ba156cfe96
Contains fixes for a non-perfect implementation in https://codereview.webrtc.org/2328433003/ Summary: Adds WebRTC.Audio.RecordedOnlyZeros UMA stat when recording stops if: - All level estimates during the audio session were zero, and - If the audio session was longer than 10 seconds. Adds four simple methods to the AudioDeviceBuffer (ADB) class to allow the ADM to update the ADB about when media starts and stops in both directions. Moves any "critical" parst out frome the timer (based on task queue) and ensures that it only does trivial logging tasks. The task queue is now owned by a unique pointer to improve control of when it starts and stops. Adds time measurements (for logging) of both total time playing out and total recording time. Units are in milliseconds. BUG=webrtc:6592 Review-Url: https://codereview.webrtc.org/2445363003 Cr-Commit-Position: refs/heads/master@{#14854}
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. This page is maintained by the Google Chrome team.
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