3ed36c0521546881656c73984456485dcab16205

stopped_
from AudioRtpReceiver and VideoRtpReceiver."
This is a reland of bb57e2d7aa9b36843233d1394422f03d12d9c31f The difference from the original CL is that a check for `state_ == kLive` inside of RemoteAudioSource::AddSink has been removed. This caused a side effect that registering the sink while the source was in an "initializing" state, failed. The last remaining state however, is `kEnded` - but since there's no logic in the class around the expected value of the states, the check inside of AddSink() doesn't provide an additional value - it's rather a surprise for developers if it doesn't succeed. So, now removed. Original change's description: > Remove `stopped_` from AudioRtpReceiver and VideoRtpReceiver. > > This simplifies the logic in these classes a bit, which makes upcoming > change easier. The `stopped_` flag in these classes was essentially > the same thing as `media_channel_ == nullptr`, which is what's > consistently used now for the same checks. > > Bug: webrtc:13540 > Change-Id: Ib60bfad9f28d5ddee8a8d5170c3f2a7ef017a5ca > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250163 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35907} Bug: webrtc:13540 Change-Id: I3e5b3046fae11cb56b50c38c5f08972a6f283dd5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251326 Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35958}
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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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation
Description
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