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Removed unused code.

Browse Source 
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
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
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@ -418,12 +418,6 @@ class RtpRtcp : public Module {
*/
virtual int32_t ResetRTT(const uint32_t remoteSSRC)= 0 ;
/*
* Sets the estimated RTT, to be used for receive only modules without
* possibility of calculating its own RTT.
*/
virtual void SetRtt(uint32_t rtt) = 0;
/*
* Force a send of a RTCP packet
* normal SR and RR are triggered via the process function

1
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -145,7 +145,6 @@ class MockRtpRtcp : public RtpRtcp {
int32_t(const uint32_t remoteSSRC, uint16_t* RTT, uint16_t* avgRTT, uint16_t* minRTT, uint16_t* maxRTT));
MOCK_METHOD1(ResetRTT,
int32_t(const uint32_t remoteSSRC));
MOCK_METHOD1(SetRtt, void(uint32_t rtt));
MOCK_METHOD1(SendRTCP,
int32_t(uint32_t rtcpPacketType));
MOCK_METHOD1(SendRTCPReferencePictureSelection,

20
webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
View File

@ -54,8 +54,7 @@ RTCPReceiver::RTCPReceiver(const int32_t id, Clock* clock,
_receivedInfoMap(),
_packetTimeOutMS(0),
_lastReceivedRrMs(0),
_lastIncreasedSequenceNumberMs(0),
_rtt(0) {
_lastIncreasedSequenceNumberMs(0) {
memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
}
@ -228,23 +227,6 @@ bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) {
return true;
}
uint16_t RTCPReceiver::RTT() const {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return 0;
}
return _rtt;
}
int RTCPReceiver::SetRTT(uint16_t rtt) {
CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
if (!_receivedReportBlockMap.empty()) {
return -1;
}
_rtt = rtt;
return 0;
}
int32_t
RTCPReceiver::NTP(uint32_t *ReceivedNTPsecs,
uint32_t *ReceivedNTPfrac,

8
webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
View File

@ -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_

5
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -910,11 +910,6 @@ int32_t ModuleRtpRtcpImpl::ResetRTT(const uint32_t remote_ssrc) {
return rtcp_receiver_.ResetRTT(remote_ssrc);
}
void ModuleRtpRtcpImpl:: SetRtt(uint32_t rtt) {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SetRtt(rtt: %u)", rtt);
rtcp_receiver_.SetRTT(static_cast<uint16_t>(rtt));
}
// Reset RTP data counters for the sending side.
int32_t ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_,

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -171,8 +171,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// Reset RoundTripTime statistics.
virtual int32_t ResetRTT(const uint32_t remote_ssrc) OVERRIDE;
virtual void SetRtt(uint32_t rtt) OVERRIDE;
// Force a send of an RTCP packet.
// Normal SR and RR are triggered via the process function.
virtual int32_t SendRTCP(uint32_t rtcp_packet_type = kRtcpReport) OVERRIDE;

2
webrtc/video_engine/vie_channel.cc
View File

@ -1711,8 +1711,6 @@ bool ViEChannel::ChannelDecodeProcess() {
void ViEChannel::OnRttUpdate(uint32_t rtt) {
vcm_.SetReceiveChannelParameters(rtt);
if (!sender_)
rtp_rtcp_->SetRtt(rtt);
}
int32_t ViEChannel::StartDecodeThread() {