| commit | 4c0536bd510a17c0d793ff3f811c25bee7f60fea | [log] [tgz] |
|---|---|---|
| author | skvlad <skvlad@webrtc.org> | Thu Jul 07 20:06:26 2016 |
| committer | skvlad <skvlad@webrtc.org> | Thu Jul 07 20:06:38 2016 |
| tree | 96f91a02bb401bfe050a105845f6a009173ecc4e | |
| parent | 44905cf49f53709cb4d0b9bdb62871d2988dbf0d [diff] |
Change GetPlayoutTimestamp message from Error to Info This message was printed to the receiver's log on every single video frame if the remote side is sending video but not audio. This could happen if, for example, one participant creates RtpSenders for both audio and video but doesn't set the audio track, or if the track is set but RtpParameters.encodings[0].active == false. This CL changes the trace level to StateInfo as it is the expected behavior in this case. R=deadbeef@webrtc.org, henrika@webrtc.org Review URL: https://codereview.webrtc.org/2125183002 . Cr-Commit-Position: refs/heads/master@{#13408}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.