Update libjingle to 59676287
R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5390 4adac7df-926f-26a2-2b94-8c16560cd09d
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@@ -535,8 +535,10 @@ class MediaChannel : public sigslot::has_slots<> {
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const std::vector<RtpHeaderExtension>& extensions) = 0;
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virtual bool SetSendRtpHeaderExtensions(
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const std::vector<RtpHeaderExtension>& extensions) = 0;
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// Sets the rate control to use when sending data.
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virtual bool SetSendBandwidth(bool autobw, int bps) = 0;
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// Sets the initial bandwidth to use when sending starts.
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virtual bool SetStartSendBandwidth(int bps) = 0;
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// Sets the maximum allowed bandwidth to use when sending data.
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virtual bool SetMaxSendBandwidth(int bps) = 0;
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// Base method to send packet using NetworkInterface.
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bool SendPacket(talk_base::Buffer* packet) {
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