Also make sure the existing length check can't overflow.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that it doesn't try to free an AVBuffer belonging
to an earlier packet when we free the local packet at the end.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids divisions by zero later.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Null buffers are useful for simulating writing to a real buffer
for the sake of measuring how many bytes are written.
Signed-off-by: Martin Storsjö <martin@martin.st>
ASF markers only have a start time, so we lose the chapter end times,
but that is ASF for you
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
this was forgotten when we changed ASF to not output the preroll time
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Abort if it is invalid if strict error checking has been requested.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the output fragments independent of their position in
the output stream, making the output work better when streamed.
QuickTime Player doesn't support fragmented mp4 without the base
data offset, though.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is a bit more work, but avoids having to fill in
the data offset field afterwards instead of directly when
the rest of the trun atom is written.
This simplifies future cases where this field needs to be set to
something different.
Signed-off-by: Martin Storsjö <martin@martin.st>
A given packet won't always come in contiguously; sometimes
they may be broken up on chunk boundaries by packets of another
channel.
This support primarily involves tracking information about the
data that's been read, so the reader can pick up where it left
off for a given channel.
As a side effect, we no longer over-report the bytes read if
(toread = MIN(size, chunk_size)) == size
Signed-off-by: Martin Storsjö <martin@martin.st>
Since the number of channels is multiplied by 36 and assigned to
to a uint16_t, make sure this calculation didn't overflow. (In
certain cases the calculation could overflow leaving the
truncated block_align at 0, leading to divisions by zero later.)
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Some files have the duration set to -1 in the mdhd atom, more
or less legitimately. (We produce such files ourselves, for the
initial duration in fragmented mp4 files.)
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This more closely corresponds to the usage of the field.
Its usage here is unrelated to the channel ID.
Signed-off-by: Martin Storsjö <martin@martin.st>
Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.
This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.
Signed-off-by: Martin Storsjö <martin@martin.st>
Sending non-monotonic packets (e.g. when the audio and video
streams are monotonic within themselves but not muxed
monotonically) will lead to negative values the RTMP timestamp
field (where timestamps are transmitted only as deltas for each
channel), and this delta can end up being incorrectly written as
a large unsigned number.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since 596e5d4783, this is not necessary anymore. It also allows to
actually disable the flushing, improving write performance (but
possibly giving worse latency in real-time streaming).
Signed-off-by: Martin Storsjö <martin@martin.st>
This is enabled by default and can be disabled with
"-fflags -flush_packets".
Inspired by a patch from Nicolas George <nicolas.george@normalesup.org>.
Signed-off-by: Martin Storsjö <martin@martin.st>
If we really want to support parameter changes, they need to be
signalled along with the AVPackets as parameter change side data,
not just changing the AVCodecContext parameters when a packet
is demuxed (since there may be other earlier packets yet undecoded).
Something similar was already done for the sample rate in 0883109b2,
but some parameters were left changeable.
This avoids having to recheck the channel count for validity for
each decoded frame in (ad)pcm decoders, unless the decoders
explicitly say that they accept parameter changes.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + BYTES_PER_FRAME_RECORD won't overflow.
Also factorize other existing error return paths.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Limit the size to INT_MAX/2 (for simplicity) to be sure that
size + FF_INPUT_BUFFER_PADDING_SIZE won't overflow.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The seektable is required for filling in ape->frames[i].pos
further down.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure the faststart vs fragmentation check works as
intended when fragmentation is enabled due to using the ismv mode.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes warnings about making integers from pointers without
a cast, and avoids the theoretical case where the lower 32 bits of
the pointer would all be zero where the implicit cast wouldn't give
the right result.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows creation of frame accurate chapter marks from sources like
DVD and BD where the precise chapter location is not known until the
chapter mark has been reached during reading.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This should improve write performance quite significantly.
---
Tested with both writing a normal mp4, by using the faststart
feature and writing a fragmented mp4 file; all turn out with the
same md5sum as before.
Signed-off-by: Martin Storsjö <martin@martin.st>
Remove the header decoding for PCM audio from mpeg.c and the
20/24bit parts from pcm.c and merge them into a new decoder in
pcm-dvd.c.
The decoder has added support for samples that span multiple
packets and modified 20/24bit group decoding. Both is needed to
decode samples that have been generated with DVD-Lab Pro 2. The
decoding of 16bit PCM and two channel 24bit is identical to
before. No other samples are known to verify the correctness of
the encoding this software does.
The complete list of tested formats is
48kHz/16bit/2-8 channels
48kHz/24bit/2-5 channels
96kHz/16bit/2-4 channels
96kHz/24bit/2 channels
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
When streaming to limelight, the app name is either a full
"appname/subaccount" or "appname/_definst_". In the latter case,
the app name can be simplified into simply "appname", but the
authentication hashing assumes the /_definst_ still to be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
If a client tries to read the file while it's being updated, the client
would get an incomplete manifest. Instead write to a separate temp file
and atomically rename it to replace the previous one.
Signed-off-by: Martin Storsjö <martin@martin.st>
The element was only being written when the value == 1. But the default
value of this element is 1, so this has no useful effect. This element
needs to be written when the value == 0.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
On failures in the write_trailer function, we could also ignore
the errors and try to finish the file despite these errors (which
would only leave an incomplete chapters track). It's probably better
to signal the error clearly to the caller though (and if this
function failed there's no guarantee that there's enough memory to
finish the trailer either).
Signed-off-by: Martin Storsjö <martin@martin.st>
QuickTime will play multiple audio tracks concurrently if this flag is
set for multiple audio tracks. And if no subtitle track has this flag
set, QuickTime will show no subtitles in the subtitle menu.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Faststart moves the moov atom to the beginning of the file and rewrites
the rest of the file after muxing is complete.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows creation of frame accurate chapter marks from sources
like DVD and BD where the precise chapter location is not known until
the chapter mark has been reached during reading.
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow emitting the current cluster that is being written before
starting a new one, simplifying how to figure out where clusters
are positioned in the output stream (for live streaming).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Seeking in certain broken files would cause ogg_read_timestamp
to fail because ogg_packet would go into a state where all packets
of stream 1 would be discarded until the end of the stream.
Bug-Id: 553
CC: libav-stable@libav.org
Signed-off-by: Jan Gerber <j@v2v.cc>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mov/mp4 muxer has support for handling negative timestamps
via edit lists (which customarily is used for handling the 1-frame
delay due to B-frames as well).
Using the muxer's native way of handling it is better than using
the generic offsetting. The generic offsetting is a bit too
crude when e.g. the timebase of one track is 1/fps, where the
edit lists can handle it accurately.
Signed-off-by: Martin Storsjö <martin@martin.st>
The counter itself shouldn't be wrapped, since it is used for
determining end_pts for the next segment - only wrap the number
used for the segment file name.
Signed-off-by: Martin Storsjö <martin@martin.st>
The hls muxer itself doesn't have any direct (object file level)
dependencies on mpegtsenc.o, and including that object file
directly doesn't ensure that it is registered so that the muxer
actually is accessible.
Signed-off-by: Martin Storsjö <martin@martin.st>
IPPROTO_IPV6 is unrelated here (it's only used in udp.c for
multicast sockopts), check for support for the sockaddr_in6
struct itself.
Signed-off-by: Martin Storsjö <martin@martin.st>
An SDP description normally only contains the target IP address
and port for the packets. This means that we don't really have
any clue where to send the RTCP RR packets - previously they're
sent to the destination IP written in the SDP (at the same port),
which rarely is the actual peer. And if the source for the packets
is on a different port than the destination, it's never correct.
With a new option, we can choose to send the packets to the
address that the latest packet on each socket arrived from.
---
Some may even argue that this should be the default - perhaps,
but I'd rather keep it optional at first. Additionally, I'm not
sure if sending RTCP RR directly back to the source is
desireable for e.g. multicast.
Signed-off-by: Martin Storsjö <martin@martin.st>
If we've received packets on the same socket before, the return
packets are sent to that address. If we've only received packets
on the other socket, try to guess the source port for the other
one assuming the basic +1/-1 logic.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the sources documentation up below the marker for deprecated
otpions. Also mention the new block parameter, that was added
in 749722209.
Signed-off-by: Martin Storsjö <martin@martin.st>
It is possible to have an initial broken header and then valid packets.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Add one copy of the function into each of the libraries, similarly
to what we do for log2_tab. When using static libs, only one
copy of the file_open.o object file gets included, while when
using shared libraries, each of them get a copy of its own.
This fixes DLL builds with a statically linked C runtime, where
each DLL effectively has got its own instance of the C runtime,
where file descriptors can't be shared across runtimes.
On systems not using msvcrt, the function is not duplicated.
Signed-off-by: Martin Storsjö <martin@martin.st>
This supports non-Linux systems (SOCK_CLOEXEC is non-standard) and
older Linux kernels to the extent possible.
Signed-off-by: Martin Storsjö <martin@martin.st>
When libavformat was changed to use the new avpriv_open function
in 51eb213d00, this silently bypassed the existing wrapper for
win32. Move the win32 wrapper into libavutil/file.c to make sure
it gets called everywhere (not just in the libavformat case).
This makes sure that non-ascii file names gets opened properly
(where file names internally are stored as utf8, but they get
converted to wchar_t and opened with _wsopen).
Signed-off-by: Martin Storsjö <martin@martin.st>
This provides at least some protection against potential accidental
corruption of AVIO buffer workspace.
Signed-off-by: Martin Storsjö <martin@martin.st>
It's only relevant for the RTSP demuxer. Similarly, the custom_io
flag is only present in the SDP demuxer options list.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also clear the AVIOContext handle after freeing, to avoid
possible dangling pointers if the later call fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
This lowers the level of warnings printed if trying to connect
to a host name that provides both v6 and v4 addresses but the
service only is available on the v4 address (often occurring for
'localhost', with servers that aren't v6-aware).
Signed-off-by: Martin Storsjö <martin@martin.st>
The common case of the pointer having increased by one packet (which results
in no change to the modulus) can be detected with a 64-bit subtraction,
which is far cheaper than a division on many platforms.
Before After
Mean StdDev Mean StdDev Change
Divisions 248.3 8.8 51.5 7.4 +381.7%
Overall 2773.2 25.6 2372.5 43.1 +16.9%
Signed-off-by: Martin Storsjö <martin@martin.st>
When a stream contains a single program, there's no point in doing a
PID -> program lookup. Normally the one and only program isn't disabled,
so no packets should be discarded.
Before After
Mean StdDev Mean StdDev Change
discard_pid() 73.8 9.4 20.2 1.5 +264.8%
Overall 2300.8 28.0 2253.1 20.6 +2.1%
Signed-off-by: Martin Storsjö <martin@martin.st>
This was being performed to ensure that a complete packet was held in
contiguous memory, prior to parsing the packet. However, the source buffer
is typically large enough that the packet was already contiguous, so it is
beneficial to return the packet by reference in most cases.
Before After
Mean StdDev Mean StdDev Change
memcpy 720.7 32.7 649.8 25.1 +10.9%
Overall 2372.7 46.1 2291.7 21.8 +3.5%
Signed-off-by: Martin Storsjö <martin@martin.st>
As long as there is enough contiguous data in the avio buffer,
just return a pointer to it instead of copying it to the caller
provided buffer.
Signed-off-by: Martin Storsjö <martin@martin.st>
A separate rtcp port can already be set when opening the rtp
protocol normally, but when doing port setup as in RTSP (where
we first need to open the local ports and pass them to the peer,
and only then receive the remote peer port numbers), we didn't
check the same url parameter as in the normal open routine.
Signed-off-by: Martin Storsjö <martin@martin.st>
I doubt that anyone ever would try to send a 1 byte packet
via the RTP protocol, but check just in case - it shouldn't
crash at least.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the chained demuxer (or more precisely, the lavf
utility code) to better fill in timestamps on packets from
these, especially for cases where one stream is a raw ADTS
stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add support for domain names, for multiple source addresses,
for exclusions, and for session level specification of addresses.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows us to explicitly fail if the caller tried to set
both inclusions and exclusions at the same time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously this only allowed literal IP addresses. When these
are conveyed in a SDP file as in RFC4570, host names are allowed
as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also parse segment durations as floating point, which is allowed
since HLS version 3.
This is based on a patch by Zhang Rui.
Signed-off-by: Martin Storsjö <martin@martin.st>
When first_timestamp was stored as-is, its actual time base
wasn't known later in the seek function.
Additionally, the logic (from 795d9594cf) for scaling it
based on stream_index is flawed - stream_index in the seek
function only specifies which stream the seek timestamp refers
to, but obviously doesn't say anything about which stream
first_timestamp belongs to.
In the cases where stream_index was >= 0 and all streams had the
same time base, this didn't matter in practice.
Seeking taking first_timestamp into account is problematic
when one variant is mpegts (with real timestamps) and one variant
is raw ADTS (with timestamps only being accumulated packet
duration), where the variants start at totally different timestamps.
Signed-off-by: Martin Storsjö <martin@martin.st>
Without the information, an application may choose audio from one
variant and video from another variant, which leads to fetching two
variants from the network. This enables av_find_best_stream() to find
matching audio and video streams, so that only one variant is fetched.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also adjust the streams timestamps according to their start
timestamp when comparing. This helps getting correctly interleaved
packets if one stream lacks timestamps (such as a plain ADTS
stream when the other variants are full mpegts) when the others
have timestamps that don't start from zero.
This probably doesn't work properly if such a stream is
temporarily disabled (via the discard flags) and then reenabled,
and such streams are hard to correctly sync against the other
streams as well - but this works better than before at least.
The segment number restriction makes sure all variants advance
roughly at the same pace as well.
Signed-off-by: Martin Storsjö <martin@martin.st>
If passing the end of one segment while initializing the
chained demuxer, the parent demuxer's streams aren't set up
yet, so we can't recheck the discard flags.
Signed-off-by: Martin Storsjö <martin@martin.st>
This serves as a safeguard; normally we want to use the dts
comparison to interleave packets from all active variants. If that
dts comparison for some reason doesn't work as intended, make sure
that all packets in all variants for a certain sequence number have
been returned before moving on to the next one.
Signed-off-by: Martin Storsjö <martin@martin.st>
Incomplete crypted files would lead to a read after buffer boundary
otherwise.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
Derived from VLC's http module.
Original authors:
Antoine Cellerier <dionoea@videolan.org>
Sébastien Escudier <sebastien-devel@celeos.eu>
Rémi Duraffort <ivoire@videolan.org>
Rémi Denis-Courmont <remi@remlab.net>
Francois Cartegnie <fcvlcdev@free.fr>
Normally, http servers shouldn't send this to us since we
don't advertise it with an Accept-Encoding header, but some
servers still do it anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure that values that are left-shifted by this constant
end up casted to 64 bit before shifting, avoiding overflow if the
value ends up larger than 2 GB.
Signed-off-by: Martin Storsjö <martin@martin.st>
This supports inclusion of one single IP address for now,
at the media level. Specifying the filter at the session level
(instead of at the media level), multiple source addresses,
exclusion, or using FQDNs instead of plain IP addresses is not
supported (yet at least).
Signed-off-by: Martin Storsjö <martin@martin.st>
If another peer is sending unicast packets to the same port that
we are listening on, those packets can end up being received despite
using source specific multicast. For those cases, manually check the
source address of received packets against the intended source address.
This only handles the case when the source list is one single IP
address for now, which probably is the most common case.
Based on a patch by Ed Torbett.
Signed-off-by: Martin Storsjö <martin@martin.st>
Blocking/exclusion is not supported yet.
The rtp protocol parameter takes the same form as the existing
sources parameter for the udp protocol.
Signed-off-by: Martin Storsjö <martin@martin.st>
If either of the deltas is too large for the multiplications to
succeed, don't use this for setting the avg frame rate.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The time scale is set in mdhd, and later validated in the
enclosing trak atom once all of its children have been parsed.
A loose mdhd atom outside of a trak atom could update the time
scale of the last stream without any validation.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The RTP timestamps can be decreasing for codecs with B-frames. For
these cases, make sure the timestamps in the MP4 file track itself
are nondecreasing, and add an offset to the RTP packet hint instead
to produce the intended RTP timestamp.
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes crashes when playing back certain RealRTSP streams.
When invoked from the RTP depacketizer, the full realmedia
demuxer isn't invoked, but only certain functions from it, where
a separate AVIOContext is passed in as parameter (for the buffer
containing the data to parse). The functions called from within
those entry points should only be using that parameter, not
s->pb. In the depacketizer case, s is the RTSP context, where ->pb
is null.
Cc: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes sure the ffurl_read_complete function actually
returns the number of bytes read, as the documentation of the
function says, even if the underlying protocol uses AVERROR_EOF
instead of 0.
Signed-off-by: Martin Storsjö <martin@martin.st>
A sid 0 would be mismatched to the attachment.
Prevent NULL pointer dereference.
Reported-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This allows handling matroska files with errors.
Fixes test4.mkv and test7.mkv from the official Matroska test suite,
and by extension Bugzilla #62.
Based on a patch by Reimar Doffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The previous allocation increment of 16384 meant that the cluster
array was allocated for 0.6 MB initially, which is a bit excessive
for cases with fragmentation where only a fraction of that ever
actually is used.
Therefore, start off at a much smaller value, and increase by
doubling (to avoid reallocating too often when writing long
non-fragmented mp4 files).
Bug-Id: 525
Signed-off-by: Martin Storsjö <martin@martin.st>
When writing fragmented mp4, the cluster array is reset when a
fragment is written. Instead of starting off reallocating the
array only based on the number of current elements in it, keep
track of how many elements there were allocated earlier.
This avoids reallocating this array needlessly when writing
fragmented mp4 files.
Bug-Id: 525
Signed-off-by: Martin Storsjö <martin@martin.st>
This makes the struct name (which isn't used anywhere) match the
name of the typedef, as for all the other structs declared in this
header.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently the demuxer shaves the blocks and exports only the
information that is useful to the decoder.
Exporting the blocks just as they are stored is simpler to understand
and will make remuxing wavpack easier.
Some fixes provided by Paul B Mahol <onemda@gmail.com>
and Michael Niedermayer <michaelni@gmx.at> and me.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
