/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
remote_ntp_time_estimator.cc | 32 uint16_t rtt = 0; local 33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL); 34 if (rtt == 0) { 35 // Waiting for valid rtt. 63 int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90;
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rtcp_receiver_help.h | 36 // RTT 37 uint16_t RTT; 71 uint16_t rtt; member in class:webrtc::RTCPHelp::RTCPPacketInformation
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rtcp_receiver.cc | 178 LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC; 181 reportBlock->RTT = 0; 188 int32_t RTCPReceiver::RTT(uint32_t remoteSSRC, 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 = 0 968 int32_t rtt = _clock->CurrentNtpInMilliseconds() - delay_rr_ms - send_time_ms; local [all...] |
rtcp_receiver_help.cc | 29 rtt(0), 99 this->rtt = report_block_info.RTT; 106 RTT(0),
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remote_ntp_time_estimator_unittest.cc | 60 EXPECT_CALL(*rtp_rtcp, RTT(_, _, _, _, _))
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rtcp_receiver.h | 74 // get rtt 75 int32_t RTT(uint32_t remoteSSRC, 76 uint16_t* RTT, 251 // Estimated rtt, zero when there is no valid estimate.
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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. 185 uint16_t rtt = 0; local 186 rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL); 187 max_rtt = (rtt > max_rtt) ? rtt : max_rtt; 189 // Report the rtt. 215 // Report rtt from receiver. 224 // Get processed rtt 931 uint16_t rtt = rtt_ms(); local 1278 uint16_t rtt = rtt_ms(); local [all...] |
rtp_rtcp_impl_unittest.cc | 253 TEST_F(RtpRtcpImplTest, Rtt) { 268 // Verify RTT. 269 uint16_t rtt; local 274 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt)); 275 EXPECT_EQ(2 * kOneWayNetworkDelayMs, 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 [all...] |
rtp_rtcp_impl.h | 166 virtual int32_t RTT(const uint32_t remote_ssrc, 167 uint16_t* rtt, 392 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt); 429 // The processed RTT from RtcpRttStats.
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
test_api_rtcp.cc | 344 uint16_t RTT; 350 EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT)); 351 EXPECT_GE(10, RTT);
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/prebuilts/gcc/linux-x86/host/x86_64-w64-mingw32-4.8/x86_64-w64-mingw32/include/ |
qos2.h | 60 UINT32 RTT;
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iphlpapi.h | 80 WINBOOL WINAPI GetRTTAndHopCount(IPAddr DestIpAddress,PULONG HopCount,ULONG MaxHops,PULONG RTT);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
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/ |
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/ |
vie_receiver.cc | 417 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
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vie_channel.cc | 95 virtual void OnRttUpdate(uint32_t rtt) { 96 owner_->OnRttUpdate(rtt); 1054 uint16_t rtt = 0; local 1095 uint16_t rtt = 0; local [all...] |
/external/chromium_org/remoting/webapp/ |
oauth2.js | 142 // Offset by a further 30 seconds to account for RTT issues.
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/external/chromium_org/third_party/webrtc/voice_engine/ |
channel.cc | 101 const uint32_t rtt) OVERRIDE { 103 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt); 548 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, 4510 uint16_t rtt = 0; local [all...] |
/external/chromium_org/third_party/WebKit/Source/devtools/front_end/toolbox/ |
OverridesUI.js | 174 var title = WebInspector.UIString("%s (%s %dms RTT)", preset.title, throughputText, latency);
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/external/iproute2/doc/ |
ip-cref.tex | [all...] |