commit | 612445ea6038c4746b8696b47447445b82b99c7a | [log] [tgz] |
---|---|---|
author | Tomas Gunnarsson <tommi@webrtc.org> | Mon Sep 21 12:31:23 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Sep 21 13:29:53 2020 |
tree | 8531d17f553eda73fcbf2c78eea6d9ad1aa0f75a | |
parent | 023e1ac7bc8e0039cc2db472031c1bbe89e80ac8 [diff] |
Remove use of asyncinvoker from WebRtcVideoSendStream. This turned out to be a bit complicated, mostly related to the tests, but here's what's changed: * No AsyncInvoker (and avoid ClearInternal) in WebRtcVideoSendStream (WVSS) * The reason it was there is due to a "design leak" from VideoSourceSinkController/VideoStreamEncoder where the former uses locks in all methods and is unaware of a threading model. That design affected downstream objects, pushed the need for an async hop into WVSS and added a lock. A suggestion was made to address this in a follow-up change, here: https://webrtc-review.googlesource.com/c/src/+/165684 * All methods in VideoSourceSinkController are now called on a known and checked sequence and this CL removes the lock. This also makes checking state consistent (i.e. calling a getter twice in a row on the same sequence, will always return the same value, avoiding race with other threads). * Handling of reporting state changes from the encoder queue to the VSSC, is done by VideoStreamEncoder. * VideoSendStreamImpl is still instantiated on the incorrect thread [1] but has two initialization steps [2]. The second one already runs on the right thread. Addressing that TODO [1] is something we should do but it has side effects to consider. For the purposes of this CL the steps relating to the encoder (setting the sink pointer) have been moved to [2]. [1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;l=94 [2] https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/video/video_send_stream.cc;drc=f4a9991cce74a37d006438ec0e366313ed33162e;l=115 Bug: webrtc:11222, webrtc:11908 Change-Id: Ie46d46e3a52bbe225951b4bd580ecb8cc9cad873 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184508 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32150}
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.