commit | 00cc836fcfa031a16d9c62375d5aa490519c3ac6 | [log] [tgz] |
---|---|---|
author | Erik Språng <sprang@webrtc.org> | Mon Nov 25 11:21:46 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Nov 25 12:13:38 2019 |
tree | 9bd84ab5c5e9752d441595aadd1947acf8037d3c | |
parent | 912b3b83b380fbcf608d5b9ad15c6aed99f3b065 [diff] |
Makes all of RtpVideoSenderTest use simulated time RtpVideoSenderTest used a SimulatedClock but the task queue factor still looked at the real-time clock when posting delayed tasks. This CL changes that so everything is using simulated time, which makes test faster and should avoid flakiness. In particular, fixing this timing issue exposed flaws in DoesNotRetrasmitAckedPackets, which was likely the root case of bug 10873, so let's re-enable on ios again. Bug: webrtc:10873,webrtc:10809 Change-Id: If8a0c244b1a34f7427543deaa2431ab1e9f124a6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160404 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29897}
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.
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.