- Better interop with WebRTC endpoints (better video quality)
- Lock-free on MediaSessionMgr for better performances on both audio and video
- Re-design the video jitter buffer for better CPU prefs and video quality. Request lost frames (RTC-NACK) as many times as required to deal with RTCP-losses. The FPS guesser is smarter.
- Fix issues on RTP timestamps on video pkts
- Update libsrtp binaries on Android and Windows (Use latest CVS)
- Better interop with other h264-rtp implementations (e.g. gstreamer, bria, cisco, polycom, lync...)
- Fix issue 233 (tinyNET does not compile on MAC + fix/patch)
- Fix issue 234 (tinyDAV does not compile on MAC)
- Fix issue 238 (iOS: Bad audio quality when audio/video call uses cpu intensive audio codec (e.g. g729 or speex))
- Fix issue 239 (Adds support for thread priority setting). Timers and audio/video threads now use high priority.
- Fix issue 242 (Hold/Resume fails when audio driver is opensl-es (Android))
- Fix issue 243 (PictureID in VP8 is not correct (only happens when there is overflow on the first 4 bytes))
- Fix issue 244 (Adds callbacks from codecs to session to signal IDR frames decoding)
- Fix issue 245 (Fail to decode h264 buffer)
- FIx issue 246 (Gnu Autotools: Detect support for monotonic timers in configure.ac)
- Add WebRTC audio add-on for Linux systems
- Add OpenSL-ES audio add-on for Android 2.3+
- Fix RTC issue: BYE not sent when the call is ended
- Update Android build scripts to detect all toolchanins (up to 8b), include striping, make NDK r7c as the default toolchain
- Add support for Adaptive echo tail
- Update Speex libraries for Android
- Disable build for speakup-jb code when speex-jb is enabled
- Prevent SIGPIPE signal on iOS when using UDP
bossiel