This should remove the test flakiness, as before this change there could be collisions from sequence numbers coming from two sequence number spaces (the media SSRC and the FlexFEC SSRC). The probability of collisions was low, and hence the flakes were far between. This change also reduces the packet loss to 5% (down from ~50%), in order for the BWE to have an easier time to ramp up. BUG=webrtc:6825 R=philipel@webrtc.org, mflodman@webrtc.org Review-Url: https://codereview.webrtc.org/2554403003 Cr-Commit-Position: refs/heads/master@{#15512}
Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ )
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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