| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver.cc | 165 int32_t RTCPReceiver::RTT(uint32_t remoteSSRC,
166 int64_t* RTT,
178 if (RTT) {
179 *RTT = reportBlock->RTT;
478 // We can calc RTT if we send a send report and get a report block back.
542 // Estimate RTT
547 int64_t RTT = 0;
550 RTT = receiveTimeMS - d - sendTimeMS;
551 if (RTT <= 0)
956 int64_t rtt = _clock->CurrentNtpInMilliseconds() - delay_rr_ms - send_time_ms; local
[all...] |
| rtcp_receiver_help.cc | 30 rtt(0),
99 this->rtt = report_block_info.RTT;
106 RTT(0),
|
| rtcp_receiver_help.h | 41 // RTT
42 int64_t RTT;
76 int64_t rtt; member in class:webrtc::RTCPHelp::RTCPPacketInformation
|
| rtcp_receiver.h | 74 // get rtt
75 int32_t RTT(uint32_t remoteSSRC,
76 int64_t* RTT,
297 // Estimated rtt, zero when there is no valid estimate.
|
| rtp_rtcp_impl.cc | 131 // Process RTT if we have received a receiver report and we haven't
132 // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
140 int64_t rtt = 0; local
141 rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
142 max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
144 // Report the rtt.
170 // Report rtt from receiver.
179 // Get processed rtt
734 int64_t rtt = rtt_ms(); local
920 int64_t rtt = rtt_ms(); local
[all...] |
| 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...] |
| rtp_rtcp_impl.h | 155 int32_t RTT(uint32_t remote_ssrc,
156 int64_t* rtt,
348 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
383 // The processed RTT from RtcpRttStats.
|
| rtcp_receiver_unittest.cc | 119 rtcp_packet_info_.rtt = rtcpPacketInformation.rtt;
372 -1, rtcp_receiver_->RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
385 0, rtcp_receiver_->RTT(kSenderSsrc, nullptr, nullptr, nullptr, nullptr));
388 EXPECT_EQ(-1, rtcp_receiver_->RTT(kSenderSsrc + 1, nullptr, nullptr, nullptr,
668 // since the RTT is not estimated.
688 // since the RTT is not estimated.
714 // since the RTT is not estimated.
[all...] |
| /prebuilts/gcc/linux-x86/host/x86_64-w64-mingw32-4.8/x86_64-w64-mingw32/include/ |
| qos2.h | 60 UINT32 RTT;
|
| iphlpapi.h | 105 WINBOOL WINAPI GetRTTAndHopCount (IPAddr DestIpAddress, PULONG HopCount, ULONG MaxHops, PULONG RTT);
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp.h | 386 virtual int32_t RTT(uint32_t remoteSSRC,
387 int64_t* RTT,
|
| /external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 158 MOCK_CONST_METHOD5(RTT,
160 int64_t* RTT,
|
| /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);
|
| vie_channel.cc | 768 // have a chance of calculating an RTT we will try with the SSRC of the
785 int64_t rtt = 0; local
786 if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) !=
790 *rtt_ms = rtt;
[all...] |
| /external/boringssl/src/ssl/test/runner/ |
| handshake_client.go | 741 // 0-RTT is implemented.
934 // If we're expecting 0.5-RTT messages from the server, read them now.
937 // BoringSSL will always send two tickets half-RTT when
938 // negotiating 0-RTT.
942 return fmt.Errorf("tls: error reading half-RTT ticket: %s", err)
946 return errors.New("tls: expected half-RTT ticket")
[all...] |
| runner.go | 86 // HalfRTTTickets is the number of half-RTT tickets the client should
87 // expect before half-RTT data when testing 0-RTT.
[all...] |
| handshake_server.go | 461 // TODO(svaldez): Check the obfuscatedTicketAge before accepting 0-RTT.
962 // Send 0.5-RTT messages.
[all...] |
| common.go | [all...] |
| conn.go | 251 // after 0-RTT data was rejected.
[all...] |
| /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...] |
| /prebuilts/go/darwin-x86/src/net/http/ |
| h2_bundle.go | [all...] |
| /prebuilts/go/linux-x86/src/net/http/ |
| h2_bundle.go | [all...] |
