Commit graph

1292 commits

Author SHA1 Message Date
Vineeth TM 6d20a1c9d9 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
2015-09-26 09:35:17 +01:00
Sebastian Dröge 9c513cc536 Back to development 2015-09-25 23:51:17 +02:00
Sebastian Dröge 8a8bb37f8d Release 1.6.0 2015-09-25 23:32:52 +02:00
Sebastian Dröge e4edaebe8e Release 1.5.91 2015-09-18 20:12:06 +02:00
Tim-Philipp Müller da8a31ac88 stream: fix docs for recently-added get/set_buffer_size API
https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-09-17 20:07:34 +01:00
Jan Schmidt 315c2f93bb 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
2015-09-09 17:57:15 +10:00
Jan Schmidt 22b618836e test-mp4: Support filenames with spaces in them. Error out on too few arguments 2015-09-03 22:20:11 +10:00
Jan Schmidt 2a41502cde test-record: Check parameter count and print out help
If no launch pipeline was supplied, print out some help
2015-09-03 22:20:11 +10:00
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
Jan Schmidt 9bfcdba42b rtsp-media: Fix small typo causing gtk-doc to complain 2015-09-03 22:16:30 +10:00
Sebastian Dröge 43bac0c2d9 Release 1.5.90 2015-08-19 14:15:23 +03:00
Hyunjun Ko 4c6b1faa6a media-factory: get port number through gst_rtsp_url_get_port
https://bugzilla.gnome.org/show_bug.cgi?id=753473
2015-08-16 12:08:49 +02:00
Francisco Velazquez 418e1fe090 media-test: Removing unnecessary assertion
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-13 18:48:51 -04:00
Xavier Claessens 8511ffe178 Document that source keeps a ref on server until it's destroyed
https://bugzilla.gnome.org/show_bug.cgi?id=749227
2015-08-10 12:18:53 -04:00
Nicolas Dufresne 3667e71b2f media-test: Test for multiple dynamic payload
https://bugzilla.gnome.org/show_bug.cgi?id=753385
2015-08-08 11:13:58 -04:00
Nicolas Dufresne 707ac9c487 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
2015-08-08 09:46:40 -04:00
Nicolas Dufresne 160b87430f Revert "rtsp-media: Only add 1 fakesink per pipeline"
This reverts commit 22bf61f16c.
2015-08-08 09:08:37 -04:00
Nicolas Dufresne 22bf61f16c 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.
2015-08-07 09:33:55 -04:00
Vineeth TM 3920e21cd0 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
2015-07-30 15:52:08 +03:00
Sebastian Dröge ae7bec97cb 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
2015-07-29 11:28:21 +01:00
Xavier Claessens 5585dc5878 threadpool: Fix possible warning in gst_rtsp_thread_pool_cleanup()
https://bugzilla.gnome.org/show_bug.cgi?id=752640
2015-07-20 16:47:05 -04:00
Stefan Sauer c809dc1735 Automatic update of common submodule
From f74b2df to 9aed1d7
2015-07-03 22:00:00 +02:00
Sebastian Dröge dc9f11c27f Back to development 2015-06-25 00:04:28 +02:00
Sebastian Dröge d8fff9627d Release 1.5.2 2015-06-24 23:44:37 +02:00
Ognyan Tonchev 8922afb88d 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
2015-06-23 14:38:29 +01:00
Nicolas Dufresne d59c7981cc Automatic update of common submodule
From 6015d26 to f74b2df
2015-06-16 17:50:26 -04:00
Ognyan Tonchev fb71b9c4e9 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
2015-06-16 11:09:37 +02:00
Sebastian Dröge fdfe97f447 test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers 2015-06-13 17:14:43 +02:00
Sebastian Dröge a5044fa1f9 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.
2015-06-12 23:35:32 +02:00
Sebastian Dröge 028a4666fa 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
2015-06-11 20:41:31 +02:00
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 93d37df0c3 docs: add missing types
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 11:34:42 +01:00
Hyunjun Ko 8c1eb6fb4f docs: add missing apis
https://bugzilla.gnome.org/show_bug.cgi?id=750764
2015-06-11 11:34:22 +01:00
Sebastian Dröge 5c5850b6b1 test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization 2015-06-10 17:14:44 +02:00
Xavier Claessens 6ec8fe44b2 GstRTSPAuth: Add client certificate authentication support
https://bugzilla.gnome.org/show_bug.cgi?id=750471
2015-06-09 19:51:46 -04:00
Sebastian Dröge af2cb6445a test-netclock-client: Use new GstClock API to wait for clock synchronization 2015-06-09 13:53:47 +02:00
Sebastian Dröge 6219766555 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.
2015-06-09 13:52:05 +02:00
Edward Hervey 2c935a7884 Automatic update of common submodule
From d9a3353 to 6015d26
2015-06-09 11:30:54 +02:00
Stefan Sauer 7436fee689 Automatic update of common submodule
From d37af32 to d9a3353
2015-06-08 23:08:34 +02:00
Stefan Sauer 1541d3dd8a Automatic update of common submodule
From 21ba2e5 to d37af32
2015-06-07 23:07:31 +02:00
Stefan Sauer 68dbad0967 Automatic update of common submodule
From c408583 to 21ba2e5
2015-06-07 17:32:29 +02:00
Stefan Sauer 260e577b9c docs: remove variables that we define in the snippet from common
This is syncing our Makefile.am with upstream gtkdoc.
2015-06-07 17:19:10 +02:00
Stefan Sauer 4aaab390c4 Automatic update of common submodule
From 44a3517 to c408583
2015-06-07 17:16:47 +02:00
Sebastian Dröge 9c75932b16 Back to development 2015-06-07 16:44:55 +02:00
Sebastian Dröge e86bbbb66c Release 1.5.1 2015-06-07 11:20:01 +02:00
Göran Jönsson 08e0c79cee 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
2015-06-03 15:09:10 +02:00
Tim-Philipp Müller 550348738c tests: Use AM_TESTS_ENVIRONMENT
Needed by the new automake test runner and the
current version of the common submodule.
2015-05-27 17:04:41 +01:00
Sebastian Dröge 8700468499 rtsp-server: Use single-include rtsp header to make sure we get all definitions 2015-05-20 17:05:47 +03:00
Sebastian Dröge 1c30c60e64 rtsp-media: Mark some more functions static 2015-05-05 16:46:57 +02:00
Sebastian Dröge bbdf0a47d1 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.
2015-05-05 16:46:19 +02:00