| /external/webrtc/webrtc/modules/bitrate_controller/ |
| D | bitrate_controller_impl.cc | 35 int64_t rtt, in OnReceivedRtcpReceiverReport() argument
70 owner_->OnReceivedRtcpReceiverReport(fraction_lost_aggregate, rtt, in OnReceivedRtcpReceiverReport()
169 int64_t rtt, in OnReceivedRtcpReceiverReport() argument
174 bandwidth_estimation_.UpdateReceiverBlock(fraction_loss, rtt, in OnReceivedRtcpReceiverReport()
183 int64_t rtt; in MaybeTriggerOnNetworkChanged() local
184 if (GetNetworkParameters(&bitrate, &fraction_loss, &rtt)) in MaybeTriggerOnNetworkChanged()
185 observer_->OnNetworkChanged(bitrate, fraction_loss, rtt); in MaybeTriggerOnNetworkChanged()
190 int64_t* rtt) { in GetNetworkParameters() argument
193 bandwidth_estimation_.CurrentEstimate(¤t_bitrate, fraction_loss, rtt); in GetNetworkParameters()
201 *rtt != last_rtt_ms_ || in GetNetworkParameters()
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| D | send_side_bandwidth_estimation.cc | 95 int64_t* rtt) const { in CurrentEstimate()
98 *rtt = last_round_trip_time_ms_; in CurrentEstimate()
108 int64_t rtt, in UpdateReceiverBlock() argument
115 last_round_trip_time_ms_ = rtt; in UpdateReceiverBlock()
139 UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8); in UpdateReceiverBlock()
143 int64_t rtt, in UpdateUmaStats() argument
161 RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), in UpdateUmaStats()
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| D | bitrate_controller_impl.h | 56 int64_t rtt,
65 int64_t* rtt);
69 int64_t rtt) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
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| D | send_side_bandwidth_estimation.h | 30 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
40 int64_t rtt,
55 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
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| D | send_side_bandwidth_estimation_unittest.cc | 35 int64_t rtt; in TEST() local
36 bwe.CurrentEstimate(&bitrate, &fraction_loss, &rtt); in TEST()
44 bwe.CurrentEstimate(&bitrate, &fraction_loss, &rtt); in TEST()
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| /external/webrtc/webrtc/video/ |
| D | call_stats.cc | 41 max_rtt_ms = std::max(it->rtt, max_rtt_ms); in GetMaxRttMs()
53 sum += it->rtt; in GetAvgRttMs()
79 virtual void OnRttUpdate(int64_t rtt) { in OnRttUpdate() argument
80 owner_->OnRttUpdate(rtt); in OnRttUpdate()
163 void CallStats::OnRttUpdate(int64_t rtt) { in OnRttUpdate() argument
165 reports_.push_back(RttTime(rtt, clock_->TimeInMilliseconds())); in OnRttUpdate()
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| D | call_stats.h | 50 : rtt(new_rtt), time(rtt_time) {} in RttTime()
51 const int64_t rtt; member
56 void OnRttUpdate(int64_t rtt);
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| D | video_encoder.cc | 155 int64_t rtt) { in SetChannelParameters() argument
158 rtt_ = rtt; in SetChannelParameters()
159 int32_t ret = encoder_->SetChannelParameters(packet_loss, rtt); in SetChannelParameters()
161 return fallback_encoder_->SetChannelParameters(packet_loss, rtt); in SetChannelParameters()
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| D | vie_receiver.cc | 427 int64_t rtt = 0; in InsertRTCPPacket() local
428 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL); in InsertRTCPPacket()
429 if (rtt == 0) { in InsertRTCPPacket()
441 ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); in InsertRTCPPacket()
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_rtcp_impl.cc | 140 int64_t rtt = 0; in Process() local
141 rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL); in Process()
142 max_rtt = (rtt > max_rtt) ? rtt : max_rtt; in Process()
531 int64_t* rtt, in RTT() argument
535 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt); in RTT()
536 if (rtt && *rtt == 0) { in RTT()
538 *rtt = rtt_ms(); in RTT()
734 int64_t rtt = rtt_ms(); in TimeToSendFullNackList() local
735 if (rtt == 0) { in TimeToSendFullNackList()
736 rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL); in TimeToSendFullNackList()
[all …]
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| D | remote_ntp_time_estimator_unittest.cc | 57 void UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac, in UpdateRtcpTimestamp() argument
60 estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, in UpdateRtcpTimestamp()
64 void ReceiveRtcpSr(int64_t rtt, in ReceiveRtcpSr() argument
68 UpdateRtcpTimestamp(rtt, ntp_seconds, ntp_fractions, rtcp_timestamp, true); in ReceiveRtcpSr()
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| D | remote_ntp_time_estimator.cc | 31 bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt, in UpdateRtcpTimestamp() argument
48 int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90; in UpdateRtcpTimestamp()
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| D | rtcp_receiver_help.cc | 30 rtt(0), in RTCPPacketInformation()
99 this->rtt = report_block_info.RTT; in AddReportInfo()
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| /external/iputils/ |
| D | clockdiff.c | 117 long rtt = 1000; variable
210 long tmo = rtt + rtt_sigma; in measure()
242 rtt = (rtt * 3 + diff)/4; in measure()
243 rtt_sigma = (rtt_sigma *3 + abs(diff-rtt))/4; in measure()
390 long tmo = rtt + rtt_sigma; in measure_opt()
458 rtt = (rtt * 3 + diff)/4; in measure_opt()
459 rtt_sigma = (rtt_sigma *3 + abs(diff-rtt))/4; in measure_opt()
680 rtt, rtt_sigma, min_rtt, in main()
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| D | ping_common.c | 11 int rtt; variable
442 int est = rtt ? rtt/8 : interval*1000; in update_interval()
541 rtt_addend += (rtt < 8*50000 ? rtt/8 : 50000); in pinger()
906 if (!rtt) in gather_statistics()
907 rtt = triptime*8; in gather_statistics()
909 rtt += triptime-rtt/8; in gather_statistics()
1058 comma, ipg/1000, ipg%1000, rtt/8000, (rtt/8)%1000); in finish()
1083 rtt/8000, (rtt/8)%1000, in status()
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| /external/webrtc/webrtc/modules/video_coding/ |
| D | media_opt_util.cc | 83 if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs) { in ProtectionFactor()
89 } else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs) { in ProtectionFactor()
121 static_cast<int>(2.0f * base_layer_framerate * parameters->rtt / 1000.0f + in ComputeMaxFramesFec()
156 parameters->numLayers < 3 && parameters->rtt < kMaxRttTurnOffFec) { in BitRateTooLowForFec()
342 parameters->rtt, packetLoss); in ProtectionFactor()
518 void VCMLossProtectionLogic::UpdateRtt(int64_t rtt) { in UpdateRtt() argument
519 _rtt = rtt; in UpdateRtt()
639 _currentParameters.rtt = _rtt; in UpdateMethod()
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| D | media_opt_util.h | 49 : rtt(0), in VCMProtectionParameters()
62 int64_t rtt; member
241 void UpdateRtt(int64_t rtt);
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| D | video_coding_impl.cc | 129 int64_t rtt) override { in SetChannelParameters() argument
130 return sender_.SetChannelParameters(target_bitrate, lossRate, rtt); in SetChannelParameters()
272 int32_t SetReceiveChannelParameters(int64_t rtt) override { in SetReceiveChannelParameters() argument
273 return receiver_.SetReceiveChannelParameters(rtt); in SetReceiveChannelParameters()
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| /external/mesa3d/src/mesa/state_tracker/ |
| D | st_atom_framebuffer.c | 56 struct pipe_resource *resource = strb->rtt ? strb->rtt->pt : strb->texture; in update_renderbuffer_surface()
124 if (strb->rtt || in update_framebuffer_state()
147 if (strb->rtt) { in update_framebuffer_state()
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| /external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
| D | simulcast_encoder_adapter_unittest.cc | 133 MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
190 void ExpectCallSetChannelParameters(uint32_t packetLoss, int64_t rtt) { in ExpectCallSetChannelParameters() argument
194 SetChannelParameters(packetLoss, rtt)) in ExpectCallSetChannelParameters()
347 const int64_t rtt = 30; in TEST_F() local
348 helper_->ExpectCallSetChannelParameters(packetLoss, rtt); in TEST_F()
349 adapter_->SetChannelParameters(packetLoss, rtt); in TEST_F()
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| D | reference_picture_selection.cc | 118 void ReferencePictureSelection::SetRtt(int64_t rtt) { in SetRtt() argument
120 rtt_ = 90 * rtt; in SetRtt()
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| D | tester_main.cc | 22 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
60 args->rtt = FLAGS_rtt; in ParseArguments()
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| /external/iproute2/ip/ |
| D | tcp_metrics.c | 219 unsigned long rtt = 0, rttvar = 0; in process_msg() local
241 if (!rtt) in process_msg()
242 rtt = (val * 1000UL) >> 3; in process_msg()
249 rtt = val >> 3; in process_msg()
262 if (rtt) in process_msg()
263 fprintf(fp, " rtt %luus", rtt); in process_msg()
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| /external/webrtc/webrtc/ |
| D | video_encoder.h | 113 virtual int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) = 0;
150 int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
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| /external/webrtc/webrtc/p2p/base/ |
| D | port.cc | 69 inline uint32_t ConservativeRTTEstimate(uint32_t rtt) { in ConservativeRTTEstimate() argument
70 return std::max(MINIMUM_RTT, std::min(MAXIMUM_RTT, 2 * rtt)); in ConservativeRTTEstimate()
1030 uint32_t rtt = ConservativeRTTEstimate(rtt_); in UpdateState() local
1040 << ", rtt=" << rtt in UpdateState()
1056 rtt, in UpdateState()
1070 << " rtt=" << rtt; in UpdateState()
1081 << ", rtt=" << rtt; in UpdateState()
1210 uint32_t rtt = request->Elapsed(); in OnConnectionRequestResponse() local
1222 << ", rtt=" << rtt in OnConnectionRequestResponse()
1227 rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1); in OnConnectionRequestResponse()
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