commit | fbcc5cb3869d1370008e40f24fc03ac8fb69c675 | [log] [tgz] |
---|---|---|
author | olka <olka@webrtc.org> | Mon Apr 10 04:38:13 2017 -0700 |
committer | Commit bot <commit-bot@chromium.org> | Mon Apr 10 11:38:13 2017 +0000 |
tree | 25b95e5bef9dfc92441d1307c236b7bf7990222a | |
parent | 925e9d762c107230fe96eb0e8c2bb4dab87417fe [diff] |
Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) Reason for revert: Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Added the GetSources() to the RtpReceiverInterface and implemented > it for the AudioRtpReceiver. > > This method returns a vector of RtpSource(both CSRC source and SSRC > source) which contains the ID of a source, the timestamp, the source > type (SSRC or CSRC) and the audio level. > > The RtpSource objects are buffered and maintained by the > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > the info of the contributing source will be pulled along the object > chain: > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > BUG=chromium:703122 > TBR=stefan@webrtc.org, danilchap@webrtc.org > > Review-Url: https://codereview.webrtc.org/2770233003 > Cr-Commit-Position: refs/heads/master@{#17591} > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053 TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2809613002 Cr-Commit-Position: refs/heads/master@{#17616}
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.