/external/webrtc/modules/audio_coding/neteq/ |
D | delay_manager.cc | 365 bool DelayManager::IsValidMinimumDelay(int delay_ms) const { in IsValidMinimumDelay() 366 return 0 <= delay_ms && delay_ms <= MinimumDelayUpperBound(); in IsValidMinimumDelay() 369 bool DelayManager::IsValidBaseMinimumDelay(int delay_ms) const { in IsValidBaseMinimumDelay() 370 return kMinBaseMinimumDelayMs <= delay_ms && in IsValidBaseMinimumDelay() 371 delay_ms <= kMaxBaseMinimumDelayMs; in IsValidBaseMinimumDelay() 374 bool DelayManager::SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 375 if (!IsValidMinimumDelay(delay_ms)) { in SetMinimumDelay() 379 minimum_delay_ms_ = delay_ms; in SetMinimumDelay() 384 bool DelayManager::SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 387 if (delay_ms != 0 && in SetMaximumDelay() [all …]
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D | delay_manager.h | 95 virtual bool SetMinimumDelay(int delay_ms); 96 virtual bool SetMaximumDelay(int delay_ms); 97 virtual bool SetBaseMinimumDelay(int delay_ms); 140 bool IsValidMinimumDelay(int delay_ms) const; 142 bool IsValidBaseMinimumDelay(int delay_ms) const;
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D | decision_logic.h | 87 bool SetMaximumDelay(int delay_ms) override { in SetMaximumDelay() argument 88 return delay_manager_->SetMaximumDelay(delay_ms); in SetMaximumDelay() 90 bool SetMinimumDelay(int delay_ms) override { in SetMinimumDelay() argument 91 return delay_manager_->SetMinimumDelay(delay_ms); in SetMinimumDelay() 93 bool SetBaseMinimumDelay(int delay_ms) override { in SetBaseMinimumDelay() argument 94 return delay_manager_->SetBaseMinimumDelay(delay_ms); in SetBaseMinimumDelay()
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/external/webrtc/tools_webrtc/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 _CreateDummynetPipe(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/pc/ |
D | jitter_buffer_delay.cc | 57 int delay_ms = in Set() local 59 delay_ms = rtc::SafeClamp(delay_ms, 0, kMaximumDelayMs); in Set() 63 media_channel_->SetBaseMinimumPlayoutDelayMs(ssrc_.value(), delay_ms); in Set()
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/external/webrtc/test/time_controller/ |
D | simulated_thread.cc | 61 int delay_ms = GetDelay(); in RunReady() local 63 if (delay_ms == kForever) { in RunReady() 66 next_run_time_ = at_time + TimeDelta::Millis(delay_ms); in RunReady() 103 int delay_ms, in PostDelayed() argument 107 rtc::Thread::PostDelayed(posted_from, delay_ms, phandler, id, pdata); in PostDelayed() 110 std::min(next_run_time_, Timestamp::Millis(rtc::TimeMillis() + delay_ms)); in PostDelayed()
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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/webrtc/modules/audio_coding/acm2/ |
D | acm_receiver.cc | 66 int AcmReceiver::SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 67 if (neteq_->SetMinimumDelay(delay_ms)) in SetMinimumDelay() 69 RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms; in SetMinimumDelay() 73 int AcmReceiver::SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 74 if (neteq_->SetMaximumDelay(delay_ms)) in SetMaximumDelay() 76 RTC_LOG(LERROR) << "AcmReceiver::SetExtraDelay " << delay_ms; in SetMaximumDelay() 80 bool AcmReceiver::SetBaseMinimumDelayMs(int delay_ms) { in SetBaseMinimumDelayMs() argument 81 return neteq_->SetBaseMinimumDelayMs(delay_ms); in SetBaseMinimumDelayMs()
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D | acm_receiver.h | 97 int SetMinimumDelay(int delay_ms); 109 int SetMaximumDelay(int delay_ms); 116 bool SetBaseMinimumDelayMs(int delay_ms);
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/external/tensorflow/tensorflow/python/profiler/integration_test/ |
D | profiler_api_test.py | 91 def test_single_worker_sampling_mode(self, delay_ms=None): argument 113 delay_ms=delay_ms, 140 self.test_single_worker_sampling_mode(delay_ms=1) 145 self.test_single_worker_sampling_mode(delay_ms=1000)
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/external/webrtc/modules/audio_coding/test/ |
D | target_delay_unittest.cc | 122 int SetMinimumDelay(int delay_ms) { in SetMinimumDelay() argument 123 return receiver_.SetMinimumDelay(delay_ms); in SetMinimumDelay() 126 int SetMaximumDelay(int delay_ms) { in SetMaximumDelay() argument 127 return receiver_.SetMaximumDelay(delay_ms); in SetMaximumDelay()
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/external/perfetto/test/ |
D | android_test_utils.cc | 94 uint32_t delay_ms) { in WaitForProcess() argument 102 delay_ms); in WaitForProcess() 109 uint32_t delay_ms) { in StartAppActivity() argument 113 WaitForProcess(app_name, checkpoint_name, task_runner, delay_ms); in StartAppActivity()
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D | android_test_utils.h | 37 uint32_t delay_ms = 1); 43 uint32_t delay_ms = 1);
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/external/webrtc/modules/audio_coding/neteq/mock/ |
D | mock_neteq_controller.h | 35 MOCK_METHOD(bool, SetMaximumDelay, (int delay_ms), (override)); 36 MOCK_METHOD(bool, SetMinimumDelay, (int delay_ms), (override)); 37 MOCK_METHOD(bool, SetBaseMinimumDelay, (int delay_ms), (override));
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/external/webrtc/rtc_base/ |
D | async_invoker.h | 114 uint32_t delay_ms, 119 DoInvokeDelayed(posted_from, thread, std::move(closure), delay_ms, id); 143 uint32_t delay_ms,
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/external/webrtc/api/neteq/ |
D | neteq_controller.h | 128 virtual bool SetMaximumDelay(int delay_ms) = 0; 129 virtual bool SetMinimumDelay(int delay_ms) = 0; 135 virtual bool SetBaseMinimumDelay(int delay_ms) = 0;
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/external/oboe/apps/fxlab/app/src/main/cpp/effects/ |
D | SlapbackEffect.h | 23 SlapbackEffect(float feedforward, float delay_ms): in SlapbackEffect() argument 25 static_cast<int>(delay_ms * SAMPLE_RATE / 1000), in SlapbackEffect()
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D | EchoEffect.h | 26 EchoEffect(float feedback, float delay_ms): in EchoEffect() argument 28 static_cast<int>(delay_ms * SAMPLE_RATE / 1000), in EchoEffect()
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D | WhiteChorusEffect.h | 25 WhiteChorusEffect(float depth_ms, float delay_ms, float noise_pass): in WhiteChorusEffect() argument 27 static_cast<int>(delay_ms * SAMPLE_RATE / 1000), in WhiteChorusEffect()
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D | DoublingEffect.h | 26 DoublingEffect(float depth_ms, float delay_ms, float noise_pass): in DoublingEffect() argument 28 static_cast<int>(delay_ms * SAMPLE_RATE / 1000), in DoublingEffect()
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/external/libchrome/base/message_loop/ |
D | message_pump_glib_unittest.cc | 74 void AddEvent(int delay_ms, OnceClosure callback) { in AddEvent() argument 75 AddEventHelper(delay_ms, std::move(callback), OnceClosure()); in AddEvent() 78 void AddDummyEvent(int delay_ms) { in AddDummyEvent() argument 79 AddEventHelper(delay_ms, OnceClosure(), OnceClosure()); in AddDummyEvent() 82 void AddEventAsTask(int delay_ms, OnceClosure task) { in AddEventAsTask() argument 83 AddEventHelper(delay_ms, OnceClosure(), std::move(task)); in AddEventAsTask() 104 void AddEventHelper(int delay_ms, OnceClosure callback, OnceClosure task) { in AddEventHelper() argument 111 Time future = last_time + TimeDelta::FromMilliseconds(delay_ms); in AddEventHelper()
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/external/webrtc/api/audio/ |
D | echo_control.h | 42 int delay_ms; member 49 virtual void SetAudioBufferDelay(int delay_ms) = 0;
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/external/webrtc/audio/ |
D | channel_receive.cc | 145 bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override; 825 void ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) { in SetMinimumPlayoutDelay() argument 829 delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs, in SetMinimumPlayoutDelay() 831 if (acm_receiver_.SetMinimumDelay(delay_ms) != 0) { in SetMinimumPlayoutDelay() 869 bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) { in SetBaseMinimumPlayoutDelayMs() argument 870 return acm_receiver_.SetBaseMinimumDelayMs(delay_ms); in SetBaseMinimumPlayoutDelayMs() 905 uint16_t delay_ms = 0; in UpdatePlayoutTimestamp() local 906 if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) { in UpdatePlayoutTimestamp() 917 playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000)); in UpdatePlayoutTimestamp() 925 playout_delay_ms_ = delay_ms; in UpdatePlayoutTimestamp()
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