| /external/webrtc/rtc_base/ |
| D | rate_statistics_unittest.cc | 30 int64_t now_ms = 0; in TEST_F() local 73 int64_t now_ms = 0; in TEST_F() local 126 int64_t now_ms = 0; in TEST_F() local 170 int64_t now_ms = 0; in TEST_F() local 204 int64_t now_ms = 0; in TEST_F() local 235 int64_t now_ms = 0; in TEST_F() local 259 int64_t now_ms = 0; in TEST_F() local 284 int64_t now_ms = 0; in TEST_F() local 293 int64_t now_ms = 0; in TEST_F() local 302 int64_t now_ms = 0; in TEST_F() local
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| D | rate_statistics.cc | 57 void RateStatistics::Update(int64_t count, int64_t now_ms) { in Update() 125 void RateStatistics::EraseOld(int64_t now_ms) { in EraseOld() 142 bool RateStatistics::SetWindowSize(int64_t window_size_ms, int64_t now_ms) { in SetWindowSize()
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| D | rate_limiter.cc | 35 int64_t now_ms = clock_->TimeInMilliseconds(); in TryUseRate() local
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| /external/webrtc/modules/remote_bitrate_estimator/ |
| D | remote_bitrate_estimator_abs_send_time_unittest.cc | 115 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 138 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 160 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 189 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 208 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 226 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 246 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 265 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local
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| D | overuse_detector.cc | 87 int64_t now_ms) { in Detect() 128 void OveruseDetector::UpdateThreshold(double modified_offset, int64_t now_ms) { in UpdateThreshold()
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| D | remote_bitrate_estimator_abs_send_time.cc | 180 RemoteBitrateEstimatorAbsSendTime::ProcessClusters(int64_t now_ms) { in ProcessClusters() 257 int64_t now_ms = clock_->TimeInMilliseconds(); in IncomingPacketInfo() local 373 void RemoteBitrateEstimatorAbsSendTime::TimeoutStreams(int64_t now_ms) { in TimeoutStreams()
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | send_side_bandwidth_estimation_unittest.cc | 40 int64_t now_ms = 0; in TestProbing() local 94 int64_t now_ms = 1000; in TEST() local 148 int64_t now_ms = 0; in TEST() local
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| D | delay_based_bwe_unittest.cc | 30 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 52 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 69 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 84 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local 103 int64_t now_ms = clock_.TimeInMilliseconds(); in TEST_F() local
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| D | trendline_estimator.cc | 272 void TrendlineEstimator::Detect(double trend, double ts_delta, int64_t now_ms) { in Detect() 314 int64_t now_ms) { in UpdateThreshold()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | neteq_quality_test.cc | 252 bool NoLoss::Lost(int now_ms) { in Lost() 258 bool UniformLoss::Lost(int now_ms) { in Lost() 271 bool GilbertElliotLoss::Lost(int now_ms) { in Lost() 293 bool FixedLossModel::Lost(int now_ms) { in Lost()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | source_tracker.cc | 27 int64_t now_ms = clock_->TimeInMilliseconds(); in OnFrameDelivered() local 56 int64_t now_ms = clock_->TimeInMilliseconds(); in GetSources() local
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| D | rtp_sender_egress.cc | 147 const int64_t now_ms = clock_->TimeInMilliseconds(); in SendPacket() local 299 const int64_t now_ms = clock_->TimeInMilliseconds(); in GetSendRates() local 424 int64_t now_ms, in UpdateDelayStatistics() 539 void RtpSenderEgress::UpdateRtpStats(int64_t now_ms, in UpdateRtpStats() 597 void RtpSenderEgress::BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc) { in BweTestLoggingPlot()
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| D | rtp_packet_history.cc | 126 int64_t now_ms = clock_->TimeInMilliseconds(); in PutRtpPacket() local 183 int64_t now_ms = clock_->TimeInMilliseconds(); in GetPacketAndSetSendTime() local 396 void RtpPacketHistory::CullOldPackets(int64_t now_ms) { in CullOldPackets()
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| /external/webrtc/modules/rtp_rtcp/source/deprecated/ |
| D | deprecated_rtp_sender_egress.cc | 99 int64_t now_ms = clock_->TimeInMilliseconds(); in SendPacket() local 224 const int64_t now_ms = clock_->TimeInMilliseconds(); in GetSendRatesLocked() local 328 int64_t now_ms, in UpdateDelayStatistics() 445 int64_t now_ms = clock_->TimeInMilliseconds(); in UpdateRtpStats() local
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| /external/webrtc/video/ |
| D | rtp_streams_synchronizer2.cc | 89 const int64_t now_ms = rtc::TimeMillis(); in UpdateDelay() local 190 int64_t now_ms = rtc::TimeMillis(); in GetStreamSyncOffsetInMs() local
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| D | rtp_streams_synchronizer.cc | 86 const int64_t now_ms = rtc::TimeMillis(); in Process() local 187 int64_t now_ms = rtc::TimeMillis(); in GetStreamSyncOffsetInMs() local
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| D | quality_limitation_reason_tracker.cc | 35 int64_t now_ms = clock_->TimeInMilliseconds(); in SetReason() local
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| D | send_statistics_proxy.cc | 224 void SendStatisticsProxy::UmaSamplesContainer::RemoveOld(int64_t now_ms) { in RemoveOld() 256 int64_t now_ms = clock_->TimeInMilliseconds(); in InsertEncodedFrame() local 699 int64_t now_ms = clock_->TimeInMilliseconds(); in OnSuspendChange() local 858 const int64_t now_ms = clock_->TimeInMilliseconds(); in UpdateEncoderFallbackStats() local 1323 int64_t now_ms = clock_->TimeInMilliseconds(); in DataCountersUpdated() local 1388 void SendStatisticsProxy::StatsTimer::Start(int64_t now_ms) { in Start() 1393 void SendStatisticsProxy::StatsTimer::Stop(int64_t now_ms) { in Stop() 1400 void SendStatisticsProxy::StatsTimer::Restart(int64_t now_ms) { in Restart()
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| D | receive_statistics_proxy.cc | 629 int64_t now_ms = clock_->TimeInMilliseconds(); in GetStats() local 698 int64_t now_ms = clock_->TimeInMilliseconds(); in OnTimingFrameInfoUpdated() local 738 uint64_t now_ms = clock_->TimeInMilliseconds(); in OnDecodedFrame() local 801 int64_t now_ms = clock_->TimeInMilliseconds(); in OnRenderedFrame() local 878 int64_t now_ms = clock_->TimeInMilliseconds(); in OnCompleteFrame() local
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| D | video_receive_stream2.cc | 628 int64_t now_ms = clock_->TimeInMilliseconds(); in StartNextDecode() local 643 int64_t now_ms = clock_->TimeInMilliseconds(); in HandleEncodedFrame() local 703 int64_t now_ms, in HandleKeyFrameGeneration() 731 void VideoReceiveStream2::HandleFrameBufferTimeout(int64_t now_ms, in HandleFrameBufferTimeout()
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| /external/webrtc/modules/video_coding/ |
| D | timing.cc | 155 int64_t now_ms, in StopDecodeTimer() argument 160 void VCMTiming::StopDecodeTimer(int32_t decode_time_ms, int64_t now_ms) { in StopDecodeTimer() 167 void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) { in IncomingTimestamp()
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| D | codec_timer.cc | 32 void VCMCodecTimer::AddTiming(int64_t decode_time_ms, int64_t now_ms) { in AddTiming()
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| D | frame_buffer2.cc | 126 int64_t FrameBuffer::FindNextFrame(int64_t now_ms) { in FindNextFrame() 222 int64_t now_ms = clock_->TimeInMilliseconds(); in GetNextFrame() local 302 int64_t now_ms) { in HasBadRenderTiming() 622 int64_t now_ms = clock_->TimeInMilliseconds(); in UpdateFrameInfoWithIncomingFrame() local
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| /external/webrtc/common_audio/ |
| D | smoothing_filter.cc | 43 const int64_t now_ms = rtc::TimeMillis(); in AddSample() local
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| /external/webrtc/rtc_base/experiments/ |
| D | quality_rampup_experiment.cc | 55 bool QualityRampupExperiment::BwHigh(int64_t now_ms, in BwHigh()
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