As the name indicates we can't just assume what the
"opaque" field contains.
This fixes a crash in third-party applications see e.g.:
http://bugzilla.mplayerhq.hu/show_bug.cgi?id=2126
This fixes also FFmpeg trac #2293, which is a different
third-party application.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* commit '56daf10e0313c5e36f43e773f457d2a99ff0df10':
mov: use the format context for logging.
flicvideo: avoid an infinite loop in byte run compression
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avoid to write more than one cuepoint per track and PTS in
mkv_write_cues(). This avoids a later assertion failure on "(bytes >=
needed_bytes)" in put_ebml_num() called from end_ebml_master(), in case
there are several cuepoints per track with the same PTS.
This may happen with files containing packets with duplicated PTS in the
same track.
Adding support for parsing AlphaMode element in the Track header
and export that information as a metadata tag. This flag indicates
presence of alpha channel data in BlockAdditional element.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Other software does not store it in this case, and the information
is provided by the codec stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The QuickTime specification does not contain any hint that the atom
must not be written in some cases and both the QuickTime and the
AVID decoders do not fail if the atom is present.
This change allows to signal (visually) interlaced streams with
a codec different from uncompressed video.
As a side-effect, this fixes ticket #2202
Matroska specification lists support for BlockAdditional element
which is not supported by ffmpeg's matroska parser. This patch
adds grammar definitions for parsing that element (and few other
related elements) and then puts the data in AVPacket.side_data
with new AVPacketSideDataType AV_PKT_DATA_MATROSKA_BLOCKADDITIONAL.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '81726a4f0b8a43e19898e2a36fdde80583bafff0':
FATE: add tests for additional flavors of asf cover art
asfdec: do not assume every AVStream has a corresponding ASFStream
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '61f9ad2dfcb3f98b7ac5777d19d0e7b61d0be01e':
asfdec: read the full Metadata Object, not just aspect ratio information
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '36fab50e90d15352e403e4cc210890810f2fb4e2':
asfdec: silence a warning
mss4, ra288: Remove unused DSPContext local codec context members
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In some ASF files this objects holds cover art and other tags. Compared to
Metadata Object it can also hold GUIDs, but we ignore these for now.
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Use the same get_tag()/get_value() as for the Extended Content Description
but handle the 16 bit vs 32 bit difference for type 2 (BOOL)
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If its start is not aligned then aligning its end will
likely break many demuxers as they check the size and not
the position.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The DTS needs to be resynched against the segment start PTS, or the
resulting DTS may result < PTS.
Reported-by: Owen Jones <riots6@gmail.com>
See thread:
Subject: [FFmpeg-user] pts/dts error using reset_timestamps while splitting a DVD
Date: Sat, 19 Jan 2013 08:58:27 +0000
Modified the fate test crc generator to print the side_data's
crc if side_data is present.
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Replace wrong "EXT-X-ALLOWCACHE" with "EXT-X-ALLOW-CACHE", and value 1/0
with YES/NO, as per spec.
Fix trac ticket #2228.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
* qatar/master:
Use proper "" quotes for local header #includes
ppc: fmtconvert: Drop two unused variables.
bink demuxer: set framerate.
Conflicts:
libavcodec/kbdwin.c
libavcodec/ppc/fmtconvert_altivec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also check number of streams and give error message why muxing failed.
This prevents muxing unsupported codec with known and supported tag.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Memory passed to av_realloc cannot be allocated using memalign.
From realloc(3):
The realloc() function changes the size of the memory block pointed to
by ptr to size bytes. (...) Unless ptr is NULL, it must have been returned
by an earlier call to malloc(), calloc() or realloc().
The issue has been found by debugallocation, a part of google-perftools:
http://code.google.com/p/gperftools/ .
Signed-off-by: Paweł Hajdan, Jr <phajdan@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e6b1c3bbe7082c71ea8ee8ac83698c156c9e4838':
pthread: make ff_thread_release_buffer idempotent.
mvi: set framerate
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4a4a7e138c92901e04db46a6b05cc6948023e5f5':
rtpenc_chain: Use the original AVFormatContext for getting payload type
rtp: Make sure the output format pointer is set
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '57ed8debb9b9cc565cc6e9f98c5b5cbb9f69097c':
wmv2: Propagate the wmv2 idct permutation type to the dsputils context
rtp: Make sure priv_data is set before reading it
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In ff_rtp_get_payload_type, the AVFormatContext is used for checking
whether the payload_type or rtpflags options are set. In rtpenc_chain,
the rtpctx struct is a newly initialized struct where no options have
been set yet, so no options can be fetched from there.
All muxers that internally chain rtp muxers have the "rtpflags" field
that allows passing such options on (which is how this worked before
8034130e06), so this works just as intended.
This makes it possible to produce H263 in RFC2190 format with chained
RTP muxers.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Not sure if this actually happens, but we do the same check when
checking payload_type further above in the function, so it might
be needed.
Signed-off-by: Martin Storsjö <martin@martin.st>
This reverts 312645e :
"Do not set codec_tag property for matroska muxers."
Also adds dummy codec_tag lists with codecs
supported in mkv but not in wav / avi.
Fixes ticket #2169.
The check `start + res < start' is broken since pointer overflow is
undefined behavior in C. Many compilers such as gcc/clang optimize
away this check.
Use `res > end - start' instead. Also change `res' to unsigned int
to avoid signed left-shift overflow.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
A negative `size' will bypass FFMIN(). In the subsequent memcpy() call,
`size' will be considered as a large positive value, leading to a buffer
overflow.
Change the type of `size' to unsigned int to avoid buffer overflow, and
simplify overflow checks accordingly. Also change a literal buffer
size to use sizeof, and limit the amount of data copied in another
memcpy call as well.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Sanity checks like `data + size >= data_end || data + size < data' are
broken, because `data + size < data' assumes pointer overflow, which is
undefined behavior in C. Many compilers such as gcc/clang optimize such
checks away.
Use `size < 0 || size >= data_end - data' instead.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
A negative `size' will bypass FFMIN(). In the subsequent memcpy() call,
`size' will be considered as a large positive value, leading to a buffer
overflow.
Change the type of `size' to unsigned int to avoid buffer overflow, and
simplify overflow checks accordingly.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Sanity checks like `data + size >= data_end || data + size < data' are
broken, because `data + size < data' assumes pointer overflow, which is
undefined behavior in C. Many compilers such as gcc/clang optimize such
checks away.
Use `size < 0 || size >= data_end - data' instead.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The check `start + res < start' is broken since pointer overflow is
undefined behavior in C. Many compilers such as gcc/clang optimize
away this check.
Use `res > end - start' instead. Also change `res' to unsigned int
to avoid signed left-shift overflow.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '4f56e773fe8a554b8c2662650aaf799c2ece2721':
x86: ac3: Fix HAVE_MMXEXT condition to only refer to external assembly
rtpenc: Start the sequence numbers from a random offset
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Expose the current sequence number via an AVOption - this can
be used both for setting the initial sequence number, or for
querying the current number.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use AVERROR_INVALIDDATA on invalid inputs, and AVERROR_EOF when no more
frames are available in an interleaved AVI.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* commit '8a4f26206d7914eaf2903954ce97cb7686933382':
dsputil: remove butterflies_float_interleave.
srtp: Move a variable to a local scope
srtp: Add tests for the crypto suite with 32/80 bit HMAC
Conflicts:
libavcodec/x86/dsputil.asm
libavcodec/x86/dsputil_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3ef6d22e1ba544ab37c73e8fc61382f13aac250f':
srtp: cosmetics: Use fewer lines for the test vectors
srtp: Don't require more input data than what actually is needed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a2a991b2ddf951454ffceb7bcedc9db93e26c610':
srtp: Improve the minimum encryption buffer size check
srtp: Add support for a few DTLS-SRTP related crypto suites
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f53490cc0c809975f8238d5a9edbd26f83bd2f84':
rtpdec/srtp: Handle CSRC fields being present
rtpdec: Check the return value from av_new_packet
ac3dec: fix non-optimal dithering of zero bit mantissas
Conflicts:
libavcodec/ac3dec.c
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c6f1dc8e4cd967ae056698eafb891a08003c211c':
rtpdec: Move setting the parsing flags to the actual depacketizers
rtpdec: Split handling of mpeg12 audio/video to a separate depacketizer
Conflicts:
libavformat/rtpdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'a717f9904227d7979473bad40c50eb40af41d01d':
mpegts: Share the cleanup code between the demuxer and lavf-internal parser functions
rtpdec_mpeg4: Return one AAC AU per AVPacket
ppc: Include string.h for memset
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This clarifies where the limit number comes from, and only
requires exactly as much padding space as will be needed.
Signed-off-by: Martin Storsjö <martin@martin.st>
The theoretical minimum for a (not totally well formed) RTCP packet
is 8 bytes, so we shouldn't require 12 bytes as minimum input.
Also return AVERROR_INVALIDDATA instead of 0 if something that is
not a proper packet is given.
Signed-off-by: Martin Storsjö <martin@martin.st>
The main difference to the existing suites from RFC 4568 is
that the version with a 32 bit HMAC still uses 80 bit HMAC
for RTCP packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
Several compilers such as clang/icc/pathscale will optimize the check
pos + size < pos (assuming size > 0) into false, since signed integer
overflow is undefined behavior in C. This breaks overflow checking.
Use a safe precondition check instead.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The lavf-internal parser functions are used when receiving
mpegts over RTP. This fixes memory leaks in this setup.
The normal mpegts demuxer close function was updated in ec7d0d2e in
2004 to fix leaks, but the parsing function used for RTP wasn't
updated and has been leaking ever since.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the returned data valid to stream copy into other
containers as well, not only for decoding straight away.
Signed-off-by: Martin Storsjö <martin@martin.st>
Makes ff_id3v2_read reset stream position at the end of ID3 data if the
header size is not matched (caused by an EOF for example).
Current behaviour (without the patch):
filesize = 400
id3 data size = 399
file offset after ff_id3v2_read is 400 instead of 399
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Write the packet unaltered if found.
Fixes ticket #1917
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The code did not account properly for packets that where added to
the end of the packet list. Also flags for such packets where not
set correctly leading to incorrect chunked interleaving.
Reported-by: bcoudurier
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The exact packing of Opus inside Matroska is not finalized.
Use A_OPUS/EXPERIMENTAL as codec name, like mkvtoolnix.
The A_OPUS name stays to let ffmpeg open files it has produced
until now, but newly produced file use the EXPERIMENTAL version.
Once the spec is stabilized it will be possible to consider
options to ensure compatibility with these files.
a small value was rounded to 0 and then treated special as if
chunked_duration was 0. This led to a inconsistency that further led
to wrong interleaving
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
