ljm_j1

文件历史

Jaya Allamsetty 99feab14c0 fix(statistics) Use RTCRtpReceiver:getSynchronizationSources for audioLevels always. Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.		2 年前
..
RTC	fix(ssrc-rewriting) Update mute state after source is re-mapped. Also add all the orphan jvb tracks to the conference when the call switches over from p2p to jvb.	2 年前
browser	fix(statistics) Use RTCRtpReceiver:getSynchronizationSources for audioLevels always. Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.	2 年前
connectivity	fix(rn,ConnectionQuality) use new APIs for target bitrate on RN	2 年前
detection	fix(statistics) Use RTCRtpReceiver:getSynchronizationSources for audioLevels always. Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.	2 年前
e2ee	feat(e2ee) add ability to verify participants using a SAS mechanism	2 年前
e2eping	Plumbing for e2e pings. (#1973)	3 年前
event	chore(lint) tame the new linter	3 年前
flags	Add a feature flag and xmpp-caps entry for joining as a visitor. (#2247)	2 年前
litemode	Use insertable streams to drop all media when in lite mode. (#2178)	2 年前
proxyconnection	fix(SignalingLayer) Update SSRC owners on leave. (#2184)	2 年前
qualitycontrol	fix(quality-control) Fix receiver constraints for default case. When the application doesn't set receiver constraints, generate default constraints in the source-name mode. Cleanup code related to the endpoint based video constraint handling for p2p connection.	2 年前
recording	fix: Always set recroding status=on when a jibri is present (#2132)	3 年前
red	modules: add RFC 2198 audio redundancy encoder (#2067)	3 年前
sdp	feat: Add initial-last-n to jingle if config.channelLastN is set. (#2232)	2 年前
settings	chore(deps) adapt to logger package rename	3 年前
statistics	fix(statistics) Use RTCRtpReceiver:getSynchronizationSources for audioLevels always. Track based stats have been removed recently and RTCRtpReceiver:getSynchronizationSources has been available in the browsers for a while now.	2 年前
transcription	fix(ts) adjust syntax so TSC doesn't complain	3 年前
util	fix(ts) update jsdoc to appease typescript	3 年前
version	chore(deps) adapt to logger package rename	3 年前
videosipgw	fix(ts) clean-up TS tests where conversion is complete	3 年前
webaudio	chore(lint) tame the new linter	3 年前
xmpp	fix(ssrc-rewriting) Update mute state after source is re-mapped. Also add all the orphan jvb tracks to the conference when the call switches over from p2p to jvb.	2 年前