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Update talk to 61699344.

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TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5560 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mallinath@webrtc.org
2014-02-17 18:49:41 +00:00
parent e3842897e2
commit 92fdfebedd
5 changed files with 33 additions and 5 deletions
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4
talk/media/base/mediachannel.h
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@@ -539,6 +539,10 @@ class MediaChannel : public sigslot::has_slots<> {
const std::vector<RtpHeaderExtension>& extensions) = 0;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
// Returns the absoulte sendtime extension id value from media channel.
virtual int GetRtpSendTimeExtnId() const {
return -1;
}
// Sets the initial bandwidth to use when sending starts.
virtual bool SetStartSendBandwidth(int bps) = 0;
// Sets the maximum allowed bandwidth to use when sending data.