- Mar 01, 2016
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Sebastian Dröge authored
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- Feb 26, 2016
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Sebastian Dröge authored
From b64f03f to 6f2d209
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- Feb 23, 2016
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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
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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
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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
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Code refactoring: Introduced a function for setting UDP sources to PLAYING state. https://bugzilla.gnome.org/show_bug.cgi?id=757488
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Code refactoring: create and configure UDP sources in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
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Code refactoring: configure RTP and RTCP sockets for UDP sinks in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
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Code refactoring: create and configure UDP sinks in a separate function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
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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
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- Feb 19, 2016
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Feb 18, 2016
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Julien Isorce authored
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
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- Feb 09, 2016
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Luis de Bethencourt authored
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
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- Feb 08, 2016
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Luis de Bethencourt authored
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
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- Feb 05, 2016
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Thiago Santos authored
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
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Thiago Santos authored
From 86e4663 to b64f03f
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- Feb 02, 2016
- Jan 28, 2016
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Sebastian Dröge authored
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.
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Jan Schmidt authored
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
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Jan Schmidt authored
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
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Jan Schmidt authored
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
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Jan Schmidt authored
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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- Jan 15, 2016
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Tim-Philipp Müller authored
CID 1348453
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- Jan 07, 2016
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When RTSP server trying update transport during multicast, it throws an assert. The assert is thrown because it is trying to get the parent of an non-existing funnel element. https://bugzilla.gnome.org/show_bug.cgi?id=760150
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- Jan 03, 2016
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Tim-Philipp Müller authored
gtk-doc can handle static inline functions just fine these days, there's no need for this stuff any more.
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- Dec 31, 2015
- Dec 30, 2015
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Dec 28, 2015
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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
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Sebastian Dröge authored
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
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- Dec 24, 2015
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Sebastian Dröge authored
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Sebastian Dröge authored
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- Dec 21, 2015
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Update the -Bsymbolic check with the version glib has. This version works with clang. https://bugzilla.gnome.org/show_bug.cgi?id=759713
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- Dec 15, 2015
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Olivier Crête authored
Live555 in VLC strips off dollar signs and then gets very confused, we don't loose too much entropy by just skipping it.
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