When parsing the Xing/Info tag, don't set the bit rate if it's an Info tag.
When parsing the stream, don't override the bit rate if it's already set,
otherwise calculate the mean bit rate from parsed frames. This way, the bit
rate will be set correctly both for CBR and VBR streams.
CC:libav-stable@libav.org
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Fixes use of uninitialized variables and possible out of array accesses
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 6cc12353a8.
Conflicts:
libavformat/version.h
Allowing to automatically select the concat demuxer raises
security concerns, as it allows a possibly hostile file to
access any file on the system. Guessing the format based on
the file name extension does not allow to enable the safe
mode designed to avoid it.
When parsing the Xing/Info tag, don't set the bit rate if it's an Info tag.
When parsing the stream, don't override the bit rate if it's already set,
otherwise calculate the mean bit rate from parsed frames. This way, the bit
rate will be set correctly both for CBR and VBR streams.
Signed-off-by: Alexander Kojevnikov <alexander@kojevnikov.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* cehoyos/master:
buildsys: only include log2_tab per library for shared builds
Add h264chroma dependency for cavs decoder to configure.
Add h264qpel dependency for snow codec to configure.
Add h264chroma dependency for vp5 and vp6 decoder to configure.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
As far as I can tell the code should not change behaviour
depending on locale in any of these places.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The warnings are false positives, older gcc versions (such as 4.5)
think the variables can be used uninitialized while they in
practice can't, while newer (4.6) gets it right.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'e671d3ad6cd7fe1d02e9b35b889a25d8c059fce9':
h264: do not copy ref count/ref2frm when updating per-frame context
flvdec: Check the return value of a malloc
Conflicts:
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c91c63b5380bf79655c09320774a022f84d76fd5':
flvdec: Don't read the VP6 header byte when setting codec type based on metadata
Conflicts:
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The callers of this function can't report errors sanely. If this
one malloc fails, don't write the extradata byte, make sure we
try to malloc it the next time we're called instead, and make sure
we still consume the input data byte.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This header byte is only present when actually reading a VP6 frame,
not when reading the codec type field in the metadata. This
potential bug has been present since 5b54a90c.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
If the first "special" character in a filename is a comma,
it can introduce protocol options, but only if there is a
colon at the end. Otherwise, it is just a filename with a
comma.
Fix trac ticket #2303.
Fix trac ticket #2300 because the duration of the segments
was computed using the timestamp of the last packet plus its
duration using the 1/90000 default time base instead of using
the chained muxer time base.
* qatar/master:
lavf: Add a fate test for the noproxy pattern matching
lavf: Handle the environment variable no_proxy more properly
Conflicts:
libavformat/Makefile
libavformat/internal.h
libavformat/tls.c
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 4a8fc1d83b.
The commit caused null pointer derefernces when using udp://
after i fixed that it caused ffmpeg to get stuck and remapped
arguments like ?ttl=255 -> ?ttl%3d255
I dont want to leave this broken thus temporary revert so we all
have some time to look at this without half the network protocols
being broken in the meantime
The handling of the environment variable no_proxy, present since
one of the initial commits (de6d9b6404), is inconsistent with
how many other applications and libraries interpret this
variable. Its bare presence does not indicate that the use of
proxies should be skipped, but it is some sort of pattern for
hosts that does not need using a proxy (e.g. for a local network).
As investigated by Rudolf Polzer, different libraries handle this
in different ways, some supporting IP address masks, some supporting
arbitrary globbing using *, some just checking that the pattern matches
the end of the hostname without regard for whether it actually is
the right domain or a domain that ends in the same string.
This simple logic should be pretty similar to the logic used by
lynx and curl.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
* qatar/master:
lavc: Move vector_fmul_window to AVFloatDSPContext
rtpdec_mpeg4: Check the remaining amount of data before reading
Conflicts:
libavcodec/dsputil.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '977d4a3b8a2dbc2fb5e747c7072485016c9cdfaa':
rtpdec_mpeg4: Check the return value from malloc
srtp: Mark a few variables as uninitialized
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0eecafc948b74c247ebbc59f18f508db5d590d0b':
configure: Make the new srtp protocol depend on the rtp protocol
lavf: Add a fate test for the SRTP functions
lavu: Add a fate test for the HMAC API
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Attempting to re-parse the headers at demuxer level is a
pandora box the way its done currently.
This allows full reconfiguration of vorbis streams
Fixes Ticket2117
Fixes Ticket2121
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Untested, due to lack of rtp stream with CSRCs, but the code as
is does not work with multiple CSRCs
Reviewed-by: Luca Abeni <lucabe72@email.it>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '094a7405e5d8463d7d167d893e04934ec1a84ecd':
x86: ABSB: port to cpuflags
sdp: Include SRTP crypto params if using the srtp protocol
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ab2ad8bd56882c0ea160b154e8b836eb71abc49d':
lavf: Add functions for SRTP decryption/encryption
lavu: Add an API for calculating HMAC (RFC 2104)
Conflicts:
doc/APIchanges
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3f111804eb5c603a344706b84b7164cbf7b4e0df':
libvpx: make vp8 and vp9 selectable
libvpx: support vp9
nut: support vp9 tag
mkv: support vp9 tag
rtpdec: Make variables that should wrap unsigned
Conflicts:
configure
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/avcodec.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ba0c72a9ae1e2954e5dcf920f7b4e9a8f8a22f3e':
build: Remove stray Makefile entry for non-existent VCR1 encoder
rtpdec: Handle more received packets than expected when sending RR
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'd0fe217e3990b003b3b3f2c2daaadfb2af590def':
rtpdec: Simplify insertion into the linked list queue
rtpdec: Remove a woefully misplaced comment
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.
If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
This supports the AES_CM_128_HMAC_SHA1_80 and
AES_CM_128_HMAC_SHA1_32 cipher suites (from RFC 4568) at the
moment. The main missing features are replay protection (which can be
added later without changing the internal API), and the F8 and null
ciphers.
Signed-off-by: Martin Storsjö <martin@martin.st>
The function is a callback that is called by ff_gen_search with
a constant stream index.
Avoid a false positive on older gcc version.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This makes the behaviour defined when they wrap around. The value
assigned to expected_prior was a uint32_t already.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without this, we'd signal a huge loss rate (due to unsigned
wraparound) if we had received one packet more than expected (that
is, one seq number sent twice). The code has a check for lost_interval
<= 0, but that doesn't do what was intended as long as the variable is
unsigned.
Signed-off-by: Martin Storsjö <martin@martin.st>
The code below the comment does not at all relate to statistics,
and even if moved to the right place, the comment adds little
value.
Signed-off-by: Martin Storsjö <martin@martin.st>
"analyzeduration" is not used to detect the input duration, but to
specify the max probe data duration. Fix option description and related
doc entry accordingly.
* commit 'abae27ed3acd0a7c54f11760c5be2d2653c4edf8':
rtpdec: Fix the calculation of expected number of packets
fate: vp3: Fix fate-vp3-coeff-level64 test dependencies
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously, we always signalled a zero time since the last RTCP
SR, which is dubious.
The code also suggested that this would be the difference in
RTP NTP time units (32.32 fixed point), while it actually is
in in 1/65536 second units. (RFC 3550 section 6.4.1)
Signed-off-by: Martin Storsjö <martin@martin.st>
This brings back some code that was added originally in 4a6cc061
but never was used, and was removed as unused in 4cc843fa. The
code is updated to actually work and is tested to return sane
values.
Signed-off-by: Martin Storsjö <martin@martin.st>
The base_seq variable is set to first_seq - 1 (in
rtp_init_sequence), so no + 1 is needed here.
This avoids reporting 1 lost packet from the start.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
rtsp: Remove an outdated comment
rtsp: Remove references to weirdly named variables in other files
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'c44784c9bb9d0ddf5d39d0dfa640816a57b8f457':
rtp: Rename a static variable to normal naming conventions
rtp: Cosmetic cleanup
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The question can be answered: No, we do not know the initial sequence
number from the SDP. In certain cases, it can be known from the
RTP-Info response header in RTSP though. (In that case, we use it as
timestamp origin, but not for rtp receiver statistics.)
Signed-off-by: Martin Storsjö <martin@martin.st>
It is unclear what the bug exactly was and if it ever was fixed,
and we don't even support decoding via faad any longer. The
comment has been present since d0deedcb in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
One of them is renamed now, but mentioning it by name serves
no purpose here. The other table mentioned ceased to exist
under that name in 4934884a1 in 2006.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes division by zero
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Reviewed-by: Matthieu Bouron <matthieu.bouron@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '7b8c5b263bc680eff5710bee5994de39d47fc15e':
vc1dec: prevent a crash due missing pred_flag parameter
matroska: Fix use after free
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtpdec_vp8: Don't trim too much data from broken frames
rtpdec_vp8: Simplify code by using an existing helper function
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'ed79093222ceb42f0c3a39095a69af0b32be5450':
rtpdec: Add a terminating null byte at the end of the SDES/CNAME
yuv4mpeg: do not use deprecated functions
oggdec: fix faulty cleanup prototype
idcin: return 0 from idcin_read_packet() on success.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
