| /external/webrtc/video/ |
| D | call_stats2.h | 65 const int64_t rtt; member
85 void OnRttUpdate(int64_t rtt) override { in OnRttUpdate()
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| D | call_stats.h | 58 const int64_t rtt; member
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| D | call_stats2.cc | 131 void CallStats::OnRttUpdate(int64_t rtt) { in OnRttUpdate()
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| D | call_stats.cc | 185 void CallStats::OnRttUpdate(int64_t rtt) { in OnRttUpdate()
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| D | call_stats2_unittest.cc | 46 void AsyncSimulateRttUpdate(int64_t rtt) { in AsyncSimulateRttUpdate()
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| D | call_stats_unittest.cc | 50 void AsyncSimulateRttUpdate(int64_t rtt) { in AsyncSimulateRttUpdate()
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| /external/grpc-grpc-java/interop-testing/src/test/java/io/grpc/testing/integration/ |
| D | ProxyTest.java | 87 long rtt = (stop - start); in smallLatency() local
117 long rtt = (stop - start); in bigLatency() local
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | remote_ntp_time_estimator_unittest.cc | 70 void UpdateRtcpTimestamp(int64_t rtt, in UpdateRtcpTimestamp()
79 void ReceiveRtcpSr(int64_t rtt, in ReceiveRtcpSr()
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| D | rtp_rtcp_impl2.cc | 449 int64_t* rtt, in RTT()
585 int64_t rtt = rtt_ms(); in TimeToSendFullNackList() local
667 int64_t rtt = rtt_ms(); in OnReceivedNack() local
745 absl::optional<TimeDelta> rtt = in PeriodicUpdate() local
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| D | remote_ntp_time_estimator.cc | 38 bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt, in UpdateRtcpTimestamp()
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| D | rtp_rtcp_impl.cc | 132 int64_t rtt = 0; in Process() local
497 int64_t* rtt, in RTT()
681 int64_t rtt = rtt_ms(); in TimeToSendFullNackList() local
760 int64_t rtt = rtt_ms(); in OnReceivedNack() local
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| /external/webrtc/p2p/base/ |
| D | connection.cc | 141 inline int ConservativeRTTEstimate(int rtt) { in ConservativeRTTEstimate()
786 int rtt = ConservativeRTTEstimate(rtt_); in UpdateState() local
882 const int64_t rtt = rtc::TimeMillis() - iter->sent_time; in HandlePiggybackCheckAcknowledgementIfAny() local
889 int rtt, in ReceivedPingResponse()
1078 int rtt = request->Elapsed(); in OnConnectionRequestResponse() local
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| D | connection_info.h | 43 size_t rtt; // The STUN RTT for this connection. member
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| /external/webrtc/modules/congestion_controller/pcc/ |
| D | rtt_tracker_unittest.cc | 23 PacketResult GetPacketWithRtt(TimeDelta rtt) { in GetPacketWithRtt()
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| /external/rust/crates/quiche/src/recovery/ |
| D | cubic.rs | 562 let rtt = Duration::from_millis(100); in cubic_congestion_avoidance() localVariable
763 let rtt = Duration::from_millis(100); in cubic_spurious_congestion_event() localVariable
808 let rtt = Duration::from_millis(100); in cubic_fast_convergence() localVariable
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| D | reno.rs | 328 let rtt = Duration::from_millis(100); in reno_congestion_avoidance() localVariable
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| /external/libnl/lib/idiag/ |
| D | idiag_vegasinfo_obj.c | 70 void idiagnl_vegasinfo_set_rtt(struct idiagnl_vegasinfo *vinfo, uint32_t rtt) in idiagnl_vegasinfo_set_rtt()
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| /external/webrtc/audio/ |
| D | channel_send.cc | 289 int64_t rtt, in OnReceivedRtcpReceiverReport()
636 int64_t rtt = GetRTT(); in ReceivedRTCPPacket() local
871 int64_t rtt = 0; in GetRTT() local
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| D | channel_receive.cc | 664 int64_t rtt = GetRTT(); in ReceivedRTCPPacket() local
958 int64_t rtt = 0; in GetRTT() local
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| /external/webrtc/audio/voip/ |
| D | audio_ingress.cc | 173 int64_t rtt = GetRoundTripTime(); in ReceivedRTCPPacket() local
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | loss_based_bandwidth_estimation.cc | 26 double GetIncreaseFactor(const LossBasedControlConfig& config, TimeDelta rtt) { in GetIncreaseFactor()
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| /external/webrtc/rtc_tools/rtc_event_log_visualizer/ |
| D | log_simulation.cc | 145 TimeDelta rtt = TimeDelta::PlusInfinity(); in OnReceiverReport() local
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| /external/iproute2/ip/ |
| D | tcp_metrics.c | 216 unsigned long rtt = 0, rttvar = 0; in process_msg() local
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| /external/libffi/src/sparc/ |
| D | ffi.c | 51 int rtt = rtype->type; in ffi_prep_cif_machdep() local
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| /external/webrtc/modules/video_coding/ |
| D | nack_module_unittest.cc | 187 TimeDelta rtt = TimeDelta::Millis(160); // + (i * 10 - 40) in TEST_P() local
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