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Author SHA1 Message Date
Jan Schmidt 27736d406e rtsp-stream: Implement UDP buffer size setting. 
Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the
UDP TX buffer size.

Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
 2015-09-03 22:19:40 +10:00
Hyunjun Ko 2a3dd3d38f rtsp-stream: add description for gst_rtsp_stream_request_aux_sender 
https://bugzilla.gnome.org/show_bug.cgi?id=750764
 2015-06-11 11:37:03 +01:00
Hyunjun Ko 4ff22ef6d2 rtsp-stream: get valid clock-rate from last-sample 
clock-rate in last-sample's caps is integer, not unsigned.
To get this value properly, variable needs to be type-casted to int.

https://bugzilla.gnome.org/show_bug.cgi?id=747614
 2015-04-27 12:41:59 +02:00
Hyunjun Ko de590b4b2a rtsp-stream: fix to get valid each stream data for request-aux-sender signal 
Because of duplicated g_signal_connect for request-aux-sender signal,
wrong stream pointer is passed to the signal handler.
Instead of passing each stream, pass stream array and get the relevant stream.

https://bugzilla.gnome.org/show_bug.cgi?id=747839
 2015-04-15 15:14:04 +02:00
Sebastian Dröge ef3bfd757b rtsp-stream: Limit the queues to 1 buffer 
We only need them to be able to pre-roll, queueing up more data here
is only going to harm latency and memory usage.
 2015-03-23 21:04:43 +01:00
Sebastian Dröge 357af7aea6 rtsp-stream: Update comment and ASCII art to the latest code 
We have a queue in front of the udpsink too to prevent the pipeline from
locking up.
 2015-03-23 20:59:52 +01:00
Nicolas Dufresne dfb053add3 rtsp-media: Properly return first rtptime 
Instead we where returning first GstBuffer timestamp. This would result
in clock skew and unwanted behaviour in RTSP playback.

https://bugzilla.gnome.org/show_bug.cgi?id=746479
 2015-03-21 11:04:05 -04:00
Nicolas Dufresne 01562286ba rtsp-stream: Don't leave buffer mapped 
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
 2015-03-18 16:44:19 -04:00
Sebastian Dröge 852cc09f54 rtsp-stream: Get the seqnum-base and other information from the last buffer in the sink 
This gives more accurate values than asking the payloader. There might be
queueing happening between the payloader and the sink.

https://bugzilla.gnome.org/show_bug.cgi?id=745704
 2015-03-09 16:00:38 +01:00
Andreas Frisch bac59c52f1 rtsp-stream: Add necessary queues between tee and multiudpsink 
https://bugzilla.gnome.org/show_bug.cgi?id=744379
 2015-02-13 11:28:43 +02:00
Sebastian Dröge 844add610d rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer 2015-02-06 09:42:50 +01:00
Sebastian Dröge ccf6c6eb53 Add initial support for RECORD 
We currently only support media that is RECORD or PLAY only, not both at once.

https://bugzilla.gnome.org/show_bug.cgi?id=743175
 2015-02-06 09:42:42 +01:00
Anila Balavan 18668bf495 rtsp-stream: RTCP and RTP transport cache cookies seperated 
RTCP packets were not sent because the same tr_cache_cookie was used for
both RTP and RTCP. So only one of the tr_cache lists were populated
depending on which one was sent first. If the tr_cache list is not
populated then no packets can be sent. Most often this happened to be
RTCP. Now seperate RTCP and RTP transport cache cookies are added which
resulted in both the tr_cache_lists to be populated regardless of which
one was sent first.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=743734
 2015-01-30 18:26:44 +01:00
Tim-Philipp Müller 6987a00fa9 rtsp-stream: fix false compiler warning 
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function
 2015-01-21 14:58:19 +00:00
Göran Jönsson 0d2de69db9 rtsp-stream: Have one copy of the transports cache for RTP and RTCP each 
Fixes crash when two threads access handle_new_sample() at the same
time, one for RTP, one for RTCP.

Otherwise, when iterating over the transports cache, it might be modified by
another thread at the same time if the transports cookie has changed.

https://bugzilla.gnome.org/show_bug.cgi?id=742954
 2015-01-16 12:52:43 +01:00
Sebastian Dröge fe8e877dd9 rtsp-stream: Set format=TIME on our app sources for TCP 2015-01-15 19:35:01 +01:00
Sebastian Dröge a44b564f59 rtsp-stream: Fix some minor memory leaks 2014-12-16 16:46:15 +01:00
Sebastian Dröge 06bfc0697b rtsp-stream: Fix compiler warnings 
rtsp-stream.c:1351:3: error: non-void function 'gst_rtsp_stream_get_retransmission_time' should return a value [-Wreturn-type]
  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
  ^

rtsp-stream.c:1384:3: error: non-void function 'gst_rtsp_stream_get_retransmission_pt' should return a value [-Wreturn-type]
  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
  ^
 2014-12-16 16:42:13 +01:00
Matthew Waters 4f40781fff media: implement ssrc-multiplexed retransmission support 
based off RFC 4588 and the server-rtpaux example in -good
 2014-12-16 16:41:08 +01:00
Göran Jönsson 058698c9cf rtsp: Ref transports in hash table. 
Also ref streams for transports.
This solves a crash when reciving a rtcp after teardown but before
client finalize.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
 2014-12-02 16:29:24 +01:00
Aleix Conchillo Flaqué 7c267928ff rtsp-stream: unref srtp decoder when leaving bin 
https://bugzilla.gnome.org/show_bug.cgi?id=739481
 2014-11-01 11:26:14 +00:00
Aleix Conchillo Flaqué 966065a018 stream: release lock even not all transports have been removed 
We don't want to keep the lock even we return FALSE because not all the
transports have been removed. This could lead into a deadlock.

https://bugzilla.gnome.org/show_bug.cgi?id=737797
 2014-10-21 10:08:44 +02:00
Srimanta Panda 376488d8c7 rtsp-media: Make sure that sequence numbers are monotonic after pause 
The sequence number is not monotonic for RTP packets after pause. The
reason is basepayloader generates a randon sequence number when the
pipeline goes from ready to pause. With this fix generation of sequence
number will be monotonic when going from pause to play request.

https://bugzilla.gnome.org/show_bug.cgi?id=736017
 2014-09-12 17:29:30 +03:00
Sebastian Dröge 1b47b6d9b0 rtsp-stream: Remove the multicast group udp sources when removing from the bin 2014-08-25 10:39:04 +03:00
Sebastian Dröge 6ba5ca447f rtsp-media: Query position and stop time only on the RTP parts of the pipeline 
The RTCP parts, in specific the RTCP udpsinks, are not flushed when
seeking and will always continue counting the time. This leads to
the NPT after a backwards seek to be something completely different
to the actual seek position.

https://bugzilla.gnome.org/show_bug.cgi?id=732644
 2014-08-12 10:54:12 +03:00
Sebastian Dröge 3159b374b9 rtsp-stream: Listen on the multicast group for RTP/RTCP packets 
When a UDP multicast transport is used it is expected that the server listens
for RTP and RTCP packets on the multicast group with the corresponding port.
Without this we will never get RTCP packets from clients in multicast mode.

https://bugzilla.gnome.org/show_bug.cgi?id=732238
 2014-07-22 14:26:49 +02:00
Wim Taymans 945c93fde0 filter: Release lock in filter functions 
Release the object lock before calling the filter functions. We need to
keep a cookie to detect when the list changed during the filter
callback. We also keep a hashtable to make sure we only call the filter
function once for each object in case of concurrent modification.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950
 2014-07-10 11:36:55 +02:00
Wim Taymans db95746f6b stream: crypto can be NULL 2014-06-27 16:55:07 +02:00
Evan Nemerson 34e6ac3b9f introspection: add (nullable) annotations to return values 
https://bugzilla.gnome.org/show_bug.cgi?id=730952
 2014-06-26 19:08:16 +02:00
Evan Nemerson d08b46f4b7 gi: improve annotations 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953
 2014-06-24 09:48:45 +02:00
Aleix Conchillo Flaqué 17322693f6 stream: add signals for new RTP/RTCP encoders 
New signals to allow the user to configure the dynamically created
encoders.

https://bugzilla.gnome.org/show_bug.cgi?id=730228
 2014-05-16 16:27:52 +02:00
Wim Taymans 377ca6ed0f stream: add method to set crypto info 
Make a method to configure the crypto information of a stream.
Set udpsrc in READY instead of PAUSED so that we can configure caps
later.
 2014-04-03 17:26:12 +02:00
Wim Taymans 3d6175c745 stream: add SRTP support 
Install srtp encoder and decoder elements in rtpbin
Add MIKEY in SDP
 2014-03-25 10:31:21 +01:00
Sebastian Rasmussen b1b5301577 gobject-introspection: Add annotations to support language bindings 
In addition a few cosmetic changes:

 * Adjust the order of arguments
 * Fix typo: occured -> occurred
 * Fix indentation after Return:-clauses

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
 2014-03-24 00:36:42 +00:00
Sebastian Rasmussen 0b617dd5bd rtsp-stream: Don't mix IPv4 and IPv6 addresses 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362
 2014-03-15 15:44:25 +01:00
Wim Taymans 2c7ffe97ca stream: take caps after the session manager 
Take the caps for the SDP after they leave the rtpbin so that we can
also get the properties added by rtpbin elements.
 2014-03-13 14:27:15 +01:00
Wim Taymans 50ca10e751 stream: release lock while pushing out packets 
Keep a cache of the transports and use this to iterate the transport
while pushing packets. This allows us to release the lock early.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
 2014-03-13 14:20:17 +01:00
Wim Taymans 48b6b8e3e6 stream: release some locks in error cases 2014-03-03 12:17:48 +01:00
Sebastian Rasmussen 81a2928c89 docs: Enable and fix gtk-doc warnings 
* Makefile: Enable gtk-doc warnings, like the rest of GStreamer
 * addresspool/mediafactory: Add missing annotation colon
 * stream: Annotate return value

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528
 2014-03-03 09:43:05 +01:00
Göran Jönsson a7f0feff23 stream: set ttl-mc before adding the socket 
Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
never be set on socket.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
 2014-02-18 11:10:51 +01:00
Sebastian Dröge 957a4a65c6 rtsp-server: Fix lots of compiler warnings with clang 2014-02-09 10:45:28 +01:00
Sebastian Dröge 902b59f823 Revert "rtsp-server: support build against last stable release" 
This reverts commit 099a10f61f.

Let us require 1.2.3 now, which is going to be released in a few
minutes.
 2014-02-09 10:19:50 +01:00
Wim Taymans 9048d87ff4 stream: handle NULL seqnum and rtptime arguments 2014-02-04 16:28:00 +01:00
Wim Taymans 71c45fce5a stream: add fallback for missing stats property 
Use a fallback when the payloader does not have a stats property

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
 2014-02-04 10:14:45 +01:00
Wim Taymans 036f2760bf stream: don't leak stats structure 
Don't leak the stats structure and deal with NULL stats.
 2014-01-28 14:51:26 +01:00
Sebastian Rasmussen 7edaa6ca20 stream: Get rtpinfo properties atomically from payloader 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
 2014-01-27 15:19:30 +01:00
Tim-Philipp Müller 099a10f61f rtsp-server: support build against last stable release 
Until 1.2.3 is out with the new get_type function and we
can require that.
 2014-01-12 16:55:21 +00:00
Wim Taymans d90ce618e4 stream: fix compilation 2014-01-07 15:28:05 +01:00
Wim Taymans ae1fe21436 stream: add property to configure profiles 2014-01-07 12:39:58 +01:00
Wim Taymans a1202effda stream: add method to check supported transport 
Add a method to check if a transport is supported
 2014-01-07 12:39:57 +01:00