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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.