| /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);
|
