Sebastian Dröge [Wed, 30 Dec 2015 14:34:30 +0000 (16:34 +0200)]
test-netclock: Use the new API to configure a clock directly
Sebastian Dröge [Wed, 30 Dec 2015 14:31:13 +0000 (16:31 +0200)]
rtsp-media: Add API to directly configure a clock on the media pipelines
Sebastian Dröge [Wed, 30 Dec 2015 14:43:17 +0000 (16:43 +0200)]
rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
Sebastian Dröge [Wed, 30 Dec 2015 14:30:38 +0000 (16:30 +0200)]
rtsp-media-factory: Add FIXME for 2.0
Sebastian Dröge [Wed, 30 Dec 2015 14:29:45 +0000 (16:29 +0200)]
rtsp-stream: Fix indentation
Sebastian Rasmussen [Tue, 22 Dec 2015 11:08:02 +0000 (12:08 +0100)]
rtsp-media: Do not prepare media after media times out
Deferred calls to start_prepare() can be deferred past the point until
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
prepared to wait. Previously there was no lock and no check for this
situation. This meant that a media could be prepared and unprepared
simultaneously by two different threads. Now a lock is in place and a
suitable check is done.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
Sebastian Dröge [Wed, 9 Dec 2015 16:24:24 +0000 (18:24 +0200)]
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
Without TEARDOWN it might be desireable to keep the media running and continue
sending data to the client, even if the RTSP connection itself is
disconnected.
Only do this for session medias that have only UDP transports. If there's at
least on TCP transport, it will stop working and cause problems when the
connection is disconnected.
https://bugzilla.gnome.org/show_bug.cgi?id=758999
Sebastian Dröge [Thu, 24 Dec 2015 14:29:33 +0000 (15:29 +0100)]
Back to development
Sebastian Dröge [Thu, 24 Dec 2015 13:54:06 +0000 (14:54 +0100)]
Release 1.7.1
Koop Mast [Sun, 20 Dec 2015 23:43:49 +0000 (00:43 +0100)]
configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.
https://bugzilla.gnome.org/show_bug.cgi?id=759713
Olivier Crête [Wed, 18 Nov 2015 03:30:54 +0000 (22:30 -0500)]
rtsp-session-pool: Avoid dollar sign ($) in session ids
Live555 in VLC strips off dollar signs and then gets very confused,
we don't loose too much entropy by just skipping it.
Xavier Claessens [Tue, 10 Nov 2015 19:17:18 +0000 (14:17 -0500)]
rtsp-server: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
Srimanta Panda [Tue, 8 Dec 2015 07:27:20 +0000 (08:27 +0100)]
rtsp-stream: fixed valgrind error
Fixed the valgrind error in unit test. The UDP source created during
gst_rtsp_stream_join_bin() was not released while destroying the rtp
bin.
https://bugzilla.gnome.org/show_bug.cgi?id=759010
Nicolas Dufresne [Mon, 7 Dec 2015 14:11:35 +0000 (09:11 -0500)]
Automatic update of common submodule
From b319909 to 86e4663
Srimanta Panda [Wed, 18 Nov 2015 10:14:39 +0000 (11:14 +0100)]
rtsp-client: suspend media during setup request
SETUP request from clients needs to suspend the media to clear the
prerolled buffers. Otherwise it will not affect the prerolled buffer
and the prerolled buffers will be incorrect (for example block-size
from setup request will not affect the prerolled buffer unless the
media is suspended).
https://bugzilla.gnome.org/show_bug.cgi?id=758268
Srimanta Panda [Fri, 4 Dec 2015 07:01:37 +0000 (08:01 +0100)]
rtsp-stream: create stream pipeline based on transport
Based on the protocol, create the rtsp stream pipeline. If only TCP or
only UDP is set as the transport protocol, it will not add the extra tee
or queue element to the pipeline. Both these elements will be added, if
it supports both TCP and UDP protocols. This improves the pipeline
performance when one protocol is present.
https://bugzilla.gnome.org/show_bug.cgi?id=758179
Sebastian Dröge [Thu, 19 Nov 2015 13:01:16 +0000 (15:01 +0200)]
rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
Adding them when not needed will start some logic inside rtpbin that might be
problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
would start up a rtpjitterbuffer and behave in weird ways.
We still set up the UDP sources for RTP receiving for a sender media to be
able to receive any packets sent by the client for NAT traversal. They will
all go to a fakesink though.
Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
NO_PREROLL, which will cause deadlocks when seeking the media as it will never
receive ASYNC_DONE after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=758319
Sebastian Dröge [Tue, 17 Nov 2015 10:44:38 +0000 (12:44 +0200)]
rtsp-stream: Disable multicast loopback for the multicast udp sources too
On POSIX this setting is for sender sockets, on Windows for receiver sockets.
Previously we were only setting this for sender sockets, which caused looped
back packets to be received on Windows if a multicast transport was used.
Jan Schmidt [Mon, 16 Nov 2015 14:12:28 +0000 (01:12 +1100)]
examples: Actually use the provided port in the record examples
Jan Schmidt [Mon, 16 Nov 2015 14:12:28 +0000 (01:12 +1100)]
test-record-auth: Add the option to build in TLS support
Jan Schmidt [Mon, 16 Nov 2015 14:12:28 +0000 (01:12 +1100)]
test-auth: Use an 'anonymous' user for unauthenticated default
There's a comment on one of the resources that 'user' and 'admin'
shouldn't even be able to see it, but they can if the default
token is 'admin2', since that gives them access anyway.
Jan Schmidt [Mon, 16 Nov 2015 14:12:28 +0000 (01:12 +1100)]
Add test-record-auth example
Jan Schmidt [Mon, 16 Nov 2015 14:12:28 +0000 (01:12 +1100)]
rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
Marcus Prebble [Wed, 11 Nov 2015 13:58:33 +0000 (14:58 +0100)]
rtsp-server: Change the logic so we don't pop a NULL context
When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
will sometimes fail. This call is made before any context is pushed
resulting in an attempt to pop a NULL context.
https://bugzilla.gnome.org/show_bug.cgi?id=757949
David Svensson Fors [Thu, 22 Oct 2015 12:32:30 +0000 (14:32 +0200)]
rtspserver: Add udp-mcast transport SETUP test
Refactor utility functions in the test file so they can handle
more than UDP and TCP as lower transport.
https://bugzilla.gnome.org/show_bug.cgi?id=756969
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 [Wed, 21 Oct 2015 13:37:19 +0000 (14:37 +0100)]
Automatic update of common submodule
From b99800a to b319909
Sebastian Dröge [Tue, 20 Oct 2015 14:29:42 +0000 (17:29 +0300)]
Use new GST_ENABLE_EXTRA_CHECKS #define
https://bugzilla.gnome.org/show_bug.cgi?id=756870
Sebastian Dröge [Wed, 21 Oct 2015 11:28:47 +0000 (14:28 +0300)]
Automatic update of common submodule
From 6babecd to b99800a
Sebastian Dröge [Fri, 2 Oct 2015 19:25:47 +0000 (22:25 +0300)]
Update GLib dependency to 2.40.0
Hyunjun Ko [Fri, 2 Oct 2015 07:11:05 +0000 (16:11 +0900)]
stream: listen to sender ssrc signals
https://bugzilla.gnome.org/show_bug.cgi?id=746747
Tim-Philipp Müller [Tue, 29 Sep 2015 12:00:51 +0000 (13:00 +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
Vineeth TM [Thu, 20 Aug 2015 08:01:24 +0000 (17:01 +0900)]
rtsp-server: Fix memory leaks when context parse fails
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753863
Sebastian Dröge [Fri, 25 Sep 2015 21:51:17 +0000 (23:51 +0200)]
Back to development
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.