/external/webrtc/tools/network_emulator/ |
config.py | 16 def __init__(self, num, name, receive_bw_kbps, send_bw_kbps, delay_ms, 22 self.delay_ms = delay_ms 36 self.queue_slots, self.delay_ms, self.packet_loss_percent)
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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, 133 delay_ms: Delay for a one-way trip of a packet. 142 'delay', '%sms' % delay_ms,
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/external/valgrind/drd/tests/ |
hold_lock.c | 14 static void delay_ms(const int ms) function 52 delay_ms(interval); 62 delay_ms(interval); 70 delay_ms(interval);
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/system/connectivity/wificond/ |
event_loop.h | 40 // |delay_ms| is delay time in milliseconds. It should not be negative. 43 int64_t delay_ms) = 0;
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looper_backed_event_loop.h | 38 int64_t delay_ms) override;
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looper_backed_event_loop.cpp | 87 int64_t delay_ms) { 89 looper_->sendMessageDelayed(ms2ns(delay_ms), looper_callback, NULL);
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/external/autotest/client/cros/cellular/wardmodem/ |
task_loop.py | 181 def post_repeated_task(self, callback, delay_ms=0): 198 @param delay_ms: The delay between repeated calls to |callback|. The 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): 222 Post the given callback once to be dispatched after |delay_ms|. 228 @param delay_ms: The delay before the call to |callback|. Default: 0 236 delay_ms = self._next_delay_ms(delay_ms) 237 self._posted_tasks[post_id] = glib.timeout_add(delay_ms, callback [all...] |
/external/webrtc/webrtc/video/ |
stream_synchronization_unittest.cc | 331 int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { 332 return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter, 336 int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { 337 return std::max((delay_ms - current_audio_delay_ms) / kSmoothingFilter, 364 int delay_ms = 200; local 368 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, 372 EXPECT_EQ(delay_ms / kSmoothingFilter, total_video_delay_ms); 378 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, 382 EXPECT_EQ(2 * delay_ms / kSmoothingFilter, total_video_delay_ms); 388 EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms 396 int delay_ms = 200; local [all...] |
/external/webrtc/webrtc/voice_engine/include/ |
voe_video_sync.h | 63 // jitter buffer delay is max of |delay_ms| and the latency that NetEq 65 virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
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/external/libbrillo/brillo/ |
backoff_entry.cc | 130 // Note: if the failure count is too high, |delay_ms| will become infinity 134 double delay_ms = policy_->initial_delay_ms; local 135 delay_ms *= pow(policy_->multiply_factor, effective_failure_count - 1); 136 delay_ms -= base::RandDouble() * policy_->jitter_factor * delay_ms; 142 delay_ms + 0.5;
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
delay_manager.cc | 387 bool DelayManager::SetMinimumDelay(int delay_ms) { 391 if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) || 393 delay_ms > 397 minimum_delay_ms_ = delay_ms; 401 bool DelayManager::SetMaximumDelay(int delay_ms) { 402 if (delay_ms == 0) { 406 } else if (delay_ms < minimum_delay_ms_ || delay_ms < packet_len_ms_) { 410 maximum_delay_ms_ = delay_ms;
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delay_manager.h | 98 virtual bool SetMinimumDelay(int delay_ms); 99 virtual bool SetMaximumDelay(int delay_ms);
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/external/webrtc/webrtc/base/ |
asyncinvoker.h | 83 // Call |functor| asynchronously on |thread| with |delay_ms|, with no callback 88 uint32_t delay_ms, 92 DoInvokeDelayed(thread, closure, delay_ms, id); 139 uint32_t delay_ms, 174 // Call |functor| asynchronously with |delay_ms|, with no callback upon 178 uint32_t delay_ms, 183 invoker_.AsyncInvokeDelayed<ReturnT, FunctorT>(thread_, functor, delay_ms,
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asyncinvoker.cc | 70 uint32_t delay_ms, 76 thread->PostDelayed(delay_ms, this, id,
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/external/webrtc/webrtc/modules/audio_coding/test/ |
target_delay_unittest.cc | 178 int SetMinimumDelay(int delay_ms) { 179 return acm_->SetMinimumPlayoutDelay(delay_ms); 182 int SetMaximumDelay(int delay_ms) { 183 return acm_->SetMaximumPlayoutDelay(delay_ms);
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/external/webrtc/webrtc/test/ |
fake_audio_device.h | 40 int32_t PlayoutDelay(uint16_t* delay_ms) const override;
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fake_audio_device.cc | 72 int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const { 73 *delay_ms = 0;
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
acm_receiver.h | 129 // - delay_ms : minimum delay in milliseconds. 134 int SetMinimumDelay(int delay_ms); 141 // - delay_ms : maximum delay in milliseconds. 146 int SetMaximumDelay(int delay_ms);
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/frameworks/av/services/audiopolicy/service/ |
AudioPolicyClientImpl.cpp | 128 int delay_ms) 131 delay_ms); 146 int delay_ms) 148 mAudioPolicyService->setParameters(io_handle, keyValuePairs.string(), delay_ms); 169 status_t AudioPolicyService::AudioPolicyClient::setVoiceVolume(float volume, int delay_ms) 171 return mAudioPolicyService->setVoiceVolume(volume, delay_ms);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
metric_recorder.cc | 166 void MetricRecorder::UpdateDelayMs(int64_t delay_ms) { 167 PushDelayMs(delay_ms, now_ms_); 168 plot_information_[kDelay].Update(now_ms_, delay_ms); 195 void MetricRecorder::PushDelayMs(int64_t delay_ms, int64_t arrival_time_ms) { 197 sum_delays_ms_ += delay_ms; 198 sum_delays_square_ms2_ += delay_ms * delay_ms; 199 if (delay_histogram_ms_.find(delay_ms) == delay_histogram_ms_.end()) { 200 delay_histogram_ms_[delay_ms] = 0; 202 ++delay_histogram_ms_[delay_ms]; [all...] |
/external/webrtc/talk/app/webrtc/test/ |
fakeaudiocapturemodule_unittest.cc | 168 uint16_t delay_ms = 10; local 169 EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms)); 170 EXPECT_EQ(0, delay_ms);
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/external/webrtc/webrtc/modules/audio_device/android/ |
audio_device_template.h | 388 int32_t PlayoutDelay(uint16_t& delay_ms) const override { 390 delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2; 391 RTC_DCHECK_GT(delay_ms, 0); 395 int32_t RecordingDelay(uint16_t& delay_ms) const override { 397 delay_ms = audio_manager_->GetDelayEstimateInMilliseconds() / 2; 398 RTC_DCHECK_GT(delay_ms, 0);
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/external/libchrome/base/message_loop/ |
message_pump_glib_unittest.cc | 69 // delay_ms is relative to the last event if any, or to Now() otherwise. 70 void AddEvent(int delay_ms, const Closure& callback) { 71 AddEventHelper(delay_ms, callback, Closure()); 74 void AddDummyEvent(int delay_ms) { 75 AddEventHelper(delay_ms, Closure(), Closure()); 78 void AddEventAsTask(int delay_ms, const Closure& task) { 79 AddEventHelper(delay_ms, Closure(), task); 101 int delay_ms, const Closure& callback, const Closure& task) { 108 Time future = last_time + TimeDelta::FromMilliseconds(delay_ms);
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/external/webrtc/webrtc/modules/audio_coding/neteq/include/ |
neteq.h | 206 virtual bool SetMinimumDelay(int delay_ms) = 0; 212 virtual bool SetMaximumDelay(int delay_ms) = 0;
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/external/v8/src/heap/ |
memory-reducer.h | 126 void ScheduleTimer(double time_ms, double delay_ms);
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