| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| H A D | remote_ntp_time_estimator.cc | 33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL);
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| H A D | rtcp_receiver.cc | 181 reportBlock->RTT = 0;
188 int32_t RTCPReceiver::RTT(uint32_t remoteSSRC, argument
189 uint16_t* RTT,
201 if (RTT) {
202 *RTT = reportBlock->RTT;
487 // We can calc RTT if we send a send report and get a report block back.
550 // Estimate RTT
555 int32_t RTT = 0; local
558 RTT
[all...] |
| H A D | rtcp_receiver_help.cc | 99 this->rtt = report_block_info.RTT;
106 RTT(0),
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| H A D | remote_ntp_time_estimator_unittest.cc | 60 EXPECT_CALL(*rtp_rtcp, RTT(_, _, _, _, _))
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| H A D | rtcp_receiver_help.h | 36 // RTT
37 uint16_t RTT; member in class:webrtc::RTCPHelp::RTCPReportBlockInformation
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| H A D | rtcp_receiver.h | 75 int32_t RTT(uint32_t remoteSSRC,
76 uint16_t* RTT,
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| H A D | rtp_rtcp_impl.cc | 176 // Process RTT if we have received a receiver report and we haven't
177 // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
186 rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
768 int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc, function in class:webrtc::ModuleRtpRtcpImpl
773 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
775 // Try to get RTT from RtcpRttStats class.
930 // Use RTT from RtcpRttStats class if provided.
933 rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
936 int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5.
938 wait_time = 100; // During startup we don't have an RTT
[all...] |
| H A D | rtp_rtcp_impl_unittest.cc | 268 // Verify RTT.
274 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
280 // No RTT from other ssrc.
282 sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
284 // Verify RTT from rtt_stats config.
308 // Verify RTT.
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| H A D | rtp_rtcp_impl.h | 166 virtual int32_t RTT(const uint32_t remote_ssrc,
429 // The processed RTT from RtcpRttStats.
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| H A D | test_api_rtcp.cc | 344 uint16_t RTT; local
350 EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT));
351 EXPECT_GE(10, RTT);
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| H A D | rtp_rtcp.h | 423 virtual int32_t RTT(const uint32_t remoteSSRC,
424 uint16_t* RTT,
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| H A D | mock_rtp_rtcp.h | 163 MOCK_CONST_METHOD5(RTT,
164 int32_t(const uint32_t remoteSSRC, uint16_t* RTT, uint16_t* avgRTT, uint16_t* minRTT, uint16_t* maxRTT));
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| H A D | vie_receiver.cc | 417 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
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| H A D | vie_channel.cc | 1035 // have a chance of calculating an RTT we will try with the SSRC of the
1055 if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
1096 rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy);
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| H A D | channel.cc | 548 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
1872 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
3428 // --- RTT
4483 "GetRTPStatistics() RTCP is disabled => valid RTT "
4492 "GetRTPStatistics() failed to measure RTT since no "
4505 // To calculate RTT we try with the SSRC of the first report block.
4514 if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
4518 "GetRTPStatistics() failed to retrieve RTT from "
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| /external/iproute2/doc/ |
| H A D | ip-cref.tex | 1292 --- the initial RTT (``Round Trip Time'') estimate.
1297 --- \threeonly the initial RTT variance estimate.
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