farstream 0.2.9 (March 11, 2020)
================================
- Add OPUS specific SDP OA negotiation to match RFC
- Switch VP8 to standard encoding name
- Removed MSN plugin, MSN Messenger is long dead
- Port away from deprecated libnice signals
- Bug fixes

farstream 0.2.8 (March 10, 2016)
================================
- Add "require-encryption" parameter to ignore unencrypted packets
- Enable building static GStreamer and transmitter plugins
- Make OPUS plugin work and select it as default codec
- Bug fixes

farstream 0.2.7 (January 29, 2015)
==================================
- Add "send-rtcp-mux" parameters to fsrtpsession
- Add MTU and buffer splitting to rtpxdatapay
- Fix crash if srtpenc/dec is missing
- Bug fixes

farstream 0.2.6 (October 9, 2014)
================================
- Add ICE-TCP support
- Now require libnice 0.1.8

farstream 0.2.5 (October 9, 2014)
================================
- Add SRTP support
- Add API to set allowed input/output caps
- Make it possible to have input that is not a raw format
- Support formats with no encoders/decoders
- Add payloader for the Microsoft Lync x-data format
- Ignore ICE-TCP in new libnice
- Fix misc bugs
- Now require GStreamer 1.4

farstream 0.2.4 (May 5, 2014)
=============================
- Install gtk-doc correctly
- Adapt SSRC handling to GStreamer 1.2 and newer
- Fix BSD build
- Assorted bug fixes

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
