commit | 990d6b875e70e12d6230dc64d10b7287a66b3fbb | [log] [tgz] |
---|---|---|
author | Mirko Bonadei <mbonadei@webrtc.org> | Wed Nov 01 02:40:35 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Nov 01 02:40:48 2017 |
tree | 55223e715e680794b6ad2954d96a7f8dd8ea9c80 | |
parent | e4be4b7b996cc7fb279d74b5001ab0817b22b6c4 [diff] |
Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" 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}
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.