feat: use getSynchronizationSources on the receiver for remote audio levels (#1245)
* feat: use getSynchronizationSources on the receiver for remote audio levels
Use getSynchronizationSources if it is supported, fallback to using getStats otherwise.
* feat/ref: Use the local audio levels from LocalStatsCollector
When using getSynchronizationSources, use the audio levels from LocalStatsCollector for NoAudioSignalDetection.js
Remove obsolete code - TalkMutedDetection feature using audio levels is not used anymore
Partially reverts 24bda8e and uses domain/roomname to report to cs.
To be sure we always report room name in small case and as mobile and jigasi report this way it will take time for them to adopt which leads to wrong stats.
In TraceablePeerConnection: we're no longer injecting a recvonly SSRC
when the local video track is muted, so it's normal that there is no
SSRC found in the local SDP when it's muted.
About RTPStatsCollector: at the time of adding this log statement a case
was missed when local audio track could be replaced in the P2P
connection when a new audio device is selected.
It should be provided by the environment, just like on web. This avoids bundling
callstats on web, where it won't be used (we dynamically download it).
This reduces the bundle size from about 1MB to 675KB, almost 40%.
Handles private messages received by speakerstats component.
Updates speakers stats with values received by the spealerstats component, all the logic is activate only in case we discover speakerstats component address from disco-info and the incoming message is coming from that component.
When the main domain advertises component with type 'speakerstats' every participant will send message to the component with the name of the room where this happen.
core: refactor initialization not to return a Promise
There is nothing asynchronous about the initialization process (anymore), thus
turn it into a synchronous method.
In addition, WebRTC support is absolute, it cannot change from not being
supported to being supported (as it plreviously could, thanks to Temasys) so get
rid of the ancillary logic to support that.
Last, introduce a way to check if WebRTC is supported in the current
environment: JitsiMeetJS.isWebRtcSupported().
Enables RTPStatsCollector for react-native. The getStats method was
supported long time ago, but the stats produced by the RTPStatsCollector
were not consumed. Now they are needed for the automated testing on RN.
Implements the promised based getStats. Enables them for Safari and FF.
Adds stats and audio levels for Safari. Enables the new getStats API for Firefox, that will get rid of the following warning:
'non-maplike pc.getStats access is deprecated, and will be removed in the near future! See http://w3c.github.io/webrtc-pc/#getstats-example for usage.'
Avoid throwing an exception if an analytics handler throws one. Emit a warning
though, so the problem can be eventually caught and fixed in the handler.
* wip: initial version of the new AnalyticsAdapter.
* ref: Restructures the ICE duration and state change events.
* ref: Restructures the JitsiLocalTrack events.
* ref: Restructures the TTFM events.
* ref: Updates the user feedback event.
* ref: Restructures the _CONNECTION_TIMES_ and TTFM events.
* ref: Restructures the BRIDGE_DOWN and NO_DATA_FROM_SOURCE events.
* ref: Restructures the FOCUS_LEFT event.
* ref: Restructures the DATA_CHANNEL_OPEN event.
* ref: Removes the ICE_FAILED event (it is a duplicate of a state change event).
* ref: Restructures the device list events.
Uses one event per device, since the new format does not allow non-atomic attributes.
* fix: Does not obey "unmute" commands from the focus.
* ref: Restructures the "remotely muted" event.
* ref: Restructures the CONFERENCE_ERROR events.
* ref: Removes the CONNECTION_INTERRUPTED event
We can use ICE_STATE_CHANGED instead.
* ref: Renames isreconnect to isReconnect.
* ref: Removes the CONNECTION_RESTORED event. Use ICE state changes instead.
* ref: Restructures the p2p events.
* ref: Restructures the jingle events.
* ref: Restructures the RTP statistics event.
* ref: Restructures the CONNECTION_FAILED and DISCONNECTED events.
* ref: Restructures the getUserMedia analytics events.
* ref: Cleans up AnalyticsEvents and restructures some of the events.
* fix: Adds error logs to the analytics adapter.
* ref: Refactor Statistics.sendEventToAll
Renames to sendEventAndLog, supports the object-based API, uses the
function where appropriate.
* fix: Addresses PR feedback.
* fix: Addresses Lyubomir's feedback.
* ref: Remove unused functions, adds documentation.
* feat: Adds a Statistics.sendAnalytics shortcut.
* ref: Uses the conference name as the default containerId.
* fix: Adrdesses Lenny's feedback.
* fix: Addresses more feedback.
* fix: Uses 'operational' as the default event type.
* doc: Updates the documentation.
* fix: Fixes adding of permanent properties.
* ref: Uses consistent naming for events' attributes.
Uses "_" as a separator instead of camel case or ".".
* feat: Adds the conference name as a permanent property automatically.
* ref: Don't expose Setting.machineId.
* fix: Adds a "p2p" attribute to jingle events.
* ref: Uses "action" instead of "name".
* ref: Uses underscore in events' attribute names.
* ref: Logs a message to the logger/console
instead of callstats in sendAnalyticsAndLog().
Reports ssrc to callstats when screen sharing is started. (#657)
Adds rtc.LOCAL_TRACK_SSRC_UPDATED event to be emitted when ssrc is updated for a local track.
Fixes adding rtc listeners for CREATE_ANSWER_FAILED, CREATE_OFFER_FAILED, SET_LOCAL_DESCRIPTION_FAILED, SET_REMOTE_DESCRIPTION_FAILED.
Saúl Ibarra Corretgé
Jaya Allamsetty
damencho
Gabriel Imre
Philipp Hancke
Russell Graves
Jaya Allamsetty
bgrozev
Hristo Terezov
Leonard Kim
Mark Madlangbayan
paweldomas
Saúl Ibarra Corretgé