- Mar 24, 2016
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Sebastian Dröge authored
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- Mar 16, 2016
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Sebastian Dröge authored
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
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- Mar 15, 2016
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Sebastian Dröge authored
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- Mar 10, 2016
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Sebastian Dröge authored
rtsp-stream: Ensure that the pipeline is live and later-added udpsrcs are 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
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Jan Schmidt authored
gst_rtsp_stream_set_client_side -> gst_rtsp_stream_is_client_side
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- Mar 05, 2016
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Sebastian Dröge authored
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
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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
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- Mar 04, 2016
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Sebastian Dröge authored
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.
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- Mar 03, 2016
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Sebastian Dröge authored
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
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- Mar 02, 2016
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Sebastian Dröge authored
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Sebastian Dröge authored
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
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- 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