| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator.cc | 32 uint16_t rtt = 0; local
33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL);
34 if (rtt == 0) {
35 // Waiting for valid rtt.
63 int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90;
|
| rtcp_receiver_help.h | 36 // RTT
37 uint16_t RTT;
71 uint16_t rtt; member in class:webrtc::RTCPHelp::RTCPPacketInformation
|
| rtcp_receiver.cc | 178 LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC;
181 reportBlock->RTT = 0;
188 int32_t RTCPReceiver::RTT(uint32_t remoteSSRC,
189 uint16_t* RTT,
201 if (RTT) {
202 *RTT = reportBlock->RTT;
487 // We can calc RTT if we send a send report and get a report block back.
550 // Estimate RTT
555 int32_t RTT = 0
968 int32_t rtt = _clock->CurrentNtpInMilliseconds() - delay_rr_ms - send_time_ms; local
[all...] |
| rtcp_receiver_help.cc | 29 rtt(0),
99 this->rtt = report_block_info.RTT;
106 RTT(0),
|
| remote_ntp_time_estimator_unittest.cc | 60 EXPECT_CALL(*rtp_rtcp, RTT(_, _, _, _, _))
|
| rtcp_receiver.h | 74 // get rtt
75 int32_t RTT(uint32_t remoteSSRC,
76 uint16_t* RTT,
251 // Estimated rtt, zero when there is no valid estimate.
|
| rtp_rtcp_impl.cc | 171 // Process RTT if we have received a receiver report and we haven't
172 // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
180 uint16_t rtt = 0; local
181 rtcp_receiver_.RTT(it->remoteSSRC, &rtt, NULL, NULL, NULL);
182 max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
184 // Report the rtt.
210 // Report rtt from receiver.
219 // Get processed rtt
875 uint16_t rtt = rtt_ms(); local
1239 uint16_t rtt = rtt_ms(); local
[all...] |
| rtp_rtcp_impl_unittest.cc | 134 TEST_F(RtpRtcpImplTest, Rtt) {
149 // Verify RTT.
150 uint16_t rtt; local
155 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
156 EXPECT_EQ(2 * kOneWayNetworkDelayMs, rtt);
161 // No RTT from other ssrc.
163 sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
165 // Verify RTT from rtt_stats config
[all...] |
| rtp_rtcp_impl.h | 167 virtual int32_t RTT(const uint32_t remote_ssrc,
168 uint16_t* rtt,
404 FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
441 // The processed RTT from RtcpRttStats.
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api_rtcp.cc | 344 uint16_t RTT;
350 EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT));
351 EXPECT_GE(10, RTT);
|
| /prebuilts/gcc/linux-x86/host/x86_64-w64-mingw32-4.8/x86_64-w64-mingw32/include/ |
| qos2.h | 60 UINT32 RTT;
|
| iphlpapi.h | 80 WINBOOL WINAPI GetRTTAndHopCount(IPAddr DestIpAddress,PULONG HopCount,ULONG MaxHops,PULONG RTT);
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_rtcp.h | 434 virtual int32_t RTT(const uint32_t remoteSSRC,
435 uint16_t* RTT,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 151 MOCK_CONST_METHOD5(RTT,
152 int32_t(const uint32_t remoteSSRC, uint16_t* RTT, uint16_t* avgRTT, uint16_t* minRTT, uint16_t* maxRTT));
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_receiver.cc | 416 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
|
| vie_channel.cc | 51 virtual void OnRttUpdate(uint32_t rtt) {
52 owner_->OnRttUpdate(rtt);
1034 uint16_t rtt = 0; local
1075 uint16_t rtt = 0; local
[all...] |
| /external/chromium_org/remoting/webapp/ |
| oauth2.js | 142 // Offset by a further 30 seconds to account for RTT issues.
|
| /external/chromium_org/third_party/webrtc/voice_engine/ |
| channel.cc | 102 const uint32_t rtt) OVERRIDE {
104 owner_->OnNetworkChanged(bitrate_bps, fraction_lost, rtt);
549 _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time,
[all...] |
| /external/iproute2/doc/ |
| ip-cref.tex | [all...] |
