3aa3ea7913
Adds a timeout to the dequeue input buffer call. This improves stability because WebRTC quickly queues frames multiple when the call starts. This might cause the decoder to run out of input buffers. Waiting for dequeueOutputBuffers call is no longer necessary. Bug: webrtc:7760 Change-Id: I503ff1cf44042c4d8610077090148d9dfef169f5 Reviewed-on: https://chromium-review.googlesource.com/548357 Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Commit-Queue: Sami Kalliomäki <sakal@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18800}
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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