/external/webrtc/src/system_wrappers/source/ |
D | cpu_measurement_harness.cc | 72 int elapsed_time_ms = 0; in Run() local 74 while (elapsed_time_ms < duration_ms_) { in Run() 75 if (((elapsed_time_ms - last_measured_time) / kCpuCheckPeriodMs) >= 1) { in Run() 76 last_measured_time = elapsed_time_ms; in Run() 83 elapsed_time_ms += work_period_ms_; in Run()
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/external/chromium_org/third_party/webrtc/modules/pacing/ |
D | paced_sender.cc | 239 int64_t elapsed_time_ms = in TimeUntilNextProcess() local 241 if (elapsed_time_ms <= 0) { in TimeUntilNextProcess() 244 if (elapsed_time_ms >= kMinPacketLimitMs) { in TimeUntilNextProcess() 247 return kMinPacketLimitMs - elapsed_time_ms; in TimeUntilNextProcess() 253 int elapsed_time_ms = (now - time_last_update_).Milliseconds(); in Process() local 259 if (elapsed_time_ms > 0) { in Process() 260 uint32_t delta_time_ms = std::min(kMaxIntervalTimeMs, elapsed_time_ms); in Process()
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/external/chromium_org/third_party/webrtc/voice_engine/ |
D | voe_base_impl.h | 83 int64_t* elapsed_time_ms, 108 int64_t* elapsed_time_ms, 146 int64_t* elapsed_time_ms,
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D | voe_base_impl.cc | 152 int64_t* elapsed_time_ms, in NeedMorePlayData() argument 163 elapsed_time_ms, ntp_time_ms); in NeedMorePlayData() 240 int64_t* elapsed_time_ms, in PullRenderData() argument 246 audio_data, elapsed_time_ms, ntp_time_ms); in PullRenderData() 1090 int64_t* elapsed_time_ms, in GetPlayoutData() argument 1113 *elapsed_time_ms = _audioFrame.elapsed_time_ms_; in GetPlayoutData()
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/external/chromium_org/third_party/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 74 int64_t* elapsed_time_ms, 131 int64_t* elapsed_time_ms, in PullRenderData() argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
D | delay_manager.cc | 340 void DelayManager::UpdateCounters(int elapsed_time_ms) { in UpdateCounters() argument 341 packet_iat_count_ms_ += elapsed_time_ms; in UpdateCounters() 342 peak_detector_.IncrementCounter(elapsed_time_ms); in UpdateCounters() 343 max_timer_ms_ += elapsed_time_ms; in UpdateCounters()
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D | decision_logic.cc | 158 const int elapsed_time_ms = output_size_samples_ / (8 * fs_mult_); in FilterBufferLevel() local 159 delay_manager_->UpdateCounters(elapsed_time_ms); in FilterBufferLevel()
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D | delay_manager.h | 80 virtual void UpdateCounters(int elapsed_time_ms);
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/external/chromium_org/content/renderer/media/ |
D | webrtc_audio_device_impl.cc | 154 int64_t elapsed_time_ms = -1; in RenderData() local 167 &elapsed_time_ms, in RenderData() 179 &elapsed_time_ms, in RenderData() 183 if (elapsed_time_ms >= 0) { in RenderData() 184 *current_time = base::TimeDelta::FromMilliseconds(elapsed_time_ms); in RenderData()
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/external/chromium_org/content/browser/download/ |
D | download_stats.cc | 597 size_t elapsed_time_ms = elapsed_time.InMilliseconds(); in RecordFileBandwidth() local 598 if (0u == elapsed_time_ms) in RecordFileBandwidth() 599 elapsed_time_ms = 1; in RecordFileBandwidth() 606 (1000 * length / elapsed_time_ms), 1, 50000000, 50); in RecordFileBandwidth() 611 disk_write_time_ms * 100 / elapsed_time_ms); in RecordFileBandwidth()
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/external/chromium_org/third_party/webrtc/test/ |
D | fake_audio_device.cc | 124 int64_t elapsed_time_ms = -1; in CaptureAudio() local 133 &elapsed_time_ms, in CaptureAudio()
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/external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
D | func_test_manager.h | 122 int64_t* elapsed_time_ms, 144 int64_t* elapsed_time_ms,
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D | func_test_manager.cc | 296 int64_t* elapsed_time_ms, in NeedMorePlayData() argument 557 int64_t* elapsed_time_ms, in PullRenderData() argument
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D | audio_device_test_api.cc | 120 int64_t* elapsed_time_ms, in NeedMorePlayData() argument 155 int64_t* elapsed_time_ms, in PullRenderData() argument
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
D | fakeaudiocapturemodule_unittest.cc | 89 int64_t* elapsed_time_ms, in NeedMorePlayData() argument 101 *elapsed_time_ms = 0; in NeedMorePlayData()
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D | fakeaudiocapturemodule.cc | 732 int64_t elapsed_time_ms = 0; in ReceiveFrameP() local 737 &elapsed_time_ms, &ntp_time_ms) != 0) { in ReceiveFrameP()
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
D | mock_delay_manager.h | 41 void(int elapsed_time_ms));
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
D | testutils.cc | 141 uint32 elapsed_time_ms = 0; in WriteTestPackets() local 150 RtpDumpPacket dump_packet(buf.Data(), buf.Length(), elapsed_time_ms, rtcp); in WriteTestPackets() 151 elapsed_time_ms += kElapsedTimeInterval; in WriteTestPackets()
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/external/chromium_org/third_party/webrtc/modules/audio_device/ |
D | audio_device_buffer.cc | 551 int64_t elapsed_time_ms = -1; in RequestPlayoutData() local 559 &elapsed_time_ms, in RequestPlayoutData()
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
D | webrtcvideoengine.cc | 225 int64 elapsed_time_ms = in DeliverFrame() local 230 capture_start_ntp_time_ms_ = ntp_time_ms - elapsed_time_ms; in DeliverFrame() 239 elapsed_time_ms * talk_base::kNumNanosecsPerMillisec; in DeliverFrame()
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