David Svensson Fors [Thu, 22 Oct 2015 07:15:21 +0000 (09:15 +0200)]
rtsp-stream: Always unref return value of gst_object_get_parent()
Fixes a leak of a GstBin in the udp-mcast case.
https://bugzilla.gnome.org/show_bug.cgi?id=756968
Tim-Philipp Müller [Tue, 29 Sep 2015 12:04:53 +0000 (13:04 +0100)]
common: update for new suppression
Makes check-valgrind pass with glib 2.46
Sebastian Rasmussen [Mon, 28 Sep 2015 15:40:59 +0000 (17:40 +0200)]
rtsp-media: Take reference to media that will be prepared
default_prepare() takes a transfer-none reference GstRTSPMedia object.
Later on a g_idle_source_new() is created and a pointer to the media
object is passed as user data. If the media is freed before the idle
source is dispatched the media object pointer is invalid, but the idle
source callback expects it to still be valid. To fix this a reference to
the media object is taken when registering the source callback function
and a corresponding release of the reference is done when the souce is
destroyed.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
Sebastian Dröge [Fri, 25 Sep 2015 21:32:52 +0000 (23:32 +0200)]
Release 1.6.0
Sebastian Dröge [Fri, 18 Sep 2015 18:12:06 +0000 (20:12 +0200)]
Release 1.5.91
Tim-Philipp Müller [Thu, 17 Sep 2015 19:07:34 +0000 (20:07 +0100)]
stream: fix docs for recently-added get/set_buffer_size API
https://bugzilla.gnome.org/show_bug.cgi?id=749095
Jan Schmidt [Fri, 4 Sep 2015 01:23:43 +0000 (11:23 +1000)]
rtsp-media: Don't crash on encrypted RTX SDP
In parse_keymgmt(), don't mutate the input string that's been passed
as const, especially since we might need the original value again if
the same key info applies to multiple streams (RTX, for example).
https://bugzilla.gnome.org/show_bug.cgi?id=754753
Jan Schmidt [Sat, 22 Aug 2015 10:59:40 +0000 (20:59 +1000)]
test-mp4: Support filenames with spaces in them. Error out on too few arguments
Jan Schmidt [Sun, 16 Aug 2015 16:36:31 +0000 (02:36 +1000)]
test-record: Check parameter count and print out help
If no launch pipeline was supplied, print out some help
Jan Schmidt [Mon, 31 Aug 2015 12:48:34 +0000 (22:48 +1000)]
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
Jan Schmidt [Mon, 31 Aug 2015 12:47:45 +0000 (22:47 +1000)]
rtsp-media: Fix small typo causing gtk-doc to complain
Sebastian Dröge [Wed, 19 Aug 2015 11:15:23 +0000 (14:15 +0300)]
Release 1.5.90
Hyunjun Ko [Wed, 12 Aug 2015 05:33:44 +0000 (14:33 +0900)]
media-factory: get port number through gst_rtsp_url_get_port
https://bugzilla.gnome.org/show_bug.cgi?id=753473
Francisco Velazquez [Thu, 13 Aug 2015 09:24:10 +0000 (11:24 +0200)]
media-test: Removing unnecessary assertion
https://bugzilla.gnome.org/show_bug.cgi?id=753385
Xavier Claessens [Thu, 23 Jul 2015 18:50:30 +0000 (14:50 -0400)]
Document that source keeps a ref on server until it's destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=749227
Nicolas Dufresne [Sat, 8 Aug 2015 15:09:57 +0000 (11:09 -0400)]
media-test: Test for multiple dynamic payload
https://bugzilla.gnome.org/show_bug.cgi?id=753385
Nicolas Dufresne [Sat, 8 Aug 2015 13:40:09 +0000 (09:40 -0400)]
media: Only add fakesink once per pipeline
The intention is to prevent going PLAYING state before pads are created.
If there was mutilple dynamic payload, it would leak few fakesink and
actually prevent from ever reaching playing state.
https://bugzilla.gnome.org/show_bug.cgi?id=753385
Nicolas Dufresne [Sat, 8 Aug 2015 13:08:37 +0000 (09:08 -0400)]
Revert "rtsp-media: Only add 1 fakesink per pipeline"
This reverts commit
22bf61f16c1210bb458fc3f53642179a0211104f.
Nicolas Dufresne [Fri, 7 Aug 2015 13:21:36 +0000 (09:21 -0400)]
rtsp-media: Only add 1 fakesink per pipeline
There should be only one fakesink per pipeline, not per dynpay. This
would lead to element naming clash.
Vineeth TM [Thu, 30 Jul 2015 06:32:43 +0000 (15:32 +0900)]
rtsp-media: assertion error due to wrong condition check
In media to caps function, reserved_keys array is being used for variable i,
leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
changed it to variable j
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Sebastian Dröge [Wed, 29 Jul 2015 10:27:05 +0000 (11:27 +0100)]
rtsp-media: Strip keys from the fmtp that we use internally in our caps
Skip keys from the fmtp, which we already use ourselves for the
caps. Some software is adding random things like clock-rate into
the fmtp, and we would otherwise here set a string-typed clock-rate
in the caps... and thus fail to create valid RTP caps
https://bugzilla.gnome.org/show_bug.cgi?id=753009
Xavier Claessens [Mon, 20 Jul 2015 20:37:44 +0000 (16:37 -0400)]
threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
https://bugzilla.gnome.org/show_bug.cgi?id=752640
Stefan Sauer [Fri, 3 Jul 2015 20:00:00 +0000 (22:00 +0200)]
Automatic update of common submodule
From f74b2df to 9aed1d7
Sebastian Dröge [Wed, 24 Jun 2015 22:04:28 +0000 (00:04 +0200)]
Back to development
Sebastian Dröge [Wed, 24 Jun 2015 21:44:37 +0000 (23:44 +0200)]
Release 1.5.2
Ognyan Tonchev [Thu, 18 Jun 2015 11:12:04 +0000 (13:12 +0200)]
rtsp-client: allow application to decide what requirements are supported
Add "check-requirements" signal and vfunc to allow application
(and subclasses) to check the requirements.
Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=749417
Nicolas Dufresne [Tue, 16 Jun 2015 21:50:26 +0000 (17:50 -0400)]
Automatic update of common submodule
From 6015d26 to f74b2df
Ognyan Tonchev [Thu, 11 Jun 2015 15:39:00 +0000 (17:39 +0200)]
rtsp-media: Always use real payloader when creating streams
A bin that contains the real payloader might be used as payloader. In this
case we have to get the real payloader for the various properties it provides.
Example use cases for this are bins that payload some media and then have
additional elements that add metadata or RTP extension headers to the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=750800
Sebastian Dröge [Sat, 13 Jun 2015 15:14:43 +0000 (17:14 +0200)]
test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
Sebastian Dröge [Fri, 12 Jun 2015 21:35:32 +0000 (23:35 +0200)]
test-netclock: Use new ntp-time-source property on rtpbin
Select the clock time to be used as NTP time source. This allows proper
synchronization between receivers, independent of sharing base times, and just
requires them to use the same clock.
Sebastian Dröge [Thu, 11 Jun 2015 18:41:31 +0000 (20:41 +0200)]
test-netclock: Setting the same base time on sender and receiver is not necessary
It's going to be fixed up by rtpbin when using ntp-sync=TRUE
Hyunjun Ko [Thu, 11 Jun 2015 08:38:52 +0000 (17:38 +0900)]
rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
https://bugzilla.gnome.org/show_bug.cgi?id=750764
Hyunjun Ko [Thu, 11 Jun 2015 09:10:12 +0000 (18:10 +0900)]
docs: add missing types
https://bugzilla.gnome.org/show_bug.cgi?id=750764
Hyunjun Ko [Thu, 11 Jun 2015 08:37:25 +0000 (17:37 +0900)]
docs: add missing apis
https://bugzilla.gnome.org/show_bug.cgi?id=750764
Sebastian Dröge [Wed, 10 Jun 2015 15:14:18 +0000 (17:14 +0200)]
test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
Xavier Claessens [Sat, 6 Jun 2015 02:35:39 +0000 (22:35 -0400)]
GstRTSPAuth: Add client certificate authentication support
https://bugzilla.gnome.org/show_bug.cgi?id=750471
Sebastian Dröge [Tue, 9 Jun 2015 11:53:47 +0000 (13:53 +0200)]
test-netclock-client: Use new GstClock API to wait for clock synchronization
Sebastian Dröge [Tue, 9 Jun 2015 11:51:02 +0000 (13:51 +0200)]
test-netclock-client: Use a GMainLoop and playbin's source-setup signal
A mainloop is needed to get glimagesink to display something on OSX, and
the source-setup signal just makes things a little bit easier.
Edward Hervey [Tue, 9 Jun 2015 09:30:54 +0000 (11:30 +0200)]
Automatic update of common submodule
From d9a3353 to 6015d26
Stefan Sauer [Mon, 8 Jun 2015 21:08:34 +0000 (23:08 +0200)]
Automatic update of common submodule
From d37af32 to d9a3353
Stefan Sauer [Sun, 7 Jun 2015 21:07:31 +0000 (23:07 +0200)]
Automatic update of common submodule
From 21ba2e5 to d37af32
Stefan Sauer [Sun, 7 Jun 2015 15:32:29 +0000 (17:32 +0200)]
Automatic update of common submodule
From c408583 to 21ba2e5
Stefan Sauer [Sun, 7 Jun 2015 15:06:40 +0000 (17:06 +0200)]
docs: remove variables that we define in the snippet from common
This is syncing our Makefile.am with upstream gtkdoc.
Stefan Sauer [Sun, 7 Jun 2015 15:16:47 +0000 (17:16 +0200)]
Automatic update of common submodule
From 44a3517 to c408583
Sebastian Dröge [Sun, 7 Jun 2015 14:44:55 +0000 (16:44 +0200)]
Back to development
Sebastian Dröge [Sun, 7 Jun 2015 09:20:01 +0000 (11:20 +0200)]
Release 1.5.1
Göran Jönsson [Mon, 25 May 2015 14:36:18 +0000 (16:36 +0200)]
rtsp-client: No flush during Teardown.
When calling gst_rtsp_watch_write_data in gstrtspconnection.c and
backlog is empty it can happen that just a part of a message will be
sent and rest is in backlog queue. If then flush during teardown
just a part of message will be sent.This can lead to client miss
teardown response since it expect to get the last part of message.
The flushing during teardown was introduced to fix a deadlock that now
is fixed more generally in handle_request by temporary setting backlog
size to unlimited.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749845
Tim-Philipp Müller [Wed, 27 May 2015 16:04:41 +0000 (17:04 +0100)]
tests: Use AM_TESTS_ENVIRONMENT
Needed by the new automake test runner and the
current version of the common submodule.
Sebastian Dröge [Wed, 20 May 2015 14:05:47 +0000 (17:05 +0300)]
rtsp-server: Use single-include rtsp header to make sure we get all definitions
Sebastian Dröge [Tue, 5 May 2015 14:46:57 +0000 (16:46 +0200)]
rtsp-media: Mark some more functions static
Sebastian Dröge [Tue, 5 May 2015 14:46:19 +0000 (16:46 +0200)]
rtsp-media: Only unblock the media in suspend() when actually changing the state
Otherwise we're going to lose a few packets for live streams during DESCRIBE.
Sebastian Dröge [Mon, 4 May 2015 14:33:08 +0000 (16:33 +0200)]
examples: Use AVPF profile for the RTX example
Sebastian Dröge [Mon, 4 May 2015 14:31:20 +0000 (16:31 +0200)]
rtsp-sdp: Only add RTX to the SDP when using a feedback profile
Hyunjun Ko [Mon, 27 Apr 2015 10:35:53 +0000 (19:35 +0900)]
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
Tim-Philipp Müller [Sun, 26 Apr 2015 14:00:05 +0000 (15:00 +0100)]
autogen.sh: only run autopoint if gettext requested in configure.ac
Not just because there happens to be a po directory.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
Tim-Philipp Müller [Sun, 26 Apr 2015 13:58:49 +0000 (14:58 +0100)]
Revert "configure.ac: uncomment gettext version setup"
This reverts commit
1545d8fef7065081079172ec264a0061039ac075.
We don't need a gettext setup here and there's no po
directory either, so no reason why autopoint would be
run in the first place.
See https://bugzilla.gnome.org/show_bug.cgi?id=748058
Alistair Buxton [Thu, 23 Apr 2015 17:53:08 +0000 (18:53 +0100)]
Fix timeout function signatures across tests and examples
Tim-Philipp Müller [Thu, 23 Apr 2015 16:27:40 +0000 (17:27 +0100)]
tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
Make sure the test environment is set up.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
Tim-Philipp Müller [Thu, 23 Apr 2015 16:22:59 +0000 (17:22 +0100)]
configure: bump automake requirement to 1.14 and autoconf to 2.69
This is only required for builds from git, people can still
build tarballs if they only have older autotools.
https://bugzilla.gnome.org//show_bug.cgi?id=747624
Vincent Penquerc'h [Mon, 20 Apr 2015 07:49:57 +0000 (08:49 +0100)]
configure.ac: uncomment gettext version setup
Fixes autogen.sh. It would run autopoint, which would complain
that it could not find the gettext version in configure.ac.
https://bugzilla.gnome.org/show_bug.cgi?id=748058
Hyunjun Ko [Wed, 15 Apr 2015 01:06:30 +0000 (10:06 +0900)]
test-video-rtx: set exact payload type to PCMA payloader
Setting wrong payload type causes failure to do retransmission through audio stream
https://bugzilla.gnome.org/show_bug.cgi?id=747839
Hyunjun Ko [Wed, 15 Apr 2015 00:45:23 +0000 (09:45 +0900)]
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
Tim-Philipp Müller [Mon, 6 Apr 2015 09:32:52 +0000 (10:32 +0100)]
Update autogen.sh to latest version from common
Fixes build after aclocal_check etc. helpers have been removed.
Tim-Philipp Müller [Fri, 3 Apr 2015 17:58:26 +0000 (18:58 +0100)]
Automatic update of common submodule
From bc76a8b to c8fb372
Sebastian Dröge [Mon, 23 Mar 2015 20:03:20 +0000 (21:03 +0100)]
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.
Sebastian Dröge [Mon, 23 Mar 2015 19:59:52 +0000 (20:59 +0100)]
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.
Nicolas Dufresne [Sat, 21 Mar 2015 15:04:05 +0000 (11:04 -0400)]
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
Nicolas Dufresne [Wed, 18 Mar 2015 20:44:19 +0000 (16:44 -0400)]
rtsp-stream: Don't leave buffer mapped
If the seq is NULL, the RTP buffer was left mapped. We should always
unmap the buffer.
Sebastian Dröge [Sun, 15 Mar 2015 12:27:39 +0000 (12:27 +0000)]
Fix typo in README
Tim-Philipp Müller [Tue, 10 Mar 2015 09:39:22 +0000 (09:39 +0000)]
Fix double semicolons
Sebastian Dröge [Mon, 9 Mar 2015 15:00:07 +0000 (16:00 +0100)]
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
Sebastian Dröge [Mon, 9 Mar 2015 12:00:25 +0000 (13:00 +0100)]
rtsp-media: Don't seek for PLAY if the position will not change
https://bugzilla.gnome.org/show_bug.cgi?id=745704
Sebastian Dröge [Mon, 9 Mar 2015 09:21:49 +0000 (10:21 +0100)]
rtsp-media: Don't include payload type in the caps for framesize
When the sdp media attribute framesize are converted to caps
the <payload> should not be included.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
Based on the patch for rtspsrc by Linus Svensson <linussn@axis.com>
Linus Svensson [Wed, 26 Feb 2014 21:34:06 +0000 (22:34 +0100)]
rtsp-sdp: add payload type to the sdp framesize attribute
The sdp framesize attribute is desribed in RFC6064. It is specified
for payloading of H263 and has the following form
a=framesize:<payload type> <width>-<height>. The <width>-<height> part
should be added to the caps in a payloader and the <payload type> should
be added by the rtsp-server.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725334
Luis de Bethencourt [Tue, 3 Mar 2015 13:51:01 +0000 (13:51 +0000)]
examples: test-uri: fix tainted variable
Insignificant but this keeps Coverity happy.
CID #1268404
Jan Schmidt [Mon, 2 Mar 2015 14:49:42 +0000 (01:49 +1100)]
examples: Add a simple example of network synch for live streams.
An example server and client that works for synchronising live streams
only - as it can't support pause/play.
Jan Schmidt [Mon, 2 Mar 2015 14:49:42 +0000 (01:49 +1100)]
rtsp-media-factory: Add functions to set/get the media gtype
Allow specifying the GType of a GstRtspMedia subclass to create
as a simpler way to get the factory to create a custom
GstRtspMedia sub-class, without subclassing GstRtspMediaFactory.
Gregor Boirie [Fri, 27 Feb 2015 16:45:42 +0000 (17:45 +0100)]
rtsp-media: fix double unlock in _get_buffer_size()
Fixes an abort when calling gst_rtsp_media_get_buffer_size()
because of double g_mutex_unlock () usage.
https://bugzilla.gnome.org/show_bug.cgi?id=745434
Kent-Inge Ingesson [Thu, 19 Feb 2015 08:43:16 +0000 (10:43 +0200)]
rtsp-session: Use monotonic time for RTSP session timeout
Changed RTSP session timeout handling to monotonic time
and deprecating the API for current system time.
This fixes timeouts when the system time changes.
https://bugzilla.gnome.org/show_bug.cgi?id=743346
Sebastian Dröge [Fri, 13 Feb 2015 10:21:16 +0000 (12:21 +0200)]
rtsp-client: Only error out in PLAY if seeking actually failed
If the media was just not seekable, we continue from whatever position we are
and let the client decide if that is what is wanted or not.
Only if the actual seek failed, we can't really recover and should error out.
Andreas Frisch [Thu, 12 Feb 2015 09:46:28 +0000 (10:46 +0100)]
rtsp-stream: Add necessary queues between tee and multiudpsink
https://bugzilla.gnome.org/show_bug.cgi?id=744379
Sebastian Dröge [Thu, 12 Feb 2015 14:48:46 +0000 (16:48 +0200)]
rtsp-media: If seeking fails, don't wait forever for the media to preroll again
Instead error out properly the same way as if the SEEKING query already
failed.
Tim-Philipp Müller [Wed, 11 Feb 2015 17:24:38 +0000 (17:24 +0000)]
rtsp-stream: minor code formatting fix
Luis de Bethencourt [Tue, 10 Feb 2015 16:39:58 +0000 (16:39 +0000)]
rtsp-media: fix logic for collect_streams
Fix the logic of gst_rtsp_media_collect_streams() so after looping collecting
all streams it knows if it got any, and can check if the transport mode is OK.
CID #1268400
Sebastian Dröge [Mon, 9 Feb 2015 09:21:50 +0000 (10:21 +0100)]
rtsp-media: Don't set the transport mode based on what elements we find
Just print a warning if the one that was set before disagrees with what
elements we found. It must already be set to something before as this
function is called after we received the SDP from ANNOUNCE in RECORD mode,
and we would reject ANNOUNCE if the RECORD flag was not set.
Tim-Philipp Müller [Sun, 8 Feb 2015 18:05:50 +0000 (18:05 +0000)]
tests: rtspserver: rename shadowed variable
We have two different 'sink' variables here,
rename one of them for clarity.
Tim-Philipp Müller [Sun, 8 Feb 2015 12:08:36 +0000 (12:08 +0000)]
rtsp-client: fix awkward if clause
Tim-Philipp Müller [Fri, 6 Feb 2015 19:34:17 +0000 (19:34 +0000)]
examples: test-uri: improve uri argument handling and accept file names
Print an error if the argument passed is not a URI and can't
be converted into one, or no arguments have been provided.
Tim-Philipp Müller [Fri, 6 Feb 2015 19:15:40 +0000 (19:15 +0000)]
examples: test-uri: don't remove mount point after 10 seconds
It's very irritating when trying to test stuff repeatedly
and serves no real purpose other than showing that it can
be done.
Tim-Philipp Müller [Wed, 21 Jan 2015 17:32:21 +0000 (17:32 +0000)]
examples: add new test-record to .gitignore
Sebastian Dröge [Wed, 28 Jan 2015 17:54:01 +0000 (18:54 +0100)]
rtsp-media: Use flags to distinguish between PLAY and RECORD media
Sebastian Dröge [Wed, 28 Jan 2015 16:49:16 +0000 (17:49 +0100)]
test-record: Set latency for playback-style example to 2s instead of 200ms
Tim-Philipp Müller [Wed, 21 Jan 2015 17:27:56 +0000 (17:27 +0000)]
tests: add some unit tests for ANNOUNCE and RECORD
https://bugzilla.gnome.org/show_bug.cgi?id=743175
Tim-Philipp Müller [Wed, 21 Jan 2015 16:32:44 +0000 (16:32 +0000)]
rtsp-client: fix a couple of leaks in handle_announce
Sebastian Dröge [Mon, 19 Jan 2015 12:20:39 +0000 (13:20 +0100)]
rtsp-media: Expose latency setting for setting the rtpbin latency
Sebastian Dröge [Sat, 17 Jan 2015 09:28:13 +0000 (10:28 +0100)]
test-record: Use GOptionContext to parse the server port and take the pipeline from the commandline
Sebastian Dröge [Fri, 16 Jan 2015 19:48:42 +0000 (20:48 +0100)]
rtsp-stream: Put the timestamp of receival of the initial packet over TCP on the first buffer
Sebastian Dröge [Fri, 9 Jan 2015 11:40:47 +0000 (12:40 +0100)]
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
Anila Balavan [Fri, 30 Jan 2015 11:50:20 +0000 (12:50 +0100)]
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
Tim-Philipp Müller [Wed, 21 Jan 2015 14:57:03 +0000 (14:57 +0000)]
rtsp-stream: fix false compiler warning
rtsp-stream.c:3034: error: ‘visited’ may be used uninitialized in this function