/external/webrtc/webrtc/p2p/base/ |
transport.h | 90 rtt(0), 104 size_t rtt; // The STUN RTT for this connection. member in struct:cricket::ConnectionInfo
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/external/u-boot/arch/arm/dts/ |
at91sam9g20ek_common.dtsi | 109 atmel,wakeup-rtt-timer; 113 atmel,rtt-rtc-time-reg = <&gpbr 0x0>;
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/external/webrtc/webrtc/call/ |
bitrate_allocator.h | 39 int64_t rtt);
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bitrate_allocator_unittest.cc | 27 int64_t rtt) { 30 last_rtt_ = rtt;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
rtcp_receiver_help.h | 41 // RTT 42 int64_t RTT; 76 int64_t rtt; member in class:webrtc::RTCPHelp::RTCPPacketInformation
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/external/webrtc/webrtc/modules/video_coding/codecs/interface/mock/ |
mock_video_codec_interface.h | 48 MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
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/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
simulcast_encoder_adapter.h | 47 int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
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vp8_impl.h | 55 virtual int SetChannelParameters(uint32_t packet_loss, int64_t rtt);
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/external/webrtc/webrtc/modules/video_coding/ |
generic_encoder.h | 33 int64_t rtt; member in struct:webrtc::EncoderParameters
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video_coding_impl.cc | 129 int64_t rtt) override { 130 return sender_.SetChannelParameters(target_bitrate, lossRate, rtt); 272 int32_t SetReceiveChannelParameters(int64_t rtt) override { 273 return receiver_.SetReceiveChannelParameters(rtt);
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video_sender.cc | 186 int64_t rtt) { 188 _mediaOpt.SetTargetRates(target_bitrate, lossRate, rtt, 194 encoder_params_ = {target_rate, lossRate, rtt, input_frame_rate};
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generic_encoder.cc | 177 params.rtt != encoder_params_.rtt; 184 encoder_->SetChannelParameters(params.loss_rate, params.rtt);
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receiver.cc | 66 void VCMReceiver::UpdateRtt(int64_t rtt) { 67 jitter_buffer_.UpdateRtt(rtt);
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video_coding_impl.h | 80 int64_t rtt); 173 int32_t SetReceiveChannelParameters(int64_t rtt);
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/external/webrtc/webrtc/modules/video_coding/include/mock/ |
mock_video_codec_interface.h | 46 MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
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/external/webrtc/webrtc/test/ |
configurable_frame_size_encoder.cc | 72 int64_t rtt) {
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/cts/apps/CtsVerifier/src/com/android/cts/verifier/wifiaware/testcase/ |
DiscoveryWithRangingTestCase.java | 20 import android.net.wifi.rtt.RangingRequest; 21 import android.net.wifi.rtt.RangingResult; 22 import android.net.wifi.rtt.WifiRttManager; 43 * 5. Wait for message ("done with RTT") 46 * 8. Send message ("done with RTT") 58 * 6. Send message ("done with RTT") 60 * 7. Wait for Message ("done with RTT") 65 * Note that the process tries to prevent RTT concurrency, each peer will execute RTT in 76 private static final byte[] MSG_RTT = "Done with RTT".getBytes() [all...] |
/cts/tests/tests/net/src/android/net/wifi/rtt/cts/ |
WifiRttTest.java | 17 package android.net.wifi.rtt.cts; 20 import android.net.wifi.rtt.RangingRequest; 21 import android.net.wifi.rtt.RangingResult; 33 * Wi-Fi RTT CTS test: range to all available Access Points which support IEEE 802.11mc. 40 // Number of RTT measurements per AP 43 // Maximum failure rate of RTT measurements (percentage) 53 * Test Wi-Fi RTT ranging operation: 55 * - Perform N (constant) RTT operations 68 "Cannot find any test APs which support RTT / IEEE 802.11mc - please verify that " 72 // Perform RTT operation [all...] |
/external/grpc-grpc-java/core/src/main/java/io/grpc/ |
InternalChannelz.java | 747 public final int rtt; field in class:InternalChannelz.TcpInfo 757 int lastDataRecv, int lastAckRecv, int pmtu, int rcvSsthresh, int rtt, int rttvar, 782 this.rtt = rtt; 814 private int rtt; field in class:InternalChannelz.TcpInfo.Builder 936 public Builder setRtt(int rtt) { 937 this.rtt = rtt; 971 lastAckSent, lastDataRecv, lastAckRecv, pmtu, rcvSsthresh, rtt, rttvar, sndSsthresh, [all...] |
/external/libnl/lib/route/ |
route_utils.c | 147 __ADD(RTAX_RTT, rtt)
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/external/webrtc/webrtc/modules/video_coding/codecs/vp9/ |
vp9_impl.h | 46 int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
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/external/webrtc/webrtc/video/ |
vie_receiver.cc | 427 int64_t rtt = 0; local 428 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL); 429 if (rtt == 0) { 430 // Waiting for valid rtt. 441 ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); 479 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
aimd_rate_control.cc | 98 void AimdRateControl::SetRtt(int64_t rtt) { 99 rtt_ = rtt;
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
rtp_rtcp_defines.h | 253 int64_t rtt, 321 virtual void OnRttUpdate(int64_t rtt) = 0;
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/external/webrtc/webrtc/modules/video_coding/test/ |
plotJitterEstimate.m | 3 [timestamps, framedata, slopes, randJitters, framestats, timetable, filtjitter, rtt, rttStatsVec] = jitterBufferTraceParser(filename);
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