/external/webrtc/webrtc/modules/pacing/ |
D | bitrate_prober.cc | 88 int64_t elapsed_time_ms = now_ms - time_last_send_ms_; in TimeUntilNextProbe() local
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D | paced_sender.cc | 352 int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000; in TimeUntilNextProcess() local 359 int64_t elapsed_time_ms = (now_us - time_last_update_us_ + 500) / 1000; in Process() local
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/external/webrtc/webrtc/test/ |
D | fake_audio_device.cc | 118 int64_t elapsed_time_ms = -1; in CaptureAudio() local
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/external/webrtc/webrtc/modules/audio_coding/neteq/ |
D | decision_logic.cc | 155 const int elapsed_time_ms = in FilterBufferLevel() local
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D | delay_manager.cc | 341 void DelayManager::UpdateCounters(int elapsed_time_ms) { in UpdateCounters()
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/external/webrtc/webrtc/voice_engine/ |
D | voe_base_impl.cc | 104 int64_t* elapsed_time_ms, in NeedMorePlayData() 169 void* audio_data, int64_t* elapsed_time_ms, in PullRenderData() 769 void* audio_data, int64_t* elapsed_time_ms, in GetPlayoutData()
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/external/grpc-grpc/src/cpp/ext/filters/census/ |
D | server_filter.cc | 184 double elapsed_time_ms = absl::ToDoubleMilliseconds(elapsed_time_); in Destroy() local
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/external/webrtc/talk/app/webrtc/test/ |
D | fakeaudiocapturemodule_unittest.cc | 89 int64_t* elapsed_time_ms, in NeedMorePlayData()
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D | fakeaudiocapturemodule.cc | 702 int64_t elapsed_time_ms = 0; in ReceiveFrameP() local
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
D | conference_transport.cc | 208 int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_); in DispatchPackets() local
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/external/webrtc/webrtc/modules/audio_device/ |
D | audio_device_buffer.cc | 537 int64_t elapsed_time_ms = -1; in RequestPlayoutData() local
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/external/webrtc/webrtc/modules/audio_device/include/ |
D | audio_device_defines.h | 131 int64_t* elapsed_time_ms, in PullRenderData()
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
D | unittest.cc | 142 int elapsed_time_ms = 0; in TestGetSetBandwidthInfo() local
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/external/webrtc/talk/media/base/ |
D | testutils.cc | 146 uint32_t elapsed_time_ms = 0; in WriteTestPackets() local
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/external/webrtc/webrtc/modules/audio_device/ios/ |
D | audio_device_unittest_ios.cc | 444 int64_t* elapsed_time_ms, in RealNeedMorePlayData()
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/external/webrtc/webrtc/modules/audio_device/test/ |
D | func_test_manager.cc | 346 int64_t* elapsed_time_ms, in NeedMorePlayData()
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D | audio_device_test_api.cc | 117 int64_t* elapsed_time_ms, in NeedMorePlayData()
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/external/webrtc/webrtc/modules/audio_device/android/ |
D | audio_device_unittest.cc | 452 int64_t* elapsed_time_ms, in RealNeedMorePlayData()
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvideoengine2.cc | 2445 int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame / in RenderFrame() local
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