commit | 5f6bf245066ccceecf208ecfefb7a603b15658f7 | [log] [tgz] |
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author | henrika <henrika@webrtc.org> | Wed Nov 01 11:06:56 2017 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Nov 01 11:04:26 2017 +0000 |
tree | 7e3d3b1fc6c2de81fcca3c06b6e42f98c0b1bb38 | |
parent | 0e1db676407de6decbaa1fdbf3c2df701c6fd701 [diff] |
Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) Second attempt to land https://webrtc-review.googlesource.com/c/src/+/16180 Now removes voice_engine dependency from peerconnection and fixes a minor const issue in NullAudioPoller. TBR=solenberg Bug: webrtc:7313 Change-Id: Ibfddbdc76118581e4a4dc64575203f84c1659e5c Reviewed-on: https://webrtc-review.googlesource.com/17784 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20526}
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.