commit | 6b143c1c0686519bc9d73223c1350cee286c8d78 | [log] [tgz] |
---|---|---|
author | Guido Urdaneta <guidou@webrtc.org> | Tue Feb 09 12:21:02 2021 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 09 12:27:32 2021 |
tree | 0ee4df4ed034ac08fe35f66d92c405c20eebfe82 | |
parent | 6e4fcac31312f2dda5b60d33874ff0cd62f94321 [diff] |
Revert "Fix unsynchronized access to mid_to_transport_ in JsepTransportController" This reverts commit 6cd405850467683cf10d05028ea0f644a68a91a4. Reason for revert: Breaks WebRTC Chromium FYI Bots First failure: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Android%20Tests%20%28dbg%29%20%28L%20Nexus5%29/1925 Failed tests: WebRtcDataBrowserTest.CallWithSctpDataAndMedia WebRtcDataBrowserTest.CallWithSctpDataOnly Original change's description: > Fix unsynchronized access to mid_to_transport_ in JsepTransportController > > * Added several thread checks to JTC to help with programmer errors. > * Avoid a few Invokes() to the network thread here and there such > as for fetching sctp transport name for getStats(). The transport > name is now cached when it changes on the network thread. > * JsepTransportController instances now get deleted on the network > thread rather than on the signaling thread + issuing an Invoke() > in the dtor. > * Moved some thread hops from JTC over to PC which is where the problem > exists and also (imho) makes it easier to see where hops happen in > the PC code. > * The sctp transport is now started asynchronously when we push down the > media description. > * PeerConnection proxy calls GetSctpTransport directly on the network > thread instead of to the signaling thread + blocking on the network > thread. > * The above changes simplified things for webrtc::SctpTransport which > allowed for removing locking from that class and delete some code. > > Bug: webrtc:9987, webrtc:12445 > Change-Id: Ic89a9426e314e1b93c81751d4f732f05fa448fbc > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205620 > Commit-Queue: Tommi <tommi@webrtc.org> > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33191} TBR=tommi@webrtc.org,hta@webrtc.org Change-Id: I7b2913d5133807589461105cf07eff3e9bb7157e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9987 Bug: webrtc:12445 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206466 Reviewed-by: Guido Urdaneta <guidou@webrtc.org> Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33204}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.