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>
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>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPT protocol implementation uses rtmphttp:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can happen if doing a new request using the same socket,
but the new request failed, which clears the urlcontext.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add spaces around operators, fix brace placement and whitespace to
match K&R style, vertically align code, remove redundant != 0 and
convert x == 0 into !x, drop useless braces.
Signed-off-by: Martin Storsjö <martin@martin.st>
This can easily happen when the caller is using a custom AVIOContext.
Behave as if the filename was an empty string in this case.
CC: libav-stable@libav.org
This makes sure all incoming packets are read and handled (and reacted
to) while sending an FLV stream over RTMP to a server. If there were
enough incoming data to fill the TCP buffers, this could potentially
make things block at unexpected places. For the upcoming RTMPT support,
we need to consume all incoming data before we can send the next
request.
Signed-off-by: Martin Storsjö <martin@martin.st>
While there is no reason for starting a frame with anything else
than a Mode A packet, some senders seem to consistently use Mode B
packets for everything. This fixes depacketization of such streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
This factorizes existing code into a new function gen_buffer_time(),
which generates the client buffer time message and sends it to the
server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow using connection parameters in order to append arbitrary
AMF data like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0" to the
Connect message. You can pass these parameters through the -rtmp_conn
option.
Signed-off-by: Martin Storsjö <martin@martin.st>
This adds the avoption mpegts_flags and converts the existing
resend_headers option into a flag, keeping the old option as
fallback for now.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't required any longer, when the mpegts muxer uses it
as a proper chained muxer.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This removes the dependency on adts.c internals, and simplifies
adding other packetization formats.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This fixes issues with opening http urls that have authentication
or redirects, introduced in commit e999b641.
Signed-off-by: Martin Storsjö <martin@martin.st>
avisynth is a non-unicode application and cannot accept UTF-8
characters. Therefore, the input filename should be converted to
the correct code page that it expects.
Signed-off-by: Martin Storsjö <martin@martin.st>
Introduce ff_http_do_new_request(), a new function which sends a new
HTTP request, reusing the existing connection to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new AVOption 'multiple_requests', which indicates if we want
to use persistent connections (ie. Connection: keep-alive).
Signed-off-by: Martin Storsjö <martin@martin.st>
Some systems abuse the static payload types 35 or 36 (which
according to IANA are unassigned) for H264.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an URLContext is passed in, its ownership is given to this
function, and is either owned by the returned AVFormatContext
on a successful return, or freed on failure.
Signed-off-by: Martin Storsjö <martin@martin.st>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
tcp_shutdown() isn't needed at the moment, but is added for
consistency to explain how the function is supposed to be used.
Signed-off-by: Martin Storsjö <martin@martin.st>
If using the new -rtmp_app and -rtmp_playpath parameters,
one can in many cases set the main url to just rtmp://server/.
If the trailing slash is omitted, path is a string of zero length,
and using path+1 will end up reading uninitialized data.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Add/fix spacing, split long lines, align assignments where suitable.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Split long comments, move long comments at the end of lines to
separate lines above, fix vertical alignment, fix up comment style
(unify trailing dots - comments had a mix of 2, 3 or 4 dots, where
it would be just as good without them at all).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is worth keeping instead of removing, in case reading this
bit becomes necessary at some later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
Skip to parse fields for additional independent substreams and its
associated dependent substreams since libavcodec's E-AC-3 decoder does not
support them yet.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This fixes crashes, where the demuxer could return 0 even
if the returned AVPacket isn't initialized at all. This
could happen if running into EOF or running out of probesize
with non-seekable sources.
Signed-off-by: Martin Storsjö <martin@martin.st>
Mashiat Sarker Shakkhar
Mans Rullgard
Ronald S. Bultje
Kostya Shishkov
Martin Storsjö
Samuel Pitoiset
Jordi Ortiz
Anton Khirnov
Janne Grunau
Alex Converse
Jindřich Makovička
Luca Barbato
Oka Motofumi
Dave Yeo
Diego Biurrun
James Zern
Sean McGovern
Ivan Kovtunov
Justin Ruggles
Joakim Plate
Yusuke Nakamura
Hendrik Leppkes