![]() Simulcast video sends SDES with CNAME items with zero length PacedSender send to much padding when there are small packets sent Pass coded_size info along in webrtc::VideoFrameīandwidth toggles between two estimates in StartUpPhase. H264 constrained baseline fails to be decoded Make AEC3 the default AEC option in WebRTC Include pacing buffer size in congestion window pushback.Ĭreate FrameBufferController interface and allow its injection for VP8 Unify congestion window and pacing buffer pushbacks. Makes send packet information non optional for feedback reports. Simulcast streams will send one key frame on all spatial layers for each FIR with different SSRC ![]() The work for this was tracked in issue 10422 Turn urls are used to specify the list of ICE servers for a PeerConnection. This syntax was part of early internet-drafts, but was made obsolete by RFC 7065. Support for user-info part of turn urls, i.e., the sign preceded by username (and optional password) in turns:, has been deleted. M75 fixes this issue and makes the event useful by firing it when negotiation isĪctually needed, in accordance with the spec. This is not really a new feature, but prior to M75, theĮvent fired incorrectly. aĬreateOffer call followed by setLocalDescription). The negotiationneeded event informs the application that session negotiation needs to be done (i.e. New standardized stats have also been implemented, particularly ones meant to be able to replace non-standard “goog” stats from the legacy getStats() API. These metrics are also saved when you “Create Dump”, but if you want to view the old non-standard stats (returned from the callback-based API) there is a drop-down menu that lets you choose. chrome://webrtc-internals using standard getStats() + new stats implementedĬhrome://webrtc-internals now displays stats returned by the standardized getStats() API, which is the promise-based API in Chrome.Information about the state of the underlying ICE transport and DTLS transport. The WebRTC APIs for RTPsenders and RTPReceivers have been extended with attributes that give RTCIceTransport and RTCDtlsTransport APIs.Native libraries for Android and iOS are built on a weekly basis and are available on JCenter and CocoaPods the Changelog is available here. The Chrome release schedule can be found here. ![]() ![]() The help we have received has been invaluable! Please take a look at this page, for some pointers on how to file a good bug report. As with previous releases, we encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found. WebRTC M75, currently available in Chrome's beta channel a n d a s n a t i v e l i b r a r i e s f o r A n d r o i d a n d i O S, contains 3 new features and over 50 bug fixes, enhancements and stability/performance improvements. WebRTC M75 branch (cut at r 27678 ) Summary ![]()
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