Refactors the way specific Codec instances are accessed in MediaStreamImpl. Passes "format parameters" received via SDP/Jingle to codecs, which can handle them (Opus).

- Fixes spelling mistakes.
- Fixes warnings.

- Fixes the Eclipse .classpath of libjitsi which was broken because of the change of the JSON implementation library.
- Fixes a call-related memory leak.

Temporary fix for the issues with a NotRealizedError exception being thrown, and the camera being turned on in audio-only calls.

- Fixes a crash in Pa_StopStream. - Makes the switching of the audio device while in a call more thorough and resilient.

- Removes the DirectSound support from PortAudio in order to prevent issues with bad audio quality.
- Attempts to recognize USB PortAudio devices on Windows as such, ignores the virtual Microsoft Sound Mapper devices to avoid user confusion. Contributed by Vincent Lucas.

Adds an Opus configuration form and changes the default Opus settings. Updates the Opus JNI binaries. Notifies encoders that implement PacketLossAwareEncoder (Opus and SILK) of packet loss information received via RTCP.

Makes a minor API change. Pyload types are now overridden one by one in order to match the way we do this for the actual payload types

Fixes a discrepancy/inaccuracy in the RTP buffer length and minimum threshold of FMJ which leaves it vularable to even the tinies hiccups of the network. Rebuilds FMJ and libjitsi with debug information such as line numbers.

Moves Constants to the serivce package so that it would be accessible for other packages that need to access a PT name

Moves Constants to the serivce package so that it would be accessible for other packages that need to access a PT name

Updates FMJ to r5 from the sourceforge SVN repository. This version uses a java.util.logging.Logger to log, and logs detailed statistics from the RTPSourceStream class. Changes MediaServiceImpl to always enable FMJ logging.

Improves device selection via portaudio for MacOSX (hostapi coreaudio): adds the possibility to get the device transport type (USB, bluetooth, etc.) and an unique identifier. Corrects problems when: loading the same device twice (for the moment only for the same hostapi), a new plugged device is selected during a call or if the selected device is unchanged.

Allows the remote party to override payload type for the media we send to it. Payload types are no more overridden locally, and remote settings are used only for the media we sent to the that participant.

Prevents multiple dialing sounds from being played for multiple peers in a single call/telephony conference or for a peer who is being added to a telephony conference when there are other peers there already.

Renames a couple of methods upon request by Emil Ivov. Fixes an endless loop on hangup. Commits work in progress on sound notifications.

Commits work in progress on refactorying the call-related user interface for the purposes of audio and video telephony conferencing over single or multiple protocols and Jitsi VideoBridge.

Corrects media stream stats documentation and a null pointer exception which occurs when the media stream is closed.

Improves sound device selection: automatically switches to new plugged device, remember user preferences in a list.