| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_audio.cc | 22 _clock(clock), in RTPSenderAudio()
220 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() - in SendTelephoneEventActive()
249 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() - in SendAudio()
300 _dtmfTimeLastSent = _clock->TimeInMilliseconds(); in SendAudio()
355 _clock->TimeInMilliseconds()); in SendAudio()
361 _clock->TimeInMilliseconds()); in SendAudio()
532 dtmfTimeStamp, _clock->TimeInMilliseconds()); in SendTelephoneEventPacket()
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| D | rtcp_receiver.cc | 35 _clock(clock), in RTCPReceiver()
274 _clock->CurrentNtp(ntp_sec, ntp_frac); in LastReceivedXrReferenceTimeInfo()
320 _lastReceived = _clock->TimeInMilliseconds(); in IncomingRTCPPacket()
451 _clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); in HandleSenderReceiverReport()
514 _lastReceivedRrMs = _clock->TimeInMilliseconds(); in HandleReportBlock()
544 _clock->CurrentNtp(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); in HandleReportBlock()
683 receiveInformation.lastTimeReceived = _clock->TimeInMilliseconds(); in UpdateReceiveInformation()
692 if (_clock->TimeInMilliseconds() > _lastReceivedRrMs + time_out_ms) { in RtcpRrTimeout()
706 if (_clock->TimeInMilliseconds() > _lastIncreasedSequenceNumberMs + in RtcpRrSequenceNumberTimeout()
719 int64_t timeNow = _clock->TimeInMilliseconds(); in UpdateRTCPReceiveInformationTimers()
[all …]
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| D | rtcp_sender.cc | 104 _clock(clock), in RTCPSender()
268 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + in SetRTCPStatus()
272 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + in SetRTCPStatus()
349 _nextTimeToSendRTCP = _clock->TimeInMilliseconds(); in SetREMBData()
393 last_frame_capture_time_ms_ = _clock->TimeInMilliseconds(); in SetLastRtpTime()
409 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100; in SetSSRC()
540 int64_t now = _clock->TimeInMilliseconds(); in TimeToSendRTCPReport()
701 _clock->TimeInMilliseconds() - last_frame_capture_time_ms_) * in BuildSR()
1837 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + timeToNext; in PrepareRTCP()
1844 _clock->CurrentNtp(NTPsec, NTPfrac); in PrepareRTCP()
[all …]
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| D | rtp_sender_audio.h | 81 Clock* _clock; variable
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| D | rtcp_receiver.h | 225 Clock* _clock; variable
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| D | rtcp_sender.h | 275 Clock* _clock; variable
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| /external/chromium_org/third_party/webrtc/modules/audio_device/dummy/ |
| D | file_audio_device.cc | 50 _clock(Clock::GetRealTimeClock()) { in FileAudioDevice()
533 uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); in PlayThreadProcess()
553 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); in PlayThreadProcess()
562 uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); in RecThreadProcess()
582 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); in RecThreadProcess()
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| D | file_audio_device.h | 197 Clock* _clock; variable
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| D | video_coding_impl.h | 40 : _clock(clock), in VCMProcessTimer()
42 _latestMs(_clock->TimeInMilliseconds()) {} in VCMProcessTimer()
48 Clock* _clock;
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| D | generic_decoder.cc | 23 _clock(clock), in VCMDecodedFrameCallback()
71 _clock->TimeInMilliseconds()); in Decoded()
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| D | generic_decoder.h | 54 Clock* _clock; variable
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| D | video_coding_impl.cc | 31 const int64_t time_since_process = _clock->TimeInMilliseconds() - in TimeUntilProcess()
42 _latestMs = _clock->TimeInMilliseconds(); in Processed()
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| /external/llvm/test/CodeGen/X86/ |
| D | coalescer-cross.ll | 30 %0 = tail call i32 @"\01_clock$UNIX2003"() nounwind ; <i32> [#uses=1]
45 declare i32 @"\01_clock$UNIX2003"()
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| D | generic_codec_test.cc | 46 _clock(clock), in GenericCodecTest()
487 int64_t startTime = _clock->TimeInMilliseconds(); in WaitForEncodedFrame()
488 while (_clock->TimeInMilliseconds() - startTime < kMaxWaitEncTimeMs*10) in WaitForEncodedFrame()
501 _clock->AdvanceTimeMilliseconds(1000/frameRate); in IncrementDebugClock()
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| D | media_opt_test.h | 61 webrtc::Clock* _clock; variable
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| D | test_callbacks.cc | 210 _clock(clock), in RTPSendCompleteCallback()
264 int64_t now = _clock->TimeInMilliseconds(); in SendPacket()
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| D | generic_codec_test.h | 52 webrtc::SimulatedClock* _clock; variable
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| D | normal_test.cc | 182 _clock(clock), in NormalTest()
325 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(framePeriod); in Perform()
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| D | normal_test.h | 113 webrtc::Clock* _clock; variable
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| D | media_opt_test.cc | 73 _clock(clock), in MediaOptTest()
200 _outgoingTransport = new RTPSendCompleteCallback(_clock); in GeneralSetup()
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| D | test_callbacks.h | 191 Clock* _clock; variable
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| D | quality_modes_test.cc | 372 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds( in Perform()
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