The point of this change is to make it possible to create the congestion controller as part of creating RtpTransportController, later pass it to the constructor of Call, and then let Call register itself as an observer. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2795643002 Cr-Commit-Position: refs/heads/master@{#17504}
Revert of CQ: Remove ARM64 trybots until toolchain issues are resolved. (patchset #1 id:1 of https://codereview.webrtc.org/2776393005/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Revert of PyLint fixes for tools-webrtc and webrtc/tools (patchset #3 id:40001 of https://codereview.webrtc.org/2736233003/ )
…
Revert of Adding PRESUBMIT check on google::protobuf (patchset #2 id:20001 of https://codereview.webrtc.org/2753823003/ )
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.
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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