/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_sender_audio.cc | 22 _clock(clock), 220 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() - 249 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() - 300 _dtmfTimeLastSent = _clock->TimeInMilliseconds(); 357 _clock->TimeInMilliseconds()); 363 _clock->TimeInMilliseconds()); 534 dtmfTimeStamp, _clock->TimeInMilliseconds());
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H A D | rtcp_receiver.cc | 35 _clock(clock), 274 _clock->CurrentNtp(ntp_sec, ntp_frac); 320 _lastReceived = _clock->TimeInMilliseconds(); 451 _clock->CurrentNtp(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); 514 _lastReceivedRrMs = _clock->TimeInMilliseconds(); 544 _clock->CurrentNtp(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); 683 receiveInformation.lastTimeReceived = _clock->TimeInMilliseconds(); 692 if (_clock->TimeInMilliseconds() > _lastReceivedRrMs + time_out_ms) { 706 if (_clock->TimeInMilliseconds() > _lastIncreasedSequenceNumberMs + 719 int64_t timeNow = _clock [all...] |
H A D | rtp_sender_audio.h | 80 Clock* _clock; member in class:webrtc::RTPSenderAudio
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H A D | rtcp_sender.cc | 85 _clock(clock), 194 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 198 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 275 _nextTimeToSendRTCP = _clock->TimeInMilliseconds(); 320 last_frame_capture_time_ms_ = _clock->TimeInMilliseconds(); 336 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100; 459 int64_t now = _clock->TimeInMilliseconds(); 617 (_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) * 1751 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + timeToNext; 1758 _clock [all...] |
H A D | rtcp_receiver.h | 225 Clock* _clock; member in class:webrtc::RTCPReceiver
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H A D | rtcp_sender.h | 281 Clock* const _clock; member in class:webrtc::RTCPSender
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | generic_decoder.h | 54 Clock* _clock; member in class:webrtc::VCMDecodedFrameCallback
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H A D | generic_decoder.cc | 23 _clock(clock), 71 _clock->TimeInMilliseconds());
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H A D | video_coding_impl.h | 41 : _clock(clock), 43 _latestMs(_clock->TimeInMilliseconds()) {} 49 Clock* _clock; member in class:webrtc::vcm::VCMProcessTimer
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H A D | video_coding_impl.cc | 31 const int64_t time_since_process = _clock->TimeInMilliseconds() - 42 _latestMs = _clock->TimeInMilliseconds();
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | media_opt_test.h | 61 webrtc::Clock* _clock; member in class:MediaOptTest
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H A D | generic_codec_test.h | 52 webrtc::SimulatedClock* _clock; member in class:webrtc::GenericCodecTest
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H A D | normal_test.h | 115 webrtc::Clock* _clock; member in class:NormalTest
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H A D | test_callbacks.cc | 210 _clock(clock), 264 int64_t now = _clock->TimeInMilliseconds();
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H A D | generic_codec_test.cc | 46 _clock(clock), 484 int64_t startTime = _clock->TimeInMilliseconds(); 485 while (_clock->TimeInMilliseconds() - startTime < kMaxWaitEncTimeMs*10) 498 _clock->AdvanceTimeMilliseconds(1000/frameRate);
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H A D | normal_test.cc | 182 _clock(clock), 325 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(framePeriod);
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H A D | media_opt_test.cc | 73 _clock(clock), 200 _outgoingTransport = new RTPSendCompleteCallback(_clock);
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H A D | test_callbacks.h | 191 Clock* _clock; member in class:webrtc::RTPSendCompleteCallback
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H A D | quality_modes_test.cc | 372 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(
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/external/chromium_org/third_party/webrtc/modules/audio_device/dummy/ |
H A D | file_audio_device.cc | 50 _clock(Clock::GetRealTimeClock()) { 534 uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); 554 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime)); 563 uint64_t currentTime = _clock->CurrentNtpInMilliseconds(); 583 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
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H A D | file_audio_device.h | 197 Clock* _clock; member in class:webrtc::FileAudioDevice
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