0.12.2 (Mar 14, 2016)
- Fixed relative links in guide.
event.user, instead of the actual deprecation error that was supposed to be thrown.
/kickAPI endpoint instead of stat mutation.
MediaBroadcaster.on('remoteSource, ...)in API reference examples.
room.startPassiveCall(), which created a passive call that doesn’t initiate any connections. Passive calls will automatically accept any incoming calls.
room.startconference(), since they didn’t do anything, and it wasn’t possible to set options for incoming calls.
AudioMeter, which enables monitoring of the audio level of a media source.
MediaSwitchermembers are now sorted based on most recent activity.
DataSourceconstructor, which allows seamless reconnects to calls with existing data.
MediaSwitcherdetect which users are active in the conference.
FileRefs that are received through the same
DataSourceand that are referring to the same remote file are now unique, which enables strict equality (
FileRefvalues in a
DataSourceare now cleared when the peer disconnects.
DataShareupdate events are now always emitted if the value changes, even if the new value is equal to the old one.
room.avatar, along with
MediaSwitcheris now smarter about switching streams and will show the previous stream to the active speaker.
cct.webRtcReadycan be used.
-m, --match <pattern>flag when running tests, if it is set, only test modules matching the pattern will be run.
cct.AdapterJSwas removed and is now private API.
cct.namespace prefix from all classes and interfaces, the members of the
cctnamespace are now documented as properties.
peerConnection.removeStreamcalls were not handled in firefox.
'liveMembers'event whenever they are changed.
MediaSwitcher.setSource(source)not updating streams correctly.
This release features a large rework of the WebRTC and media APIs. A new interface for adding functionality to a call has been added, called RtcComponent. It is the new base for data, file, media, and document sharing, and also provides an interface for building new custom components that work seamlessly in both conferences and two-way calls.
The basic media API has moved on from only having sources that can be attached to calls with
setLocalSource. It is now possible to set up pipelines of media processing nodes, enabling things like mixing video and audio, recording, relaying, adding filters, splitting, merging, etc. The new media pipeline API is built to be extensible and easy to use, while still allowing low level access. A core part of the design is composability, and allowing nodes to be seamlessly combined into container nodes, which can then be used like any other node.
The new APIs are mostly backwards compatible, only some of the more advanced features like data and dom sharing have had breaking changes. The
call.getRemoteSource() APIs are intact.
room.startConferenceCallhas been renamed to
console.debugas the default log handler for debug level messages.
call.detach, for adding and removing RtcComponents, and also updated the conference equivalents.
ProxySinkand a new
AudioMeterto use the new media API.
valueproperty instead of
ScreenSourcein Chrome, and always use default video width and height unless explicitly set.