/external/webrtc/call/ |
D | fake_network_pipe.cc | 98 last_log_time_us_(clock_->TimeInMicroseconds()) {} in FakeNetworkPipe() 112 last_log_time_us_(clock_->TimeInMicroseconds()) { in FakeNetworkPipe() 202 int64_t time_now_us = clock_->TimeInMicroseconds(); in EnqueuePacket() 213 int64_t time_now_us = clock_->TimeInMicroseconds(); in EnqueuePacket() 268 time_now_us = clock_->TimeInMicroseconds(); in Process() 360 int64_t delay_us = *delivery_us - clock_->TimeInMicroseconds(); in TimeUntilNextProcess() 384 return clock_->TimeInMicroseconds(); in GetTimeInMicroseconds()
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D | call.cc | 1332 packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds()); in DeliverRtp()
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/external/webrtc/system_wrappers/include/ |
D | clock.h | 37 return Timestamp::Micros(TimeInMicroseconds()); in CurrentTime() 40 virtual int64_t TimeInMicroseconds() { return CurrentTime().us(); } in TimeInMicroseconds() function
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/external/webrtc/modules/audio_processing/test/ |
D | performance_timer.cc | 30 start_timestamp_us_ = clock_->TimeInMicroseconds(); in StartTimer() 35 timestamps_us_.push_back(clock_->TimeInMicroseconds() - *start_timestamp_us_); in StopTimer()
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/external/webrtc/modules/video_coding/ |
D | receiver_unittest.cc | 265 int64_t start_time = TimeInMicroseconds(); in AdvanceTimeMicroseconds() 273 TimeInMicroseconds()); in AdvanceTimeMicroseconds() 282 if (TimeInMicroseconds() < end_time) { in AdvanceTimeMicroseconds() 283 SimulatedClock::AdvanceTimeMicroseconds(end_time - TimeInMicroseconds()); in AdvanceTimeMicroseconds()
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D | jitter_estimator.cc | 206 _latestNackTimestamp = clock_->TimeInMicroseconds(); in FrameNacked() 298 uint64_t now = clock_->TimeInMicroseconds(); in EstimateRandomJitter() 385 uint64_t now = clock_->TimeInMicroseconds(); in GetJitterEstimate()
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/external/webrtc/modules/congestion_controller/goog_cc/ |
D | delay_based_bwe_unittest_helper.cc | 157 clock_.TimeInMicroseconds())), in DelayBasedBweTest() 171 clock_.TimeInMicroseconds())), in DelayBasedBweTest() 232 stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds()); in GenerateAndProcessFrame() 239 clock_.TimeInMicroseconds()); in GenerateAndProcessFrame() 266 clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds()); in GenerateAndProcessFrame()
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/external/webrtc/rtc_tools/rtc_event_log_visualizer/ |
D | analyzer.cc | 1204 cc_config.constraints.at_time = Timestamp::Micros(clock.TimeInMicroseconds()); in CreateSendSideBweSimulationGraph() 1260 clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds()); in CreateSendSideBweSimulationGraph() 1261 if (clock.TimeInMicroseconds() >= NextRtpTime()) { in CreateSendSideBweSimulationGraph() 1262 RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime()); in CreateSendSideBweSimulationGraph() 1301 if (clock.TimeInMicroseconds() >= NextRtcpTime()) { in CreateSendSideBweSimulationGraph() 1302 RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime()); in CreateSendSideBweSimulationGraph() 1324 float x = config_.GetCallTimeSec(clock.TimeInMicroseconds()); in CreateSendSideBweSimulationGraph() 1337 if (clock.TimeInMicroseconds() >= NextProcessTime()) { in CreateSendSideBweSimulationGraph() 1338 RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextProcessTime()); in CreateSendSideBweSimulationGraph() 1340 msg.at_time = Timestamp::Micros(clock.TimeInMicroseconds()); in CreateSendSideBweSimulationGraph() [all …]
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/external/webrtc/modules/audio_processing/ |
D | audio_processing_performance_unittest.cc | 271 const int64_t start_time = clock_->TimeInMicroseconds(); in ProcessCapture() 275 const int64_t end_time = clock_->TimeInMicroseconds(); in ProcessCapture() 299 const int64_t start_time = clock_->TimeInMicroseconds(); in ProcessRender() 303 const int64_t end_time = clock_->TimeInMicroseconds(); in ProcessRender()
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/external/webrtc/modules/remote_bitrate_estimator/ |
D | remote_bitrate_estimator_unittest_helper.cc | 204 clock_.TimeInMicroseconds())), in RemoteBitrateEstimatorTest() 257 stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds()); in GenerateAndProcessFrame() 265 clock_.TimeInMicroseconds()); in GenerateAndProcessFrame() 278 clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds()); in GenerateAndProcessFrame()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | remote_ntp_time_estimator_unittest.cc | 47 return TimeMicrosToNtp(remote_clock_.TimeInMicroseconds()); in GetRemoteNtpTime()
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D | flexfec_sender.cc | 80 random_(clock_->TimeInMicroseconds()), in FlexfecSender()
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D | rtcp_receiver.cc | 367 CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds())); in ConsumeReceivedXrReferenceTimeInfo() 526 last_received_sr_ntp_ = TimeMicrosToNtp(clock_->TimeInMicroseconds()); in HandleSenderReport() 819 CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds())); in HandleXrReceiveReferenceTime() 853 uint32_t now_ntp = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds())); in HandleXrDlrrReportBlock()
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D | rtcp_receiver_unittest.cc | 257 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST() 289 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST() 320 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST() 874 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST() 898 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST() 976 NtpTime now = TimeMicrosToNtp(mocks.clock.TimeInMicroseconds()); in TEST() 1000 NtpTime now = TimeMicrosToNtp(mocks.clock.TimeInMicroseconds()); in TEST() 1410 CompactNtp(TimeMicrosToNtp(mocks.clock.TimeInMicroseconds())); in TEST()
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D | rtcp_sender.cc | 155 random_(clock_->TimeInMicroseconds()), in RTCPSender() 768 clock_->TimeInMicroseconds()); in ComputeCompoundRTCPPacket() 878 uint32_t now = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds())); in CreateReportBlocks()
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D | rtp_packet_history_unittest.cc | 572 fake_clock_.TimeInMicroseconds()); in TEST_P() 601 hist_.PutRtpPacket(std::move(packet), fake_clock_.TimeInMicroseconds()); in TEST_P() 621 fake_clock_.TimeInMicroseconds()); in TEST_P()
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D | rtcp_sender_unittest.cc | 148 NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds()); in TEST_F() 498 NtpTime ntp = TimeMicrosToNtp(clock_.TimeInMicroseconds()); in TEST_F()
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D | rtp_sender.cc | 160 random_(clock_->TimeInMicroseconds()), in RTPSender()
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/external/webrtc/system_wrappers/source/ |
D | clock.cc | 44 int64_t TimeInMicroseconds() override { return rtc::TimeMicros(); } in TimeInMicroseconds() function in webrtc::RealTimeClock
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/external/webrtc/test/ |
D | frame_generator_capturer.cc | 192 .set_timestamp_us(clock_->TimeInMicroseconds()) in InsertFrame()
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/external/webrtc/video/adaptation/ |
D | video_stream_encoder_resource_manager.cc | 414 clock_->TimeInMicroseconds()); in SetStartBitrate()
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/external/webrtc/video/ |
D | video_stream_encoder.cc | 940 int64_t current_time_us = clock_->TimeInMicroseconds(); in OnFrame()
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