Currently,
files muxed without a creation time get demuxed with a
Jan 1st 1970 creation timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
rtsp.h relies on network.h and the latter conditionally defines fallback OS
structures that rely on configure tests, which are only run if networking
is enabled.
Align IEC 61937 length_code for DTS-HD so that
(length_code & 0xf) == 0x8. This is reportedly needed with some
receivers.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The implicit network initialization is set to be removed in the
future, but is kept for compatibility. By not doing the implicit
initialization for non-network protocols, we avoid the warning
about avformat_network_init() not being called for these, where
it really doesn't make much sense.
Signed-off-by: Martin Storsjö <martin@martin.st>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes an invalid free() with ass in avi. The sample in bug 98 passes
parts of AVPacket.data as buffer for the AVIOContext. Since the packet
is quite large fill_buffer tries to reallocate the buffer before doing
nothing. Fixes bug 98.
The fate-h264-bsf-mp4toannexb failures were caused by an integer
overflow of the unneeded multiplication.
Inspired by patch by: Michael Niedermayer <michaelni@gmx.at>
According to draft-pantos-http-live-streaming-07, 6.3.4,
the duration of the last media segment in the playlist
should be used as initial minimum reload delay.
Signed-off-by: Martin Storsjö <martin@martin.st>
With the current default PES packet size, and very small audio bitrates,
audio packet duration gets too long. For players, which wait for a whole
audio packet (or more) it takes a very long time to start playing sound.
For 24kbps audio, one PES packet is about 1 second long. On Motorola STBs,
we observe about 3 second delay before the playback starts with the
default setting.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Do not assume the audio packets being always smaller than
DEFAULT_PES_PAYLOAD_SIZE.
Signed-off-by: Jindřich Makovička <makovick@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The 'fiel' atoms can be found in H.264 tracks clobbering the extradata.
MJPEG supports non field based extradata, and this data should be
preserved when copying.
This fixes demuxing of file where the first packet is not audio. Such files
are generated by our idroq muxer. It also fixes demuxing of audio only
idroq files.
Compared to just overwriting the old extradata, this has the
advantage of letting the decoder know exactly when the
extradata changed (otherwise it is changed immediately when the
new extradata packet is demuxed, even if there's old queued packets
awaiting to be decoded). This makes it easier for decoders to
actually react to the change, so they won't have to inspect
the extradata for each packet to see if it might have changed.
This works when sequentially playing a file with sample rate
changes, but if seeking past a new extradata packet in the
file, it obviously doesn't work properly. That case doesn't
work in flash player either, so it's probably ok not to handle
it.
Signed-off-by: Martin Storsjö <martin@martin.st>
