The original landed cl is in patchset 1. The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well. Original cl description: Let ViEEncoder handle resolution changes. This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder. With this change, many variables in WebRtcVideoSendStream no longer need to be locked. BUG=webrtc:5687, webrtc:6371, webrtc:5332 Review-Url: https://codereview.webrtc.org/2386573002 Cr-Commit-Position: refs/heads/master@{#14467}
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
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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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