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farstream 0.2.3 (April 15, 2013) ================================ - Use generic marshallers - Fix building by gold linker (Emanuele Aina) - Fix leaks, found by Havard Graff and others - Fix building with automake 1.13 (Nuno Araujo) - Lower PulseAudio latencies (Arun Raghavan) - Fix codec intersection - Add API to make the API be introspection accessible, fixing the Python example - Use GSocket and other win32 portability improvements farstream 0.2.2 (November 13, 2012) ================================= - Update and fix the default properties for vp8enc farstream 0.2.1 (October 4, 2012) ================================= - Fix bug where nothing would be sent - Fix various bugs in ElementAddedNotifier - Fix the GPL headers - Misc bug fixes farstream 0.2.0 (September 25, 2012) ==================================== - Official GStreamer 1.0 release - Ported python example to GStreamer 1.0 and GTK+ 3 - Use GLib 2.32 APIs - Made API more introspection friendly - Ignore Error messages from the decoders - Prefer Opus and VP8 - Various bug fixes farstream 0.1.91 (September 13, 2012) ===================================== - Port to GStreamer 1.0 API farstream 0.1.2 (March 23, 2012) ================================ - Ignore config while comparing send codecs, fixes H.264 and Theora negotation - Require GLib 2.30, do not allow APIs added after and ignore later deprecations - Add default element properties for rawconference - Set better latency/buffer time for pulse src/sink - Remove the buffer-time property on the shm transmitter, because the gst-plugins-bad plugin has a bug, we will restore it once a new gst-plugins-bad version has been released farstream 0.1.1 (February 20, 2012) =================================== - Initial release of Farstream - Not parallel installable with Farsight2 - Added GObject Introspection annotations - Added parser functions for the GstMessages - shm transmiiter:Add a property to control the maximum bufferring time - API changes from Farsight2: * Remove the "error" signal from the participants (they have no methods, no behavior, and emit no errors) * Remove the "cname" parameter from the participant constructor and make the "cname" property specific to RTP and remove it from fs_conference_new_participant() * Remove the debug msg in the error messages * Remove the special hack for ptime in FsCodec and make it a regular parameter * Pass sdes struct as-is to fsrtpconference * Replace FS_DTMF_METHOD_IN_BAND with FS_DTMF_METHOD_SOUND * Return NULL in "codecs" unless they are ready (and "codecs-ready" is not needed anymore) * Replace set_candidates by add_candidates and use force_candidates for rawudp * Set transmitter after creating stream * Fixes possible race: One has a session with one stream, the user creates a new stream, then packets in the new stream come in (with the new stream's cname/ssrc) before the "src-pad-added" signal is connected... ie doesn't link... failure ensues.... solution? Giving the user a chance to link src-pad-added before setting the transmitter ? * Remove fs-interfaces (moved to libnice) * Renamed fs-enum-types.h to fs-enumtypes.h for consistency * Renamed fs-conference-iface.h to fs-conference.h * Rename fs_stream_get_src_pads_iterator() to fs_stream_iterate_src_pads() for consistency * Remove the FS_ERROR_UNKNOWN_CNAME error entirely * Add a _destroy method to session/stream and have the parent keep a ref.. so the session/stream need to be destroyed/closed and we can simplify the teardown code quite a bit * Moved the header files from <gst/farsight/.. to <farstream/...> - Now requires gst-plugins-bad 0.10.23
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