/external/webrtc/webrtc/p2p/base/ |
port.cc | 59 // We will restrict RTT estimates (when used for determining state) to be 64 // When we don't have any RTT data, we have to pick something reasonable. We 68 // Computes our estimate of the RTT given the current estimate. 69 inline uint32_t ConservativeRTTEstimate(uint32_t rtt) { 70 return std::max(MINIMUM_RTT, std::min(MAXIMUM_RTT, 2 * rtt)); 73 // Weighting of the old rtt value to new data. 1030 uint32_t rtt = ConservativeRTTEstimate(rtt_); local 1210 uint32_t rtt = request->Elapsed(); local [all...] |
pseudotcp.cc | 713 int32_t rtt = rtc::TimeDiff(now, seg.tsecr); local 714 if (rtt >= 0) { 716 m_rx_srtt = rtt; 717 m_rx_rttvar = rtt / 2; 719 uint32_t unsigned_rtt = static_cast<uint32_t>(rtt); 724 m_rx_srtt = (7 * m_rx_srtt + rtt) / 8; 730 LOG(LS_INFO) << "rtt: " << rtt [all...] |
p2ptransportchannel.cc | 29 // The minimum improvement in RTT that justifies a switch. 133 return a->rtt() < b->rtt(); 177 return b_conn->rtt() <= a_conn->rtt() + kMinImprovement; [all...] |
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
packet_sender.cc | 147 int64_t rtt) { 311 int64_t rtt) { 312 VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
|
/developers/build/prebuilts/gradle/WifiRttScan/Application/src/main/java/com/example/android/wifirttscan/ |
MainActivity.java | 29 import android.net.wifi.rtt.RangingRequest; 175 + " RTT capable.");
|
/developers/samples/android/connectivity/wifirtt/WifiRttScan/Application/src/main/java/com/example/android/wifirttscan/ |
MainActivity.java | 29 import android.net.wifi.rtt.RangingRequest; 175 + " RTT capable.");
|
/device/linaro/poplar/wifi/wifi_hal/ |
common.h | 70 /* define all RTT related commands between 0x1100 and 0x11FF */ 226 u64 rtt; // in nanoseconds member in struct:wifi_gscan_result 227 u64 rtt_sd; // standard deviation in rtt
|
/external/webrtc/talk/media/webrtc/ |
fakewebrtcvideoengine.h | 173 virtual int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
rtp_rtcp_impl_unittest.cc | 244 // Min required delay until retransmit = 5 + RTT ms (RTT = 0). 273 // Min required delay until retransmit = 5 + RTT ms (RTT = 0). 290 TEST_F(RtpRtcpImplTest, Rtt) { 305 // Verify RTT. 306 int64_t rtt; local 311 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt)); 312 EXPECT_EQ(2 * kOneWayNetworkDelayMs, rtt); [all...] |
/external/webrtc/webrtc/modules/video_coding/test/ |
test_util.cc | 28 rtt(0),
|
/cts/apps/CtsVerifier/src/com/android/cts/verifier/wifiaware/ |
CallbackUtils.java | 29 import android.net.wifi.rtt.RangingResult; 30 import android.net.wifi.rtt.RangingResultCallback;
|
/external/scapy/scapy/arch/ |
unix.py | 74 dest,mask,gw,netif,mxfrg,rtt,ref,flg = lspl[:8] 76 dest,mask,gw,mxfrg,rtt,ref,flg = lspl[:7]
|
/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
transport_feedback_adapter_unittest.cc | 68 int64_t rtt, 79 int64_t rtt,
|
/external/webrtc/webrtc/modules/bitrate_controller/ |
bitrate_controller_unittest.cc | 48 int64_t rtt) { 51 last_rtt_ = rtt;
|
/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
simulcast_encoder_adapter.cc | 318 int64_t rtt) { 320 streaminfos_[stream_idx].encoder->SetChannelParameters(packet_loss, rtt);
|
/external/webrtc/webrtc/modules/video_coding/ |
video_receiver.cc | 167 int32_t VideoReceiver::SetReceiveChannelParameters(int64_t rtt) { 169 _receiver.UpdateRtt(rtt); 502 // extra delay when RTT is above kLowRttNackMs.
|
/external/webrtc/webrtc/p2p/stunprober/ |
stunprober.cc | 56 int64_t rtt() { return received_time_ms - sent_time_ms; } function in struct:stunprober::StunProber::Requester::Request 478 rtt_sum += request->rtt();
|
/external/webrtc/webrtc/video/ |
vie_encoder.cc | 100 int64_t rtt) { 101 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt); 575 << " rtt " << round_trip_time_ms;
|
video_encoder_unittest.cc | 53 int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
|
/external/iperf3/src/ |
iperf.h | 102 int rtt; member in struct:iperf_interval_results
|
/external/iputils/ |
ping_common.h | 169 extern int rtt;
|
/external/webrtc/webrtc/test/ |
fake_encoder.cc | 126 int32_t FakeEncoder::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
|
/external/webrtc/webrtc/voice_engine/ |
channel.cc | 203 int64_t rtt, 495 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, 1815 int64_t rtt = GetRTT(true); local 4076 int64_t rtt = 0; local [all...] |
/external/u-boot/arch/arm/dts/ |
at91sam9m10g45ek.dts | 208 atmel,rtt-rtc-time-reg = <&gpbr 0x0>;
|
/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/estimators/ |
nada_unittest.cc | 90 int64_t rtt) { 93 last_rtt_ = rtt;
|