| commit | b5eeeee13e6801f5f32444caebdd4b601fc96871 | [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 | 06a2bb53f3f4a52ded988cbce7e6f7da150ff150 | |
| parent | 05fb319d932116ccd7fe4d27d420e9694c560d69 [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-Original-Commit-Position: refs/heads/master@{#17591} Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc Cr-Mirrored-Commit: 292084c3765d9f3ee406ca2ec86eae206b540053