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Removed unused code.

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R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org
2013-11-04 08:35:50 +00:00
parent e2df8b7f01
commit 766154aa1d
7 changed files with 1 additions and 43 deletions
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8
webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
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@ -78,10 +78,6 @@ public:
uint16_t* minRTT,
uint16_t* maxRTT) const;
uint16_t RTT() const;
int SetRTT(uint16_t rtt);
int32_t ResetRTT(const uint32_t remoteSSRC);
int32_t SenderInfoReceived(RTCPSenderInfo* senderInfo) const;
@ -266,10 +262,6 @@ protected:
// delivered RTP packet to the remote side.
int64_t _lastIncreasedSequenceNumberMs;
// Externally set RTT. This value can only be used if there are no valid
// RTT estimates.
uint16_t _rtt;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_