/external/webrtc/webrtc/video/ |
D | stream_synchronization_unittest.cc | 331 int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { in MaxAudioDelayIncrease() argument 332 return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter, in MaxAudioDelayIncrease() 336 int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { in MaxAudioDelayDecrease() argument 337 return std::max((delay_ms - current_audio_delay_ms) / kSmoothingFilter, in MaxAudioDelayDecrease() 364 int delay_ms = 200; in TEST_F() local 368 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, in TEST_F() 372 EXPECT_EQ(delay_ms / kSmoothingFilter, total_video_delay_ms); in TEST_F() 378 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, in TEST_F() 382 EXPECT_EQ(2 * delay_ms / kSmoothingFilter, total_video_delay_ms); in TEST_F() 388 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, in TEST_F() [all …]
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D | receive_statistics_proxy.cc | 70 int delay_ms = delay_counter_.Avg(kMinRequiredDecodeSamples); in UpdateHistograms() local 71 if (delay_ms != -1) in UpdateHistograms() 72 RTC_HISTOGRAM_COUNTS_SPARSE_10000("WebRTC.Video.OnewayDelayInMs", delay_ms); in UpdateHistograms()
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/external/autotest/client/cros/cellular/wardmodem/ |
D | task_loop.py | 181 def post_repeated_task(self, callback, delay_ms=0): argument 209 next_delay_ms = self._next_delay_ms(delay_ms) 216 delay_ms) 220 def post_task_after_delay(self, callback, delay_ms, *args, **kwargs): argument 236 delay_ms = self._next_delay_ms(delay_ms) 237 self._posted_tasks[post_id] = glib.timeout_add(delay_ms, callback, 301 def _execute_repeated_task(self, post_id, callback, delay_ms): argument 318 next_delay_ms = self._next_delay_ms(delay_ms) 325 delay_ms)
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/external/chromium-trace/catapult/telemetry/third_party/webpagereplay/ |
D | net_configs.py | 33 'dialup': NetConfig(down= '49Kbit/s', up= '30Kbit/s', delay_ms= '120'), 34 '3g': NetConfig(down= '1638Kbit/s', up= '768Kbit/s', delay_ms= '150'), 35 'dsl': NetConfig(down= '1536Kbit/s', up= '384Kbit/s', delay_ms= '50'), 36 'cable': NetConfig(down= '5Mbit/s', up= '1Mbit/s', delay_ms= '28'), 37 'fios': NetConfig(down= '20Mbit/s', up= '5Mbit/s', delay_ms= '4'),
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D | trafficshaper_test.py | 193 for delay_ms in (100, 175): 194 with self.TrafficShaper(delay_ms=delay_ms): 198 self.assertValuesAlmostEqual(delay_ms, connect_time, tolerance=0.12) 253 for delay_ms in (100, 170): 254 expected_ms = delay_ms / 2 256 with self.TrafficShaper(delay_ms=delay_ms):
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D | trafficshaper.py | 59 delay_ms='0', argument 81 self.delay_ms = delay_ms 104 self.delay_ms == '0' and self.packet_loss_rate == '0'): 111 half_delay_ms = int(self.delay_ms) / 2 # split over up/down links
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D | dnsproxy.py | 157 def __init__(self, is_record_mode, delay_ms): argument 159 self.delay_ms = int(delay_ms) 163 time.sleep(self.delay_ms * 1000.0)
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/external/webrtc/tools/network_emulator/ |
D | config.py | 16 def __init__(self, num, name, receive_bw_kbps, send_bw_kbps, delay_ms, argument 22 self.delay_ms = delay_ms 36 self.queue_slots, self.delay_ms, self.packet_loss_percent)
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D | network_emulator.py | 65 self._connection_config.delay_ms, 71 self._connection_config.delay_ms, 127 def _create_dummynet_pipe(self, bandwidth_kbps, delay_ms, packet_loss_percent, argument 142 'delay', '%sms' % delay_ms,
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D | emulate.py | 84 default=_DEFAULT_PRESET.delay_ms, 148 if options.delay is not _DEFAULT_PRESET.delay_ms: 149 connection_config.delay_ms = options.delay 180 connection_config.delay_ms,
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | delay_manager.cc | 387 bool DelayManager::SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 391 if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) || in SetMinimumDelay() 393 delay_ms > in SetMinimumDelay() 397 minimum_delay_ms_ = delay_ms; in SetMinimumDelay() 401 bool DelayManager::SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 402 if (delay_ms == 0) { in SetMaximumDelay() 406 } else if (delay_ms < minimum_delay_ms_ || delay_ms < packet_len_ms_) { in SetMaximumDelay() 410 maximum_delay_ms_ = delay_ms; in SetMaximumDelay()
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D | delay_manager.h | 98 virtual bool SetMinimumDelay(int delay_ms); 99 virtual bool SetMaximumDelay(int delay_ms);
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/external/valgrind/drd/tests/ |
D | hold_lock.c | 14 static void delay_ms(const int ms) in delay_ms() function 52 delay_ms(interval); in main() 62 delay_ms(interval); in main() 70 delay_ms(interval); in main()
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/external/libbrillo/brillo/ |
D | backoff_entry.cc | 134 double delay_ms = policy_->initial_delay_ms; in CalculateReleaseTime() local 135 delay_ms *= pow(policy_->multiply_factor, effective_failure_count - 1); in CalculateReleaseTime() 136 delay_ms -= base::RandDouble() * policy_->jitter_factor * delay_ms; in CalculateReleaseTime() 142 delay_ms + 0.5; in CalculateReleaseTime()
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/external/libweave/src/ |
D | backoff_entry.cc | 133 double delay_ms = policy_->initial_delay_ms; in CalculateReleaseTime() local 134 delay_ms *= pow(policy_->multiply_factor, effective_failure_count - 1); in CalculateReleaseTime() 135 delay_ms -= base::RandDouble() * policy_->jitter_factor * delay_ms; in CalculateReleaseTime() 141 delay_ms + 0.5; in CalculateReleaseTime()
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
D | metric_recorder.cc | 166 void MetricRecorder::UpdateDelayMs(int64_t delay_ms) { in UpdateDelayMs() argument 167 PushDelayMs(delay_ms, now_ms_); in UpdateDelayMs() 168 plot_information_[kDelay].Update(now_ms_, delay_ms); in UpdateDelayMs() 195 void MetricRecorder::PushDelayMs(int64_t delay_ms, int64_t arrival_time_ms) { in PushDelayMs() argument 197 sum_delays_ms_ += delay_ms; in PushDelayMs() 198 sum_delays_square_ms2_ += delay_ms * delay_ms; in PushDelayMs() 199 if (delay_histogram_ms_.find(delay_ms) == delay_histogram_ms_.end()) { in PushDelayMs() 200 delay_histogram_ms_[delay_ms] = 0; in PushDelayMs() 202 ++delay_histogram_ms_[delay_ms]; in PushDelayMs()
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D | bwe_test_framework_unittest.cc | 302 void TestDelayFilter(int64_t delay_ms) { in TestDelayFilter() argument 303 filter_.SetOneWayDelayMs(delay_ms); in TestDelayFilter() 308 TestDelayFilter(delay_ms, 0, 0); in TestDelayFilter() 310 for (int i = 0; i < delay_ms; ++i) { in TestDelayFilter() 315 TestDelayFilter(delay_ms, 0, 0); in TestDelayFilter() 318 TestDelayFilter(delay_ms, 0, 0); in TestDelayFilter() 320 for (int i = 1; i < delay_ms + 1; ++i) { in TestDelayFilter() 325 filter_.SetOneWayDelayMs(2 * delay_ms); in TestDelayFilter() 327 TestDelayFilter(delay_ms, 13, 13); in TestDelayFilter() 328 TestDelayFilter(delay_ms, 0, 0); in TestDelayFilter() [all …]
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/external/webrtc/webrtc/base/ |
D | asyncinvoker.h | 88 uint32_t delay_ms, 92 DoInvokeDelayed(thread, closure, delay_ms, id); 139 uint32_t delay_ms, 178 uint32_t delay_ms, 183 invoker_.AsyncInvokeDelayed<ReturnT, FunctorT>(thread_, functor, delay_ms,
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/external/libchrome/base/message_loop/ |
D | message_pump_glib_unittest.cc | 70 void AddEvent(int delay_ms, const Closure& callback) { in AddEvent() argument 71 AddEventHelper(delay_ms, callback, Closure()); in AddEvent() 74 void AddDummyEvent(int delay_ms) { in AddDummyEvent() argument 75 AddEventHelper(delay_ms, Closure(), Closure()); in AddDummyEvent() 78 void AddEventAsTask(int delay_ms, const Closure& task) { in AddEventAsTask() argument 79 AddEventHelper(delay_ms, Closure(), task); in AddEventAsTask() 101 int delay_ms, const Closure& callback, const Closure& task) { in AddEventHelper() argument 108 Time future = last_time + TimeDelta::FromMilliseconds(delay_ms); in AddEventHelper()
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/external/webrtc/webrtc/modules/audio_device/android/ |
D | audio_device_template.h | 388 int32_t PlayoutDelay(uint16_t& delay_ms) const override { in PlayoutDelay() argument 390 delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2; in PlayoutDelay() 391 RTC_DCHECK_GT(delay_ms, 0); in PlayoutDelay() 395 int32_t RecordingDelay(uint16_t& delay_ms) const override { in RecordingDelay() argument 397 delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2; in RecordingDelay() 398 RTC_DCHECK_GT(delay_ms, 0); in RecordingDelay()
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/external/webrtc/webrtc/modules/audio_coding/test/ |
D | target_delay_unittest.cc | 178 int SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 179 return acm_->SetMinimumPlayoutDelay(delay_ms); in SetMinimumDelay() 182 int SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 183 return acm_->SetMaximumPlayoutDelay(delay_ms); in SetMaximumDelay()
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | acm_receiver.cc | 141 int AcmReceiver::SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 142 if (neteq_->SetMinimumDelay(delay_ms)) in SetMinimumDelay() 144 LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms; in SetMinimumDelay() 148 int AcmReceiver::SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 149 if (neteq_->SetMaximumDelay(delay_ms)) in SetMaximumDelay() 151 LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms; in SetMaximumDelay()
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D | acm_receiver.h | 134 int SetMinimumDelay(int delay_ms); 146 int SetMaximumDelay(int delay_ms);
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/external/webrtc/talk/app/webrtc/test/ |
D | fakeaudiocapturemodule_unittest.cc | 168 uint16_t delay_ms = 10; in TEST_F() local 169 EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms)); in TEST_F() 170 EXPECT_EQ(0, delay_ms); in TEST_F()
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/external/webrtc/webrtc/test/ |
D | fake_audio_device.cc | 72 int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const { in PlayoutDelay() 73 *delay_ms = 0; in PlayoutDelay()
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