| commit | 292084c3765d9f3ee406ca2ec86eae206b540053 | [log] [tgz] |
|---|---|---|
| author | zhihuang <zhihuang@webrtc.org> | Fri Apr 07 17:57:22 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Apr 07 17:57:22 2017 |
| tree | ea822bdcaf5049bf528188d64e129275e9a0bccf | |
| parent | bb16a483f2eae187dadd38da4220cb2bba0259d5 [diff] |
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}
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.