* complete rewrite to add auto detection for thirdparties libraries
* all features are enabled by default unless --without-xxx is used
* adds support for pkg-config to ease integration on other projects (tp, webrtc2sip...)
****** FIRT REVISION KNOW TO FULLY WORK WITH TP ******
- Detect support for h264, h263, theora and mp4v-es at runtime instead of using macros at compile-time
- Allows configuring audio ptime and video fps
- Adds support for congestion control
- Complete support for RTCP-REMB (http://tools.ietf.org/html/draft-alvestrand-rmcat-remb-00)
- Change way the bitrate is computed (use width, height, motion-rank and fps)
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- 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)
- Begin adding support for multiple transports
- Begin adding support for p2p, mcu and mediaproxy modes
- Fix issue gathering issue on Windows 7 and later (tied to webrtc4all issue #1)
- 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
- 1080p (Full HD): all platforms supports full HD video negotiation. Off course it depends on your CPU and network bandwidth. The preferred video size could be changed from the QoS/QoS screen.
- Adaptive video jitter buffer: A video jitter buffer with advanced features like error correction, packet loss retransmission, delay recovery...
- RTP/AVPF profile as per RFC 4585
- RTCP: Full support for RTCP (3550) and many extensions such as: PLI (RFC 4585), SLI (RFC 4585), RPSI (RFC 4585), FIR (RFC 5104), NACK (4585), TMMBN (RFC 5104)...
- rtcp-mux as per 5761
- Negotiation of Generic Image Attributes in the SDP as per RFC 6236
- Source-Specific Media Attributes in SDP as per draft-lennox-mmusic-sdp-source-attributes-01
- Explicit Call Transfer as per 3GPP TS 24.629
bossiel