| /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;
|
