* Add support to ARM64 configuration for win32 build
* Add changes notes to webrtc and webrtc_aec3 third party sources
* Remove reference to libwebrtc_aec3 from libpjproject. Add reference it manually if libwebrtc_aec3 is needed.
* Fix build error on Release configuration.
- Add new APIs to update/refresh video conference bridge port: `pjmedia_vid_conf_update_port(), pjsua_vid_conf_update_port(), VideoMedia::update()`.
- Use the new API in PJSUA-LIB to update renderer & stream decoder in format changed event.
* - Avoid SSL socket parent/listener getting destroyed during handshake by increasing parent's reference count.
- Add missing SSL socket close when the newly accepted SSL socket is discarded in SIP TLS transport.
* - Fix silly mistake: accepted active socket created without group lock in SSL socket.
- Replace assertion with normal validation check of SSL socket instance in OpenSSL verification callback (verify_cb()) to avoid crash, e.g: if somehow race condition with SSL socket destroy happens or OpenSSL application data index somehow gets corrupted.
- Add fullscreen mode PJMEDIA_VID_DEV_FULLSCREEN_DESKTOP (no video mode change), which is mapped to SDL_WINDOW_FULLSCREEN_DESKTOP.
- Fix resizing while in full-screen.
- Update PJSUA, PJSUA2 & pjsua app, e.g: fullscreen setting was boolean (fullscreen enabled/disabled), now it is enum: disabled, fullscreen, or fullscreen desktop.
- Retry running pjsua upon failure (due to address-in-use error, happened quite frequently on MacOS in GitHub CI).
- Minor fixes: missing 'self', add slight delay after closing socket to reduce possibility of address-in-use error, make sure sip port is not equal to telnet port.
* Avoid deadlock when restarting SIP UDP transport due to holding pjsua
lock.
* Add callback to lock/unlock any lock held when waiting for the read spin loop finish.
* Use simpler approach by unlocking before restarting UDP transport.
* Add doc to pjsip_udp_transport_restart() and pjsip_udp_transport_restart2() of the possibility of deadlock.
Nanang Izzuddin