Changes to the mixer unittests: Removed the tests related to the former 'OutputMixer', as it's going to be removed. Removed incorrect comparison tests with the old mixer because doing identical mixing decisions with the old mixer proved unviable. When the new mixer went from kMaximumAmountOfMixedAudioSources in the last iteration to kMaximumAmountOfMixedAudioSources+1, it could hit an RTC_NOTREACHED(); Added fix to mixer and test AudioMixer.RampedOutSourcesShouldNotBeMarkedMixed that covers that case. NOTRY=True Review-Url: https://codereview.webrtc.org/2253153004 Cr-Commit-Position: refs/heads/master@{#13880}
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
Revert of Add field_trial_default dependency to libjingle_peerconnection (patchset #3 id:40001 of https://codereview.webrtc.org/2120673004/ )
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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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