delta/farstream.git - gitlab.freedesktop.org: farstream/farstream.git

	Commit message (Collapse)	Author	Age	Files	Lines
*	rtpsubstream: Don't free codec after setting it inside substreamfarstream-0.1	Olivier Crête	2013-03-29	1	-8/+6
\| \| \| \|	Bug discovered by Havard Graff
*	rtpsubstream: Free lock to prevent leak	Olivier Crête	2013-03-21	1	-0/+2
\| \| \| \|	Based on patch from Havard Graff
*	fs-rtp-stream: plug session-leak	Havard Graff	2013-03-21	1	-0/+3
\| \| \| \| \|	It might be prettier to use goto and have one point of exit with one unref?
*	fs-rtp-substream: plug capsleak	Havard Graff	2013-03-21	1	-0/+1
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*	fs-codec: plug memory leak	Havard Graff	2013-03-21	1	-2/+0
\| \| \| \|	encoding_name is already g_strdup'ed in codec_new
*	python: Don't hide fs_session/stream_destroy	Olivier Crête	2013-03-18	1	-1/+0
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*	Add dependency to enable parallel build of introspection	Olivier Crête	2012-08-14	1	-0/+1
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*	fs-codec: Remove outdated TODO	Olivier Crête	2012-06-29	1	-2/+0
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*	Gtk-doc tags belong after the description	Olivier Crête	2012-05-25	2	-18/+13
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*	The right macro is GLIB_VERSION_MAX_ALLOWED	Guillaume Desmottes	2012-04-04	1	-1/+1
\| \| \| \|	https://bugs.freedesktop.org/show_bug.cgi?id=48285
*	tests: Remove used of deprecated g_strcasecmp	Olivier Crête	2012-03-26	1	-1/+1
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*	tests: Link against libgstbase when it is used	Olivier Crête	2012-03-26	1	-0/+2
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*	rtpdiscovercodecs: Typecast factory into feature to make compiler happy	Olivier Crête	2012-03-23	1	-1/+2
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*	Version 0.1.2.1	Olivier Crête	2012-03-23	1	-1/+1
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*	Version 0.1.20.1.2	Olivier Crête	2012-03-23	2	-2/+15
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*	Add empty overrides file as the new gtkdoc creates one and to make distcheck ↵	Olivier Crête	2012-03-23	3	-3/+2
\| \| \| \|	happy
*	Ignore rank==0 from auto discovery	Olivier Crête	2012-03-23	1	-1/+5
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*	tests: Make debug messages into GST_DEBUG	Olivier Crête	2012-03-23	1	-3/+3
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*	Update doc generator from common	Olivier Crête	2012-03-23	1	-2/+13
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*	Revert "shm-stream-transmitter: Add property to control the buffer-time"	Olivier Crête	2012-03-23	3	-32/+2
\| \| \| \| \| \| \| \|	This reverts commit 7a8dd5ef57afc9dd279366b6c07a38c30493f927. Because the patch in shmsink was bad and did not deal with timestamps going backwards. We must re-apply this patch once a new upstream GStreamer has been released.
*	rtpcodecnego: Add more debugs on local codec list creation	Olivier Crête	2012-03-21	1	-0/+7
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*	rtpdiscovercodecs: Remove duplicate codecs	Olivier Crête	2012-03-21	1	-0/+29
\| \| \| \| \|	Only keep the first codec if there is more than one way to produce the same RTP codec
*	Set better latency/buffer time for pulse src/sink	Nicolas Dufresne	2012-03-15	1	-3/+4
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*	Add default element properties for rawconference	Nicolas Dufresne	2012-03-15	2	-0/+20
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*	example: Call the right function	Olivier Crête	2012-02-29	1	-1/+1
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*	Update glade file	Olivier Crête	2012-02-29	1	-97/+115
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*	Require GLib 2.30, do not allow APIs added after and ignore later deprecations	Olivier Crête	2012-02-29	2	-2/+5
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*	Don't emit element-added signal without a parent	Olivier Crête	2012-02-28	1	-1/+2
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*	rtpcodecnego: Ignore config while comparing send codecs	Olivier Crête	2012-02-28	1	-3/+10
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*	Remove check for gst <0.13	Olivier Crête	2012-02-28	2	-12/+0
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*	Update common	Olivier Crête	2012-02-22	2	-90/+117
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*	Add "do-timestamp" property to the transmitters	Olivier Crête	2012-02-21	6	-12/+87
\| \| \| \| \|	Make it possible for the "raw" plugin to not have the transmitter put timestamps on the buffers.
*	Version 0.1.1.1	Olivier Crête	2012-02-20	1	-1/+1
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*	Version 0.1.10.1.1	Olivier Crête	2012-02-20	2	-1/+44
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*	Fix python examples	Olivier Crête	2012-02-20	3	-32/+24
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*	Fix python bindings	Olivier Crête	2012-02-20	8	-23/+112
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*	Update defs	Olivier Crête	2012-02-20	1	-0/+118
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*	Add override for fs_stream_set_transmitter	Olivier Crête	2012-02-20	2	-1/+111
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*	python: Use force for force	Olivier Crête	2012-02-20	1	-1/+1
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*	Rename fs2-gui-user-frame.ui to fs-..	Olivier Crête	2012-02-15	1	-0/+0
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*	rtpsession: Wait until stream is destroyed to flush transmitters	Olivier Crête	2012-02-10	1	-7/+7
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*	session: Remove extra ; where it doesn't belong	Olivier Crête	2012-02-10	1	-1/+1
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*	Require new gst-plugins-bad, it's needed for DTMF to work correctly	Olivier Crête	2012-02-10	1	-3/+3
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*	Update GIR annotations	Olivier Crête	2012-01-10	5	-13/+18
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*	Add stream message parsers to the doc	Olivier Crête	2012-01-10	1	-0/+6
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*	Move the FsSession's conference property to the base class	Olivier Crête	2012-01-10	4	-25/+24
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*	rtpsession: Set boolean property to exactly "1" if true	Olivier Crête	2012-01-09	1	-1/+1
\| \| \| \|	It seems boolean properties don't accept values > 1
*	Fix fs_stream_parse_component_state_changed()	Jonathon Jongsma	2011-12-20	1	-4/+4
\| \| \| \| \|	Check for the right message name and parse the state as an enum, rather than a uint.
*	Improve error message for missing stream transmitter	Jonathon Jongsma	2011-12-16	1	-1/+1
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*	shm-stream-transmitter: Add property to control the buffer-time	Sjoerd Simons	2011-11-14	3	-2/+32
\| \| \| \| \| \| \| \| \| \| \|	Allow the transmitter to buffer how much can be maximally queued in the shmsink before blocking. Default to 20 miliseconds. For audio network packets tend to be around 20ms is size, so 20ms is a suitable size for audio. For video, 20 miliseconds is less then one frame at 30 fps, so with 20ms as default at most one video frame will be queued up in the shm sink.