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
This commit is contained in:
sergeyu@chromium.org
2014-01-15 23:15:54 +00:00
parent 7a2ca7c621
commit 4b26e2eee3
47 changed files with 1203 additions and 718 deletions

View File

@@ -535,8 +535,10 @@ class MediaChannel : public sigslot::has_slots<> {
const std::vector<RtpHeaderExtension>& extensions) = 0;
virtual bool SetSendRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) = 0;
// Sets the rate control to use when sending data.
virtual bool SetSendBandwidth(bool autobw, int bps) = 0;
// 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.
virtual bool SetMaxSendBandwidth(int bps) = 0;
// Base method to send packet using NetworkInterface.
bool SendPacket(talk_base::Buffer* packet) {