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2016-05-25g-i: pass compiler env to g-ir-scannerHEADmasterTim-Philipp Müller1-0/+1
It's what introspection.mak does as well. Should fix spurious build failures on gnome-continuous (caused by g-ir-scanner getting compiler details via python which is broken in some environments so passing the compiler details bypasses that).
2016-05-19rtsp-session: RFC2326 does not allow a space between ; and timeout in the ↵Ian1-1/+1
Session header This works with rtspsrc and live555, but fails with e.g. ffmpeg. https://bugzilla.gnome.org/show_bug.cgi?id=766619
2016-04-29rtspclientsink: Check return value of sscanfEdward Hervey1-5/+3
And just make sure we always have 0/0 if we have an error CID #1352031
2016-04-29rtsp-stream: Fix crash on cleanup with shared media and multiple udpsrcJake Foytik3-172/+287
- Unicast udpsrcs are now managed in a hash table. This allows for proper cleanup in with shared streams and fixes a memory leak. - Unicast udpsrcs are now properly cleaned up when shared connections exit. See the update_transport() function. - Create unit test for shared media. https://bugzilla.gnome.org/show_bug.cgi?id=764744
2016-04-29rtsp-stream: Always bind to ANY when address is a multicast address and not ↵Sebastian Dröge1-10/+2
only on Windows For IPv6 addresses, binding to a multicast group does not work on Linux either. Always bind to ANY and then later join the multicast group. https://bugzilla.gnome.org/show_bug.cgi?id=764679
2016-04-14Automatic update of common submoduleJulien Isorce1-0/+0
From 6f2d209 to ac2f647
2016-04-06rtsp-thread-pool: explained why GSource is a part of ThreadImplPatricia Muscalu1-0/+2
Clarified why it is necessary to add source information to GstRTSPThreadImpl. See the reported bug in GLib: https://bugzilla.gnome.org/show_bug.cgi?id=720186 for more information. https://bugzilla.gnome.org/show_bug.cgi?id=761702
2016-04-04examples: Clean up CFLAGS/LDADD even moreSebastian Dröge1-13/+9
The internal .la should come first and is part of LDADD, as is GST_CFLAGS/LIBS.
2016-04-04examples: Clean up CFLAGS/LDADD to link with the correct versions of all ↵Sebastian Dröge1-5/+15
libraries
2016-04-03rtsp-server: Use $(GST_NET_LIBS) / $(GST_NET_CFLAGS)Sebastian Dröge1-3/+4
2016-04-03rtsp-server: Implement clock signalling according to RFC7273Sebastian Dröge8-0/+363
For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2016-03-25rtspclientsink: Add support for setting the multicast interfaceSebastian Dröge1-0/+1
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-25rtsp-media: Add support for setting the multicast interfaceSebastian Dröge6-2/+208
https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-24rtspclientsink: use new gst_element_class_add_static_pad_template()Vineeth TM1-2/+1
https://bugzilla.gnome.org/show_bug.cgi?id=763196
2016-03-24Back to developmentSebastian Dröge1-6/+6
2016-03-24Release 1.8.0Sebastian Dröge5-19/+830
2016-03-16rtsp-stream: Don't set the state of the appsrc from PLAYING to PAUSED again ↵Sebastian Dröge1-2/+0
during setup This would get us NO_PREROLL in the bin again and break seeking. Thanks to Carlos Rafael Giani for helping to debug this! https://bugzilla.gnome.org/show_bug.cgi?id=740509
2016-03-15Release 1.7.91Sebastian Dröge5-13/+103
2016-03-10rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are ↵Sebastian Dröge1-3/+22
syncing the state with the parent bin Without this, RECORD pipelines are broken because a) we wait for ASYNC_DONE which never happens anymore because udpsrc would be added later. Previously it was there earlier and due to NO_PREROLL caused the pipeline to preroll immediately b) the udpsrc for the pipeline is added later and never set to PLAYING state, as the corresponding code previously was only for PLAY pipelines. https://bugzilla.gnome.org/show_bug.cgi?id=763281
2016-03-11rtsp-stream: Fix typo in the docstringJan Schmidt1-1/+1
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
2016-03-05rtsp-stream: Disable multicast loopback for all our socketsSebastian Dröge1-0/+2
On Windows this is a receiver-side setting, on Linux a sender-side setting. As we provide a socket ourselves to udpsrc, udpsrc is never setting the multicast loopback setting on the socket... while udpsink does which unfortunately has no effect here on Windows but on Linux. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-05stream tests: added new testsPatricia Muscalu1-0/+209
Test a case when the address pool only contains multicast addresses and the client is requesting unicast udp. Added tests for multicast ports allocation. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-04rtsp-stream: Only bind multicast sockets to ANY on WindowsSebastian Dröge1-3/+10
On Linux it is still needed to bind to the multicast address to filter out random other packets, while on Windows binding to multicast addresses just fails.
2016-03-03rtsp-stream: Only use the address pool for unicast UDP if it contains ↵Sebastian Dröge1-9/+8
unicast addresses Otherwise we fail to allocate UDP ports if the pool only contains multicast addresses, which is something that used to work before. For unicast addresses if the pool contains none, we just allocate them as if there is no pool at all. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-02rtsp-server: Fix indentationSebastian Dröge2-20/+25
2016-03-02rtsp-stream: Don't bind the sockets to multicast addressesSebastian Dröge1-0/+9
This works on Linux but fails completely on Windows. You're supposed to bind to ANY and then join the multicast group. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-03-01Release 1.7.90Sebastian Dröge5-28/+134
2016-02-26Automatic update of common submoduleSebastian Dröge1-0/+0
From b64f03f to 6f2d209
2016-02-24rtspsink: Fix some leaks in rtspclientsink and the unit test.Jan Schmidt2-2/+9
https://bugzilla.gnome.org/show_bug.cgi?id=762525
2016-02-23tests: unit test fixesPatricia Muscalu4-42/+72
Removed port allocation test from the media suite. The port allocation failure is now in the stream suite. rtspserver: Make sure that the media is suspended after the DESCRIBE request before reconfiguring the UDP sinks. rtspclientsink: In the RECORD case we have to set async property to false for the appsink element in the test in order to make sure that the media pipeline doesn't hang in start_preroll(). https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: postpone UDP socket allocation until SETUPPatricia Muscalu3-180/+186
Postpone the allocation of the UDP sockets until we know what transport has been chosen by the client. Both unicast and multicast UDP sources are created in one function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: postpone the creation of the UDP sourcesPatricia Muscalu1-16/+59
Code refactoring: allocate the UDP ports after the sender and the reciver parts have been created. We postpone the creation of the UDP sources until the UDP ports have been allocated. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: added function for setting UDP sources to PLAYING statePatricia Muscalu1-0/+54
Code refactoring: Introduced a function for setting UDP sources to PLAYING state. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: added function for creating and configuring UDP sourcesPatricia Muscalu1-38/+43
Code refactoring: create and configure UDP sources in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: added function for RTP/RTCP socket configurationPatricia Muscalu1-8/+19
Code refactoring: configure RTP and RTCP sockets for UDP sinks in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: added function for creating and configuring UDP sinksPatricia Muscalu1-64/+85
Code refactoring: create and configure UDP sinks in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23rtsp-stream: added helper function for creating the sender/receiver partsPatricia Muscalu1-127/+166
Code refactoring: introduced helper function for creating the receiver and the sender parts of the streaming pipeline. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-19Back to developmentSebastian Dröge1-5/+5
2016-02-19Release 1.7.2Sebastian Dröge5-30/+259
2016-02-18uninstalled.pc: add support for non libtool build systemsJulien Isorce1-2/+2
Currently the .la path is provided which requires to use libtool as mentioned in the GStreamer manual section-helloworld-compilerun.html. It is fine as long as the application is built using libtool. So currently it is not possible to compile a GStreamer application within gst-uninstalled with CMake or other build system different than autotools. This patch allows to do the following in gst-uninstalled env: gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \ gstreamer-rtsp-server-1.0) Previously it required to prepend libtool --mode=link https://bugzilla.gnome.org/show_bug.cgi?id=720778
2016-02-09rtspclientsink: remove check for impossible conditionLuis de Bethencourt1-2/+0
Goto error label checks stream to see if it needs to be unreferenced before returning, but this goto jumps happens before the stream is ever set, so it will always be NULL in this error label. CID #1352034
2016-02-08rtspclientsink: clean switch statementsLuis de Bethencourt1-0/+2
Coverity demands for fallthrough statements to be clearly commented, to distinguish from accidental fall throughs. And it also needs all cases to finish with a break, even if the break is never going to be executed like in the case of a continue jump. CID #1352039 CID #1352040
2016-02-05tests: extend the AM_TESTS_ENVIRONMENT from check.makThiago Santos1-2/+1
To get the CK_DEFAULT_TIMEOUT defined for all tests Also removes a 120 seconds timeout that was set as default explicitly in this module https://bugzilla.gnome.org/show_bug.cgi?id=761472
2016-02-05Automatic update of common submoduleThiago Santos2-0/+2
From 86e4663 to b64f03f
2016-02-02rtsp-media: fix state_lock not locked again when preroll failsSteven Hoving1-3/+5
https://bugzilla.gnome.org/show_bug.cgi?id=761399
2016-01-28configure: Move plugin specific flags below all the othersSebastian Dröge1-25/+25
They use some of the other flags, like $GST_ALL_LDFLAGS which is adding -no-undefined. And -no-undefined is required on Windows to build DLLs.
2016-01-29rtspclientsink: Simplify slightly using new -base APIJan Schmidt1-120/+16
Use the new Mikey and SDP API in the base plugins libs to simplify some code. https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29rtspsink: Add rtspclientsink elementJan Schmidt9-4/+5428
Add an rtspclientsink element that accepts streams for which there is a registered payloader and sends them to an RTSP server using RECORD. Sending is synchronised to the pipeline clock. Payload-types are automatically selected. The 'new-payloader' signal is fired for custom configuration of payloaders when they are created. Can now stream a movie like this: receiver: ./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \ decodebin name=depay1 ! audioconvert ! autoaudiosink )" sender: gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \ queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \ https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29rtsp-stream: Add functions for using rtsp-stream from the clientJan Schmidt2-2/+69
Add a boolean to indicate that the rtsp-stream is running on the 'client' side of an RTSP connection, for sending streams via RECORD. In that case, the roles of the client/server ports in transport setup are swapped. https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-29rtsp-sdp: Add gst_rtsp_sdp_from_stream()Jan Schmidt2-23/+41
A new function that adds info from a GstRTSPStream into an SDP message. https://bugzilla.gnome.org/show_bug.cgi?id=758180
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