This is a reland of dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8 Original change's description: > Storing frame if encoder is paused. > > Adds a pending frame to VideoStreamEncoder that is used to store frames > that are not sent because the encoder is paused. If the encoder is > resumed within 200 ms, the pending frame will be encoded and sent. This > ensures that resuming a stream instantly starts sending frames if it is > possible. > > This also protects against a race between submitting the first frame > and enabling the encoder that caused flakiness in end to end tests > when using the task queue based congestion controller. > > Bug: webrtc:8415 > Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a > Reviewed-on: https://webrtc-review.googlesource.com/67141 > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22781} Bug: webrtc:8415 Change-Id: I0ea7d4d679e7845907cfbe9a120f128ff2180e4b Reviewed-on: https://webrtc-review.googlesource.com/68580 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22810}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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