990d6b875e70e12d6230dc64d10b7287a66b3fbb
This reverts commit 90bace095806a635411edd40fb8490a144e59e63. Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it. Original change's description: > Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API > > (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) > > This SetAudioPlayout method lets applications disable audio playout while > still processing incoming audio data and generating statistics on the > received audio. > > This may be useful if the application wants to set up media flows as > soon as possible, but isn't ready to play audio yet. Currently, native > applications don't have any API point to control this, unless they > completely implement their own AudioDeviceModule. > > The SetAudioRecording works in a similar fashion but for the recorded > audio. One difference is that calling SetAudioRecording(false) does not > keep any audio processing alive. > > TBR=solenberg > > Bug: webrtc:7313 > Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa > Reviewed-on: https://webrtc-review.googlesource.com/16180 > Reviewed-by: Henrik Andreassson <henrika@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Commit-Queue: Henrik Andreassson <henrika@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20499} TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7313 Reviewed-on: https://webrtc-review.googlesource.com/17701 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20512}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
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