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2012-04-04The right macro is GLIB_VERSION_MAX_ALLOWEDHEADmasterGuillaume Desmottes1-1/+1
https://bugs.freedesktop.org/show_bug.cgi?id=48285
2012-03-26tests: Remove used of deprecated g_strcasecmpOlivier Crête1-1/+1
2012-03-26tests: Link against libgstbase when it is usedOlivier Crête1-0/+2
2012-03-23rtpdiscovercodecs: Typecast factory into feature to make compiler happyOlivier Crête1-1/+2
2012-03-23Version 0.1.2.1Olivier Crête1-1/+1
2012-03-23Version 0.1.2Olivier Crête2-2/+15
2012-03-23Add empty overrides file as the new gtkdoc creates one and to make distcheck ↵Olivier Crête3-3/+2
happy
2012-03-23Ignore rank==0 from auto discoveryOlivier Crête1-1/+5
2012-03-23tests: Make debug messages into GST_DEBUGOlivier Crête1-3/+3
2012-03-23Update doc generator from commonOlivier Crête1-2/+13
2012-03-23Revert "shm-stream-transmitter: Add property to control the buffer-time"Olivier Crête3-32/+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.
2012-03-21rtpcodecnego: Add more debugs on local codec list creationOlivier Crête1-0/+7
2012-03-21rtpdiscovercodecs: Remove duplicate codecsOlivier Crête1-0/+29
Only keep the first codec if there is more than one way to produce the same RTP codec
2012-03-15Set better latency/buffer time for pulse src/sinkNicolas Dufresne1-3/+4
2012-03-15Add default element properties for rawconferenceNicolas Dufresne2-0/+20
2012-02-29example: Call the right functionOlivier Crête1-1/+1
2012-02-29Update glade fileOlivier Crête1-97/+115
2012-02-29Require GLib 2.30, do not allow APIs added after and ignore later deprecationsOlivier Crête2-2/+5
2012-02-28Don't emit element-added signal without a parentOlivier Crête1-1/+2
2012-02-28rtpcodecnego: Ignore config while comparing send codecsOlivier Crête1-3/+10
2012-02-28Remove check for gst <0.13Olivier Crête2-12/+0
2012-02-22Update commonOlivier Crête2-90/+117
2012-02-22Add "do-timestamp" property to the transmittersOlivier Crête6-12/+87
Make it possible for the "raw" plugin to not have the transmitter put timestamps on the buffers.
2012-02-20Version 0.1.1.1Olivier Crête1-1/+1
2012-02-20Version 0.1.1Olivier Crête2-1/+44
2012-02-20Fix python examplesOlivier Crête3-32/+24
2012-02-20Fix python bindingsOlivier Crête8-23/+112
2012-02-20Update defsOlivier Crête1-0/+118
2012-02-20Add override for fs_stream_set_transmitterOlivier Crête2-1/+111
2012-02-20python: Use force for forceOlivier Crête1-1/+1
2012-02-15Rename fs2-gui-user-frame.ui to fs-..Olivier Crête1-0/+0
2012-02-10rtpsession: Wait until stream is destroyed to flush transmittersOlivier Crête1-7/+7
2012-02-10session: Remove extra ; where it doesn't belongOlivier Crête1-1/+1
2012-02-10Require new gst-plugins-bad, it's needed for DTMF to work correctlyOlivier Crête1-3/+3
2012-01-10Update GIR annotationsOlivier Crête5-13/+18
2012-01-10Add stream message parsers to the docOlivier Crête1-0/+6
2012-01-10Move the FsSession's conference property to the base classOlivier Crête4-25/+24
2012-01-09rtpsession: Set boolean property to exactly "1" if trueOlivier Crête1-1/+1
It seems boolean properties don't accept values > 1
2011-12-20Fix fs_stream_parse_component_state_changed()Jonathon Jongsma1-4/+4
Check for the right message name and parse the state as an enum, rather than a uint.
2011-12-16Improve error message for missing stream transmitterJonathon Jongsma1-1/+1
2011-11-14shm-stream-transmitter: Add property to control the buffer-timeSjoerd Simons3-2/+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.
2011-11-14Add parsing functions for all GstMessagesOlivier Crête6-19/+322
2011-11-08session: Put the messages argument into a real tableOlivier Crête1-22/+62
2011-11-08session: Add functions to parse the custom GstMessagesOlivier Crête3-0/+194
2011-11-07session: Make the volume into a gucharOlivier Crête2-2/+2
2011-11-07session: Document the farsight-telephony-event-started/stopped messagesOlivier Crête4-6/+52
Also use the specific registered types where appropriate
2011-11-07rtpsession: Document the dtmf messages a bit moreOlivier Crête1-3/+7
2011-11-07Remove non-existing fs_add_constantsOlivier Crête1-2/+0
2011-11-04stream: Clarify the origin of the transmitter parametersOlivier Crête1-0/+3
2011-11-03rtpsession: Queue events inside instead of letting the sources do itOlivier Crête5-33/+352
Also, make it possible to iterate the sessions without holding the object lock Also, we track when the sources are done processing an event so that we can minimize the number of dropped events if we change the source.