In this loop, |i| is the "index". And the memory allocated should be at
least the current "count", which is |i + 1|.
BUG=801821
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: 7738135736989908991 - -7898362169240453118 cannot be represented in type 'long'
Fixes: Chromium bug 796778
Reported-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: runtime error: signed integer overflow: 5 * -2147483647 cannot be represented in type 'int'
Fixes: Chromium bug 817338
Reviewed-by: Matt Wolenetz <wolenetz@google.com>
Reported-by: Matt Wolenetz <wolenetz@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
ff_get_extradata() frees any existing extradata before allocating now,
and using av_free() here leaves a dangling pointer that will result in
a double free.
Fixes a regression since 0ca33b1d4e.
Tested-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Numeric hosts in certificates are not very common, but supported by LibreSSL.
Forward the IP address to make verification work in this case.
Fixes ticket #7029
This prevents leaks in the rare cases the function is called when extradata
already exists.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
mov_read_ctts ignores ctts entries having count <= 0. Generally, the
aggregate of all ctts entries' count fields resulting from mov_read_ctts
can be less than the corresponding sample_count.
mov_build_index attempts to normalize any existing ctts_data counts to
be 1, to make a 1-1 mapping of a ctts_data entry to a sample.
That 1-1 mapping left a tail of uninitialized ctts_data entries when the
aggregate, normalized ctts_count < sample_count.
Even more generally, later usage of ctts_data may depend on the entire
ctts_allocated_size having been initialized.
This change memsets the entire allocation of the normalized ctts_data in
mov_build_index, to prevent use of uninitialized data later.
BUG=816787
Change-Id: I7fd7db255e3aeed076ee32c90cb2df211741c052
Reviewed-on: https://chromium-review.googlesource.com/947110
Reviewed-by: Xiaohan Wang <xhwang@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
fix ticket: #7044
Get the first video stream to reference for split segment
when there have more than one video stream
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Reviewed-by: Karthick Jeyapal <kjeyapal@akamai.com>
The allocated size of sc->ctts_data is
(st->nb_index_entries + entries) * sizeof(*sc->ctts_data).
The size to memset at offset sc->ctts_data + sc->ctts_count should be
(st->nb_index_entries + entries - sc->ctts_count) *
sizeof(*sc->ctts_data))
The current code missed |entries| I believe, which was introduced in
https://patchwork.ffmpeg.org/patch/5541/.
However, after offline discussion, it seems the original code is much
more clear to read (before https://patchwork.ffmpeg.org/patch/5541/).
Hence this CL revert the memset logic to it's previous state by
remembering the |old_ctts_allocated_size|, and only memset the newly
allocated entries.
BUG=812567
Change-Id: Ibe94c7138e5818bfaae76866bfa6619a9b8a2b6b
Reviewed-on: https://chromium-review.googlesource.com/934925
Reviewed-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Obviously this is still not perfect, but better then it was. Using the first
index table and mxf->current_edit_unit is still hardcoded in many places, so
this change has hopefully the less chance of breaking anything that works
now.
Signed-off-by: Marton Balint <cus@passwd.hu>
Use body_offset of the partitions to search for the partition with the given
offset in the essence. This makes the function find the correct partition for
non frame-wrapped essences as well, where only the essence data is part of the
the edit unit byte count, not the KLV-s.
Signed-off-by: Marton Balint <cus@passwd.hu>
In order to do that we have to parse the EssenceContainerData and assign the
proper body_sid and index_sid to the tracks from the corresponding source
packages.
This fixes packets returned in the wrong stream for some OP1-b files.
Based on a patch by Alex Mogurenko from https://github.com/da8eat/FFmpeg
Reference: http://mogurenko.com/2018/01/02/mxf-op1b-ffmpeg-part1/
Signed-off-by: Marton Balint <cus@passwd.hu>
The reference point for a KAG is the first byte of the key of a Partition Pack.
Fixes ticket #2817.
Fixes ticket #5317.
Signed-off-by: Marton Balint <cus@passwd.hu>
This only affected demuxers that didn't return reference counted packets.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Prevents use of uninitialized values.
Fixes ticket #7038.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
The supplementary audio descriptor is defined in ETSI EN 300 468 and
provides more details regarding accessibility audio tracks, especially
the normative annex J contains a detailed description of its use.
Its language code (if present) overrides the language code of an also
present ISO 639 language descriptor.
Note that this also changes the priority of multiple descriptors with
language from "the last descriptor with language within the ES loop" to
"specific descriptor over general ISO 639 descriptor".
Signed-off-by: Aman Gupta <aman@tmm1.net>
Currently using HLS_TEMP together with HLS_SECOND_LEVEL_SEGMENT_DURATION
or HLS_SECOND_LEVEL_SEGMENT_SIZE gives error at end of each segment
writing and the final segment file names do not contain the desired
data. This patch fixes this bug by delaying the initilization of
original segment filename after actual temp file renaming will skip the
interfering.
Signed-off-by: Bela Bodecs <bodecsb@vivanet.hu>
we found some very old videos which suffered from
corruption after 9e6a242755, but were fine
before.
These had "End of AC stream reached in vp6_parse_coeff" warnings in logs.
These also had flv Packet mismatch warnings.
Adding FlixEngine to the list of flv muxers which produce broken packet
sizes fixes this corruption.
FlixEngine is very old and not maintained or available anymore (since
2010), so we won't need to worry about newer versions fixing the issue.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The MPEG-PS muxer only accepts PCM streams having up to 8 channels
and the following sampling rates: 32/44.1/48/96 kHz.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Defined in a recent revision of https://www.webmproject.org/docs/container/
This prevents storing the contents of CodecPrivate into extradata for
a codec that doesn't need nor expect any. It will among other things
prevent matroska specific binary data from being dumped onto other
formats during remuxing.
Signed-off-by: James Almer <jamrial@gmail.com>
PCM_S16BE stream packets in MPEG-PS have a 3-byte header and
are recognized as PCM_DVD by the demuxer which prevents their
correct remuxing in MPEG-1/2 PS.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This makes sure no field is ever used uninitialized.
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes two warnings:
libavformat/rtpdec.c:155:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
libavformat/rtpdec.c:168:20: warning: return discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
An index should never be equal to the count. Hence we must make sure
*ctts_index < ctts_count.
Reviewed-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If a subdemuxer has the updated metadata event flag set, the metadata is copied
to the corresponding stream. The flag is cleared on the subdemuxer and the
appropriate event flag is set on the stream.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
avcodec bump missed in 7e8eba2d87
avformat bump missed in ff46124b0d and
0694d87024
avdevice bump missed in 0fd475704e
Signed-off-by: James Almer <jamrial@gmail.com>
Demuxing only. Muxing is disabled as altref frame handling is not
defined in the spec, and there's no way to know the presence of
such frames during stream initialization.
Based on a patch by Steven Liu.
Fixes ticket #7000
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
The ID3v2 "unsynchronization scheme" requires replacing any 0xFF 0x00
sequences with 0xFF. This has to be done on every byte of the source
data, while the current code skipped a byte after a replacement. This
meant 0xFF 0x00 0xFF 00 was translated to 0xFF 0xFF 0x00 instead of 0xFF
0xFF. It feels a bit messy to do this correctly with the avio use. But
fortunately, this translation can be done in-place, so we can just do it
in memory.
Inspired by what taglib does.
Also see 9ae80e6a9c. (The sample file for
that commit is gone, so it could not be retested.)
The "timeout" option name inherently clashes with the meaning of the
HTTP libavformat protocol option with the same name. Rename it after a
deprecation period to make it compatible with the HTTP one.
There's at least one known file with a TrueHD stream that hasn't
been correctly muxed, and requires full frame parsing and repack.
Signed-off-by: James Almer <jamrial@gmail.com>
If first_seq_no is not within the bounds of timelines then a segfault
will occur.
This patch removes the use of first_seq_no within the timelines array
It also adds first_seq_no to the value returned by calc_next_seg_no_from_timelines
(which allows for different values of 'startNumber')
Signed-off-by: Brendan McGrath <redmcg@redmandi.dyndns.org>
This will replace the 1024 character limited filename field. Compatiblity for
output contexts are provided by copying filename field to URL if URL is unset
and by providing an internal function for muxers to set both url and filename
at once.
Signed-off-by: Marton Balint <cus@passwd.hu>
AVERROR_EXIT happens when the user's interrupt callback signals that
playback should be aborted. In this case, the demuxer shouldn't print a
warning, as it's expected that all network accesses are stopped.
The seek function can just return an error if seeking is unavailable,
but often this is too late. Add a flag that signals that the stream is
unseekable, and use it in HLS.
This makes little sense due to how HLS works, and only causes some
additional annoyances if the HLS read_seek function fails (for example
if it's a live stream). It was most likely unintended.
Currently http end of chunk is signalled implicitly in dashenc_io_open().
This mean playlists http writes would have to wait upto a segment duration to signal end of chunk causing delays.
This patch will fix that problem and improve performance.
Enables getting access to ID3 PRIV tags from the command-line or metadata API
when demuxing. The PRIV owner is stored as the metadata key prepended with
"id3v2_priv.", and the data is stored as the metadata value. As PRIV tags may
contain arbitrary data, non-printable characters, including NULL bytes, are
escaped as \xXX.
Similarly, any metadata tags that begin with "id3v2_priv." are inserted as ID3
PRIV tags into the output (assuming the format supports ID3). \xXX sequences in
the value are un-escaped to their byte value.
Signed-off-by: wm4 <nfxjfg@googlemail.com>
1 Add function 'resolve_content_path' to propagate the baseURL from
upper level nodes.
* if no baseURL is available, the path of mpd file will be set as the baseURL.
2 Remove checking for newly established connection.
3 Establish the communication protocol in each connection rather than
applying one protocol to all connection.
If no representation bandwidth value is set, the url value returned
by get_content_url is corrupt (as it has been freed).
This change ensures the url string is not freed unless it is about
to be reused
Changes since v1:
1 removed the unneeded 'if' statement (as pointed out by Michael Niedermayer
2 added comment to make it clear why the av_free was required
Signed-off-by: Brendan McGrath <redmcg@redmandi.dyndns.org>
If codecpar->extradata is not set (for example, when the stream goes
through the 'tee' muxer), then a segfault occurs.
This patch ensures the data variable is not null before attempting
to access it
Before the var_stream_map option was available - I was using the tee
muxer to create each resolution as an individual stream.
When running this configuration after the most recent hlsenc change
I hit a segfault
The most simple command which recreates the segfault is:
ffmpeg -i in.ts -map 0:a -map 0:v -c:a aac -c:v h264 -f tee [select=\'a,v\':f=hls]tv_hls_hd.m3u8
Signed-off-by: Brendan McGrath <redmcg@redmandi.dyndns.org>
It was sort of optional before - if you didn't call it, networking was
initialized on demand, and an ugly warning was logged. Also, the doxygen
comments threatened that it would be made strictly required one day.
Make it explicitly optional. I would prefer to deprecate it fully, but
there might still be legitimate reasons to use this. But the average
user won't need it.
This is needed only for two reasons: to initialize TLS libraries like
OpenSSL and GnuTLS, and winsock.
OpenSSL and GnuTLS were already silently initialized on demand if the
global network init function was not called. They also have various
thread-safety acrobatics, which make concurrent initialization within
libavformat safe. In addition, the libraries are moving towards making
their global init functions safe, which removes all need for central
global init. In particular, GnuTLS 3.5.16 and OpenSSL 1.1.0g have been
found to have safe init functions. In all cases, they use internal
reference counters to avoid that the global uninit functions interfere
with concurrent uses of the library by other API users who called global
init.
winsock should be thread-safe as well, and maintains an internal
reference counter as well.
Since we still support ancient TLS libraries, which do not have this
fixed, and since it's unknown whether winsock and GnuTLS
reinitialization is costly in any way, don't deprecate the libavformat
functions yet.
The condition was a bit too long, and most editors will break the line
and turn it into an unreadable mess. Move out some of the conditions.
This should not change the behavior.
dash_read_seek() is called only once to issue a seek
of *all* streams to the specified timestamp. But to
avoid reopening each stream, do a "dry run" for streams
that are in a discarded state.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The primary goal was making it viable to play YouTube/Vimeo/... videos
using the native demuxer, since mpv currently uses a workaround in form
of Edit Decision Lists (EDL).
Implemented features:
1 Exposing id / bitrate as stream metadata (similar to the HLS demuxer)
2 Support for multiple video and audio streams
3 A few minor parts of the specification that are in use at YouTube
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
libopenmpt 0.3 deprecates openmpt_module_create_from_memory() and
provides a replacement function openmpt_module_create_from_memory2().
Detecting libopenmpt 0.3 can be done at build time via the API
version macros provided by libopenmpt. libopenmpt 0.2 did not provide
all required macros, however libopenmpt documents the required #define
shims that can be safely added for libopenmpt 0.2.
Using openmpt_module_create_from_memory2() instead of
openmpt_module_create_from_memory() avoids the deprecation warning
when building ffmpeg with libopenmpt 0.3.
openmpt_module_create_from_memory2() provides more fine-grained error
reporting and in particular allows distinguishing out-of-memory from
input file parsing errors. Return appropriate ffmpeg errors
accordingly.
libopenmpt 0.3 is ABI and API compatible with applications built
against libopenmpt 0.2. Building ffmpeg with libopenmpt 0.2 is still
supported.
Signed-off-by: Jörn Heusipp <osmanx@problemloesungsmaschine.de>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Apple's AIFF protocol clearly states that each chucnk which is odd sized a padding should be added.
In the old version of aiffdec adding of padding was done in `get_meta`. And in case of unknown chunk name it was done in defalut case.
The new version has deleted the padding in default case and added padding adding after the switch.
But the new version didn't removed the padding adding in the `get_meta` function so in some cases padding was added twice which leaded to a bug.
Fixes: sample.aiff
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Since the function used to be exposed for ffserver's sake, it was renamed
to the same name as a system's aton function which caused a collision.
Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
Certain AIFF files encode XA ADPCM compressed audio using a chunk
with the tag `APCM`. Aside from this custom chunk type, they're
otherwise standard AIFF files. I've only observed these files in the
Sega Saturn game Sonic Jam so far.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If the stream was aborted using the libavformat interrupt callback, we
don't want it to log the reconnect warning. (Exiting after logging this
warning worked well, so this is only for avoiding the ugly warning.)
It makes no sense to return an error after the first reconnect, and then
somehow resume the next time it's called. Usually this will lead to
demuxer errors. Make reconnecting block instead, until it has either
successfully reconnected, or given up.
Also make the wait reasonably interruptible. Since there is no mechanism
for this in the API, polling is the best we can do. This behaves roughly
the same as other interruptible network functions in libavformat.
(The original code would work if it returned AVERROR(EAGAIN) or so,
which would make retry_transfer_wrapper() repeat the read call. But I
think having an explicit loop for this is better anyway.)
I also snuck in a fix for reconnect_at_eof. It has to check for
AVERROR_EOF, not 0.
fix CID: 1426991
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
There is no POSIX error code for EOF - recv() signals EOF by simply
returning 0. But libavformat recently changed its conventions and
requires an explicit AVERROR_EOF, or it might get into an endless retry
loop, consuming 100% CPU while doing nothing.
pts_wrap_bits defaults to 33 (like MPEG), that causes valid
timestamps to be unwrapped and become invalid.
Inspired by a patch by Wu Zhiqiang <mymoeyard@gmail.com>.
Encrypted HLS segments have regular http:// urls, but open_input()
actually prefixes them with crypto+ before calling open_url(), so
they end up using the crypto protocol and not the http protocol.
This means invoking ff_http_do_new_request will fail, so we avoid
calling it in the first place. After the earlier http.c commit,
the failure results in a warning printed to the user. In earlier
versions, the failure would cause a segfault.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This is required for AV playout from master.m3u8.
Otherwise master.m3u8 lists only video-only and/or audio-only streams.
Reviewed-by: Steven Liu <lq@chinaffmpeg.org>
Some http/1.0 implementations, like python's SimpleHTTPServer, can only support one client connection at a time. Making a second request while the first is still connected leads to a deadlock.
This change enables multiple connections for http/1.1 servers only, which need to support keepalive by default and should have no problem with concurrent requests.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Can be used by the api user to figure out what http features the server supports based on the response received.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This makes do_new_request fail early when dealing with a http/1.0 server, avoiding unnecessary "reconnecting" warnings shown to the user.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Fixes compile error when building with network or protocols disabled.
This code would never be reached (because the demuxer fails much earlier on http playlists or segments), so it doesn't matter much what we do here as long as it compiles.
Signed-off-by: Aman Gupta <aman@tmm1.net>
It's completely absurd that libavcodec would care about libavformat
locking, but it was there because the lock manager was in libavcodec.
This is more stright forward. Changes ABI, but we don't require ABI
compatibility currently.
Currently http end of chunk is signalled implicitly in hlsenc_io_open().
This mean playlists http writes would have to wait upto a segment duration to signal end of chunk causing delays.
This patch will fix that problem and improve performance.
AVERROR_EOF is an internal error which means the http socket is no longer
valid for new requests. It informs the caller that a new connection must
be established, and as such does not need to be surfaced to the user as
a warning.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This fixes a deadlock when using the hls demuxer's new http_persistent feature
to stream a youtube live stream over HTTPS. The youtube servers are http/1.1
compliant, but return a "Connecton: close". Before this commit, the demuxer
would attempt to send a new request on the partially shutdown connection and
cause a deadlock in the tls protocol.
Signed-off-by: Aman Gupta <aman@tmm1.net>
This improves network throughput of the hls demuxer by avoiding
the latency introduced by downloading segments one at a time.
The problem is particularly noticable over high-latency network
connections: for instance, if RTT is 250ms, there will a 250ms idle
period between when one segment response is read and the next one
starts.
The obvious solution to this is to use HTTP pipelining, where a
second request can be sent (on the persistent http/1.1 connection)
before the first response is fully read. Unfortunately the way the
http protocol is implemented in avformat makes implementing pipleining
very complex.
Instead, this commit simulates pipelining using two separate persistent
http connections. This has the advantage of working independently of
the http_persistent option, and can be used with http/1.0 servers as
well. The pair of connections is swapped every time a new segment starts
downloading, and a request for the next segment is sent on the secondary
connection right away. This means the second response will be ready and
waiting by the time the current response is fully read.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
This teaches the HLS demuxer to use the HTTP protocols
multiple_requests=1 option, to take advantage of "Connection:
Keep-Alive" when downloading playlists and segments from the HLS server.
With the new option, you can avoid TCP connection and TLS negotiation
overhead, which is particularly beneficial when streaming via a
high-latency internet connection.
Similar to the http_persistent option recently implemented in hlsenc.c
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
This mimics logging that was added in 53e0d5d724 for security
purposes.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This will prevent improper use of ff_http_do_new_request() if the user
tries to send a request for a different host to a previously connected
persistent http/1.1 connection.
Signed-off-by: Aman Gupta <aman@tmm1.net>
Signed-off-by: Karthick J <kjeyapal@akamai.com>
Before this patch persistent http connections would work only for media segments.
The playlists were still opening a new connection everytime.
This patch extends persistent http connections to playlists as well.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Since close is not called, during http persistent connection,
flush needs to be called so that output is written on time.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
With FCPX 10.4, Apple has expanded the set of colorspace, primaries,
and trc flags officially supported in QuickTime files. The expanded set
matches the codepoints used in ffmpeg and many other specs.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Every bitstream filter behaves as intended now, so there's no need to
wait for the first packet of every stream.
Signed-off-by: James Almer <jamrial@gmail.com>
The current edit unit cannot be reliably determined for the last packet of a
video stream, because we can't query the start offset of the next edit unit
from the index. This caused missing timestamps for the last video packet.
Therefore from now on, we allow setting the PTS even if we are not sure of the
current edit unit if mxf_set_current_edit_unit returned a specific failure, and
the assumed current edit unit is the last.
Fixes last packet timestamp of:
ffprobe -fflags nofillin -show_packets tests/data/lavf/lavf.mxf -select_streams v
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes ticket #6804. All of the ogg header and packet parsers may
return standard AVERROR codes; these return values should not be
treated as success.
Additionally changes oggparsevorbis, to not give up too early
with certain types of poorly muxed files.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
An invalid file may cause huge alloc. Delay expansion of ctts entries
until the number of samples is known in mov_build_index.
Fixes: 23
Found-by: zhao dongzhuo, AD-lab of Venustech
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix suggested by Luca Barbato.
This was causing spurious EOFs when using -rtsp_transport udp, as
reported in https://bugzilla.libav.org/show_bug.cgi?id=1103
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This is to take full advantage of Common Media Application Format(CMAF).
Now server can generate one content and serve both HLS and DASH players.
Reviewed-by: Steven Liu <lq@onvideo.cn>
On some failure paths, the error code is not correctly set.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In function mov_read_custom(), it returns 0 on the path that av_malloc()
returns a NULL pointer. 0 indicates success. An error code should be
assigned to ret.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This patch fixes Bug #6868
Sometimes end_pts is getting initialized to audio stream's
first pts, while the duration is calculated based on video stream's pts.
In this patch the end_pts is initialized with the correct stream's first pts.
Reviewed-by: Steven Liu <lq@onvideo.cn>
Tested-by: beloko
The function avformat_new_stream() returns a NULL pointer on failure.
However, in function rtp_mpegts_write_header(), its return value is not
validated before it is dereferenced. Check the return value against NULL
to avoid potential NULL dereference.
Signed-off-by: Pan Bian <bianpan2016@163.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Prevents out of array accesses. Adressess ticket #6873
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Writes one set of field framing information for progressive streams and
two sets for interlaced streams. Fixes ticket #6383.
Unfortunately the OpenDML v1.02 document is not very specific on what
value to use for start_line when frame data is not coming from a
capturing device, so this is just using 0/1 depending on the field order
as a best-effort guess.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
