| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender_audio.cc | 22 _clock(clock),
220 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() -
249 int64_t delaySinceLastDTMF = _clock->TimeInMilliseconds() -
300 _dtmfTimeLastSent = _clock->TimeInMilliseconds();
355 _clock->TimeInMilliseconds());
361 _clock->TimeInMilliseconds());
532 dtmfTimeStamp, _clock->TimeInMilliseconds());
|
| 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->TimeInMilliseconds()
[all...] |
| rtp_sender_audio.h | 81 Clock* _clock; member in class:webrtc::RTPSenderAudio
|
| rtcp_sender.cc | 104 _clock(clock),
268 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
272 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
349 _nextTimeToSendRTCP = _clock->TimeInMilliseconds();
393 last_frame_capture_time_ms_ = _clock->TimeInMilliseconds();
409 _nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100;
540 int64_t now = _clock->TimeInMilliseconds();
701 _clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
[all...] |
| rtcp_receiver.h | 225 Clock* _clock; member in class:webrtc::RTCPReceiver
|
| rtcp_sender.h | 275 Clock* _clock; member in class:webrtc::RTCPSender
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| generic_decoder.h | 54 Clock* _clock; member in class:webrtc::VCMDecodedFrameCallback
|
| generic_decoder.cc | 23 _clock(clock),
71 _clock->TimeInMilliseconds());
|
| video_coding_impl.h | 40 : _clock(clock),
42 _latestMs(_clock->TimeInMilliseconds()) {}
48 Clock* _clock; member in class:webrtc::vcm::VCMProcessTimer
|
| video_coding_impl.cc | 31 const int64_t time_since_process = _clock->TimeInMilliseconds() -
42 _latestMs = _clock->TimeInMilliseconds();
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| media_opt_test.h | 61 webrtc::Clock* _clock; member in class:MediaOptTest
|
| generic_codec_test.h | 52 webrtc::SimulatedClock* _clock; member in class:webrtc::GenericCodecTest
|
| normal_test.h | 113 webrtc::Clock* _clock; member in class:NormalTest
|
| test_callbacks.cc | 210 _clock(clock),
264 int64_t now = _clock->TimeInMilliseconds();
|
| generic_codec_test.cc | 46 _clock(clock),
487 int64_t startTime = _clock->TimeInMilliseconds();
488 while (_clock->TimeInMilliseconds() - startTime < kMaxWaitEncTimeMs*10)
501 _clock->AdvanceTimeMilliseconds(1000/frameRate);
|
| test_callbacks.h | 191 Clock* _clock; member in class:webrtc::RTPSendCompleteCallback
|
| media_opt_test.cc | 73 _clock(clock),
200 _outgoingTransport = new RTPSendCompleteCallback(_clock);
|
| normal_test.cc | 182 _clock(clock),
325 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(framePeriod);
|
| quality_modes_test.cc | 372 static_cast<SimulatedClock*>(_clock)->AdvanceTimeMilliseconds(
|
| /external/chromium_org/third_party/webrtc/modules/audio_device/dummy/ |
| file_audio_device.cc | 50 _clock(Clock::GetRealTimeClock()) {
533 uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
553 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
562 uint64_t currentTime = _clock->CurrentNtpInMilliseconds();
582 SleepMs(10 - (_clock->CurrentNtpInMilliseconds() - currentTime));
|
| file_audio_device.h | 197 Clock* _clock; member in class:webrtc::FileAudioDevice
|
