* 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)
-
Allow changing video size after the decoding process start
Allow rotating without scaling to keep ratio
Use right values to compute DirectShow display ratio
- 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