| /external/webrtc/video/end_to_end_tests/ |
| D | probing_tests.cc | 78 int64_t start_time_ms = clock_->TimeInMilliseconds(); local
80 if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
130 int64_t start_time_ms = clock_->TimeInMilliseconds(); local
132 if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
234 int64_t start_time_ms = clock_->TimeInMilliseconds(); local
237 if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
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| D | stats_tests.cc | 756 int64_t start_time_ms = clock_->TimeInMilliseconds(); in TEST_F() local
761 ASSERT_GE(start_time_ms + kDefaultTimeoutMs, clock_->TimeInMilliseconds()) in TEST_F()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | neteq_test.cc | 102 const int64_t start_time_ms = *input_->NextEventTime(); in RunToNextGetAudio() local
103 int64_t time_now_ms = start_time_ms; in RunToNextGetAudio()
199 input_->NextEventTime().value_or(time_now_ms) - start_time_ms; in RunToNextGetAudio()
247 start_time_ms) in RunToNextGetAudio()
277 input_->NextEventTime().value_or(time_now_ms) - start_time_ms; in RunToNextGetAudio()
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| D | neteq_performance_test.cc | 79 int64_t start_time_ms = clock->TimeInMilliseconds(); in Run() local
124 return end_time_ms - start_time_ms; in Run()
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| /external/webrtc/modules/audio_coding/codecs/opus/ |
| D | opus_complexity_unittest.cc | 39 const int64_t start_time_ms = rtc::TimeMillis(); in RunComplexityTest() local
48 return rtc::TimeMillis() - start_time_ms; in RunComplexityTest()
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| /external/webrtc/modules/video_coding/ |
| D | receiver.cc | 80 const int64_t start_time_ms = clock_->TimeInMilliseconds(); in FrameForDecoding() local
140 static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms); in FrameForDecoding()
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| /external/libwebm/ |
| D | sample_muxer_metadata.cc | 171 const time_ms_t start_time_ms = cue.start_time.presentation(); in AddChapter() local
175 const uint64_t start_time_ns = start_time_ms * kNsPerMs; in AddChapter()
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| /external/webrtc/call/ |
| D | call_perf_tests.cc | 83 int start_time_ms,
370 int start_time_ms, in TestCaptureNtpTime() argument
377 int start_time_ms, in TestCaptureNtpTime() argument
383 start_time_ms_(start_time_ms), in TestCaptureNtpTime()
505 } test(net_config, threshold_ms, start_time_ms, run_time_ms); in TestCaptureNtpTime()
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | goog_cc_network_control.cc | 205 absl::optional<int64_t> start_time_ms = in OnProcessInterval() local
207 probe_controller_->SetAlrStartTimeMs(start_time_ms); in OnProcessInterval()
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| D | goog_cc_network_control_unittest.cc | 215 int64_t start_time_ms = current_time_.ms(); in PacketTransmissionAndFeedbackBlock() local
216 while (current_time_.ms() - start_time_ms < runtime_ms) { in PacketTransmissionAndFeedbackBlock()
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_packet_history_unittest.cc | 769 const int64_t start_time_ms = fake_clock_.TimeInMilliseconds(); in TEST_P() local
788 start_time_ms + expected_time_offset_ms); in TEST_P()
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| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_event_log_parser.h | 340 int64_t start_time_ms() const { return start_time_us_ / 1000; } in RTC_POP_IGNORING_WUNDEF()
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| D | rtc_event_log_unittest.cc | 742 EXPECT_EQ(parsed_log.first_log_segment().start_time_ms(), in ReadAndVerifyLog()
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| /external/webrtc/video/ |
| D | video_stream_encoder_unittest.cc | 3802 const int64_t start_time_ms = rtc::TimeMillis(); in TEST_F() local
3803 while (rtc::TimeMillis() - start_time_ms < kProcessIntervalMs) { in TEST_F()
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