Since d607861, service can not be NULL, only "0".
An UDP address with neither local port nor address leaves both
service and node to their default value, and POSIX specifies
that they are not allowed to be both NULL; "0" is equivalent
to an unspecified port for all currently known protocols.
Fix CID 1341570.
Fixes Ticket5467 "Lossless j2k information no longer shown"
Based on suggestion by Hendrik Leppkes
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Missing docs found by: nevcairiel
RFC: should we add support so that the C field names always work as av option names/keys ?
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The second one is not explicitly needed, as res is not reset, but it is there
for consistency.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
This fixes partially completed send()
Avoids holding the mutex during send()
fixes race conditions in error handling
removes copied non thread specific blocking code
Fixes deadlocks on closure
Fixes data loss on closure
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit enables sending UDP packets in a background thread with specified delay.
When sending packets without a delay some devices with small RX buffer
( MAG200 STB, for example) will drop tail packets in bursts causing
decoding errors.
To use it specify "fifo_size" with "packet_gap" .
The output url will looks like udp://xxx:yyy?fifo_size=<output fifo
size>&packet_gap=<delay in usecs>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Previously, we required the minimum number of bytes required for
the full box. Don't strictly require the astronomical body and additional
notes fields, but do require an altitude field (which currently isn't
parsed). This matches the initial length check at the start of the function
(which doesn't know about the variable length place field).
Signed-off-by: Martin Storsjö <martin@martin.st>
This was missed in e1eb0fc960, when ff_interleaved_peek was
changed to include const during the evolution of the patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
As long as caller only writes packets using av_interleaved_write_frame
with no manual flushing, this should allow us to always have accurate
durations at the end of fragments, since there should be at least
one queued packet in each stream (except for the stream where the
current packet is being written, but if the muxer itself does the
cutting of fragments, it also has info about the next packet for that
stream).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows callers with avio write callbacks to get the bytestream
positions that correspond to keyframes, suitable for live streaming.
In the simplest form, a caller could expect that a header is written
to the bytestream during the avformat_write_header, and the data
output to the avio context during e.g. av_write_frame corresponds
exactly to the current packet passed in.
When combined with av_interleaved_write_frame, and with muxers that
do buffering (most muxers that do some sort of fragmenting or
clustering), the mapping from input data to bytestream positions
is nontrivial.
This allows callers to get directly information about what part
of the bytestream is what, without having to resort to assumptions
about the muxer behaviour.
One keyframe/fragment/block can still be split into multiple (if
they are larger than the aviocontext buffer), which would call
the callback with e.g. AVIO_DATA_MARKER_SYNC_POINT, followed by
AVIO_DATA_MARKER_UNKNOWN for the second time it is called with
the following data.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it to get stream duration, sample rate, channel count and initial padding
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Previously a partial log message without newline was printed in case of
loglevel=warning.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '0abb07bad7026a945a31ba4047e6583c8b3fa3da':
movenc: Update a comment to reflect how the code actually behaves
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '06edef3d5e072ef3c4face9ce946d2d9c36cc477':
Generate the lists of enabled protocols/bsfs from configure.
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'd12b5b2f135aade4099f4b26b0fe678656158c13':
build: Split test programs off into separate files
Some conversions done by: James Almer <jamrial@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
It doesn't matter what the actual reason for not returning
an AVPacket was - if we didn't return any packet and we have
the next one queued, parse it immediately. (rtp_parse_queued_packet
always consumes a queued packet if one exists, so there's no risk
for infinite loops.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Leaking this private structure opens up the possibility that it may
be re-used when parsing later packets in the stream. This is
problematic if the later packets are not the same codec type (e.g.
private allocated during Vorbis parsing, but later packets are Opus
and the private is assumed to be the oggopus_private type in
opus_header()).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Larger values would imply file durations of astronomic proportions and cause
overflows
Fixes integer overflow
Fixes: usan_int64_overflow
Found-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'e3453fd44480d903338c663238bf280215dd9a07':
matroska: Write the field order information
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Replace av_copy_packet and deprecated av_dup_packet by
creating reference using av_packet_ref.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
TeeSlave.bsfs is array of pointers to AVBitStreamFilterContext,
so element size should be really size of a pointer, not size
of TeeSlave structure.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
The declarations that this comment referred to were removed
in 2439f2ca8 - there is no unbuffered IO in this header now.
Signed-off-by: Martin Storsjö <martin@martin.st>
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adds per slave option 'onfail' to the tee muxer allowing an output to
fail, so other slave outputs can continue.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
In open_slave failure can happen before bsfs array is initialized,
close_slave must check that bsfs is not NULL before accessing
tee_slave->bsfs[i] element.
Slave muxer expects write_trailer to be called if it's
write_header suceeded (so resources allocated in write_header
are freed). Therefore if failure happens after successfull
write_header call, we must ensure that write_trailer of
that particular slave is called.
Some cleanups are made by Marton Balint.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '3ee2ec5ec1e39a438f89302d949c93a1b5d365a2':
unix: Use rw_timeout for setting the connect timeout
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This codepath isn't quite as bad as it used to sound, if fragments
are cut automatically at video packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '933dec0e29ec4d2cb83474279a6c52d62fdb7310':
file: Add an option for following a file that is being written
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'd44f3e4059506a182f59218b1e967d42b01e097c':
avio: Apply avoptions on the URLContext itself as well
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '65a802401c6cc136576bb2e613c0577cbf622aa8':
build: Add component for the SRTP common code
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
