/external/chromium-trace/catapult/telemetry/third_party/web-page-replay/ |
net_configs.py | 28 NetConfig = collections.namedtuple('NetConfig', ['down', 'up', 'delay_ms']) 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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trafficshaper.py | 59 delay_ms='0', 71 delay_ms: Propagation delay in milliseconds. '0' means no delay. 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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trafficshaper_test.py | 188 """Verify that it takes |delay_ms| to establish a TCP connection.""" 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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/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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/system/connectivity/dhcp_client/ |
event_dispatcher_interface.h | 31 int64_t delay_ms) = 0;
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message_loop_event_dispatcher.cc | 36 int64_t delay_ms) { 40 FROM_HERE, task, base::TimeDelta::FromMilliseconds(delay_ms));
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message_loop_event_dispatcher.h | 36 int64_t delay_ms) override;
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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/apmanager/ |
event_dispatcher.h | 37 int64_t delay_ms);
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mock_event_dispatcher.h | 33 int64_t delay_ms));
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event_dispatcher.cc | 47 int64_t delay_ms) { 51 FROM_HERE, task, base::TimeDelta::FromMilliseconds(delay_ms));
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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/libweave/src/ |
backoff_entry.cc | 129 // Note: if the failure count is too high, |delay_ms| will become infinity 133 double delay_ms = policy_->initial_delay_ms; local 134 delay_ms *= pow(policy_->multiply_factor, effective_failure_count - 1); 135 delay_ms -= base::RandDouble() * policy_->jitter_factor * delay_ms; 141 delay_ms + 0.5;
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/system/connectivity/shill/ |
mock_event_dispatcher.h | 36 int64_t delay_ms));
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event_dispatcher.cc | 51 void EventDispatcher::PostDelayedTask(const Closure& task, int64_t delay_ms) { 53 FROM_HERE, task, base::TimeDelta::FromMilliseconds(delay_ms));
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event_dispatcher.h | 52 virtual void PostDelayedTask(const base::Closure& task, int64_t delay_ms);
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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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/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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