This heaader is required for close() for sockets in network
code. For winsock, the equivalent function is defined in the
winsock2.h header.
This avoids having the HAVE_UNISTD_H in all files dealing with
raw sockets.
Signed-off-by: Martin Storsjö <martin@martin.st>
On MSVC, gmtime returns NULL for values outside of their supported
range (and these show up in our fate test). This doesn't seem
to affect the actual fate test result.
Signed-off-by: Martin Storsjö <martin@martin.st>
Currently if a pattern is given we search for up to the fifth file name in
that sequence. This option sets that limit to an arbitrary number.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Currently if a pattern is given we look for up to the fifth file name in
the sequence. This option sets that limit to an arbitrary number.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This adds the capability to start counting file number from an arbitrary
integer.
This includes a few lines of trivial code from FFmpeg codebase.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This adds the capability to start counting file number from an arbitrary
integer instead of always starting at 1.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
To ensure the full range of values is still used, also adjust all uses of this function to loop from 0
instead of 1. This way only 60.00 is added and nothing lost.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This muxer supports CODEC_ID_SRT with the timestamps in the packet data
and CODEC_ID_TEXT with the timestamps in the packet fields.
Makes -scodec copy work from Matroska.
Also use ff_neterrno() instead of errno directly (which doesn't work
on windows), for getting the error code.
Signed-off-by: Martin Storsjö <martin@martin.st>
getnameinfo doesn't set errno on failure, it returns an error code,
which should be handled by gai_strerror instead of the normal
strerror.
Signed-off-by: Martin Storsjö <martin@martin.st>
Rtmpt is effectively half duplex - the server can't return any
data unless we send a request (to which the server responds). If
we don't have any data to send currently, and the server didn't
return any data either, wait a little before doing the next request.
This avoids busy looping with idle posts with empty replies, while
waiting for more data from the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.
This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.
Signed-off-by: Martin Storsjö <martin@martin.st>
This seems to be the correct mode to send, according to the
original RTSP RFC, and matches the method RECORD which is
sent later when starting to send data.
Darwin Streaming Server works fine with either of them.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires all NAL units to fit within single RTP packets. It
doesn't change the actual packetization for packets that fit, but
errors out and gives a helpful hint if the NAL units would have to
be split, and signals the right packetization mode in the SDP.
Signed-off-by: Martin Storsjö <martin@martin.st>