| /external/chromium_org/third_party/mesa/src/src/mesa/state_tracker/ |
| st_cb_fbo.h | 61 struct st_texture_object *rtt; /**< GL render to texture's texture */ member in struct:st_renderbuffer
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| call_stats.h | 47 void OnRttUpdate(uint32_t rtt);
52 // Helper struct keeping track of the time a rtt value is reported.
55 : rtt(new_rtt), time(rtt_time) {}
56 const uint32_t rtt; member in struct:webrtc::CallStats::RttTime
66 // The last RTT in the statistics update (zero if there is no valid estimate).
69 // All Rtt reports within valid time interval, oldest first.
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| call_stats_unittest.cc | 111 uint32_t rtt = 100; local
112 rtcp_rtt_stats->OnRttUpdate(rtt);
116 EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
118 EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
125 rtcp_rtt_stats->OnRttUpdate(rtt);
127 EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
129 EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
135 rtcp_rtt_stats->OnRttUpdate(rtt);
137 EXPECT_CALL(stats_observer_1, OnRttUpdate(rtt))
139 EXPECT_CALL(stats_observer_2, OnRttUpdate(rtt))
[all...] |
| /external/mesa3d/src/mesa/state_tracker/ |
| st_cb_fbo.h | 61 struct st_texture_object *rtt; /**< GL render to texture's texture */ member in struct:st_renderbuffer
|
| /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
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| rtp_rtcp_impl_unittest.cc | 253 TEST_F(RtpRtcpImplTest, Rtt) {
268 // Verify RTT.
269 uint16_t rtt; local
274 sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
275 EXPECT_EQ(2 * kOneWayNetworkDelayMs, rtt);
280 // No RTT from other ssrc.
282 sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
284 // Verify RTT from rtt_stats config
[all...] |
| /external/chromium_org/net/quic/congestion_control/ |
| rtt_stats.h | 5 // A convenience class to store rtt samples and calculate smoothed rtt.
26 // Returns true if any RTT measurements have been made.
29 // Updates the RTT from an incoming ack which is received |send_delta| after
40 // Forces RttStats to sample a new recent min rtt within the next
50 // Sets an initial RTT to be used for SmoothedRtt before any RTT updates.
67 return recent_min_rtt_.rtt;
74 // Sets how old a recent min rtt sample can be.
82 // Used to track a sampled RTT window
87 QuicTime::Delta rtt; member in struct:net::RttStats::RttSample
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/ |
| packet_loss_test.cc | 169 int rtt = 0; local
171 rtt = _rttFrames * (1000 / _inst.maxFramerate);
173 rtt);
|
| /frameworks/base/libs/common_time/ |
| diag_thread.h | 41 int32_t rtt);
52 int32_t rtt; member in struct:android::DiagThread::__anon39403
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| clock_recovery.h | 82 // The maximum allowed error rtt time for packets to be used for control
90 int64_t rtt; member in struct:android::ClockRecoveryLoop::__anon39401
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| /external/chromium_org/net/quic/ |
| quic_sent_packet_manager_test.cc | 349 QuicTime::Delta rtt = QuicTime::Delta::FromMilliseconds(15); local
350 clock_.AdvanceTime(rtt);
358 // 2 should be unacked, since it may provide an RTT measurement.
373 QuicTime::Delta rtt = QuicTime::Delta::FromMilliseconds(15); local
374 clock_.AdvanceTime(rtt);
398 QuicTime::Delta rtt = QuicTime::Delta::FromMilliseconds(15); local
399 clock_.AdvanceTime(rtt);
410 clock_.AdvanceTime(rtt);
457 // Since 2 was marked for retransmit, when 1 is acked, 2 is kept for RTT.
477 QuicTime::Delta rtt = QuicTime::Delta::FromMilliseconds(15) local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/p2p/client/ |
| connectivitychecker.h | 51 : rtt(-1), error(0) {}
56 int32 rtt; member in struct:cricket::ConnectInfo
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| /external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
| bitrate_controller_impl.cc | 36 uint16_t rtt,
71 owner_->OnReceivedRtcpReceiverReport(fraction_lost_aggregate, rtt,
153 uint32_t rtt; local
154 bandwidth_estimation_.CurrentEstimate(¤t_estimate, &loss, &rtt);
248 const uint32_t rtt,
253 fraction_loss, rtt, number_of_packets, now_ms);
260 uint32_t rtt; local
261 bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
267 rtt != last_rtt_ms_ ||
272 last_rtt_ms_ = rtt;
371 uint32_t rtt; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| test_util.h | 40 int rtt; member in class:CmdArgs
|
| /hardware/libhardware_legacy/include/hardware_legacy/ |
| rtt.h | 23 /* RTT Type */
31 /* RTT configuration */
34 wifi_rtt_type type; // optional - rtt type hint. RTT_TYPE_INVALID implies best effort
38 unsigned interval; // interval of RTT measurement (unit ms) when continuous = true
39 unsigned num_measurements; // total number of RTT measurements when continuous = true
40 unsigned num_samples_per_measurement; // num of packets in each RTT measurement
44 /* RTT results */
49 wifi_rtt_type type; // RTT type
55 wifi_timespan rtt; // round trip time in nanoseconds member in struct:__anon41493
56 wifi_timespan rtt_sd; // rtt standard deviation in nanosecond
[all...] |
| gscan.h | 63 wifi_timespan rtt; // in nanoseconds member in struct:__anon41461
64 wifi_timespan rtt_sd; // standard deviation in rtt
|
| /external/chromium_org/content/renderer/media/ |
| rtc_video_encoder.cc | 653 int32_t RTCVideoEncoder::SetChannelParameters(uint32_t packet_loss, int rtt) {
655 << ", rtt=" << rtt; local
|
| /external/iputils/ |
| clockdiff.c | 117 long rtt = 1000; variable
210 long tmo = rtt + rtt_sigma;
242 rtt = (rtt * 3 + diff)/4;
243 rtt_sigma = (rtt_sigma *3 + abs(diff-rtt))/4;
390 long tmo = rtt + rtt_sigma;
458 rtt = (rtt * 3 + diff)/4;
459 rtt_sigma = (rtt_sigma *3 + abs(diff-rtt))/4;
679 printf("\nhost=%s rtt=%ld(%ld)ms/%ldms delta=%dms/%dms %s", hisname
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| pseudotcp.cc | 720 int32 rtt = rtc::TimeDiff(now, seg.tsecr); local
721 if (rtt >= 0) {
723 m_rx_srtt = rtt;
724 m_rx_rttvar = rtt / 2;
726 uint32 unsigned_rtt = static_cast<uint32>(rtt);
730 m_rx_srtt = (7 * m_rx_srtt + rtt) / 8;
735 LOG(LS_INFO) << "rtt: " << rtt
[all...] |
| transport.h | 149 rtt(0),
161 size_t rtt; // The STUN RTT for this connection. member in struct:cricket::ConnectionInfo
|
| /external/chromium_org/third_party/usrsctp/usrsctplib/netinet/ |
| sctp_cc_functions.c | 254 if (net->rtt > net->cc_mod.rtcc.lbw_rtt + rtt_offset) {
256 * rtt increased
258 * update the rtt either.
266 ((net->cc_mod.rtcc.lbw_rtt << 32) | net->rtt),
289 ((net->cc_mod.rtcc.lbw_rtt << 32) | net->rtt),
303 if (net->rtt < net->cc_mod.rtcc.lbw_rtt-rtt_offset) {
305 * rtt decreased, there could be more room.
306 * we update both the bw and the rtt here to
315 ((net->cc_mod.rtcc.lbw_rtt << 32) | net->rtt),
329 ((net->cc_mod.rtcc.lbw_rtt << 32) | net->rtt),
610 uint64_t probepoint, rtt, vtag; local
1221 int rtt; local
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| media_opt_util.h | 51 VCMProtectionParameters() : rtt(0), lossPr(0.0f), bitRate(0.0f),
58 int rtt; member in struct:webrtc::media_optimization::VCMProtectionParameters
269 // - rtt : Round-trip time in seconds.
270 void UpdateRtt(uint32_t rtt);
|
| /external/tcpdump/ |
| sctpHeader.h | 199 /* for both RTT request/response the
219 struct sctpHBrequest rtt; member in struct:sctpHBsender
|
| /external/chromium_org/chrome/browser/net/ |
| network_stats.cc | 431 base::TimeDelta rtt = local
436 packet_rtt_[packet_index] = (rtt >= min_rtt) ? rtt : min_rtt;
605 // Only record RTT for these packet indices.
618 // No need to record RTT for PacketSizeTest.
774 "NetConnectivity5.%s.Sent%d.Success.RTT.Packet%02d.%d.%dB",
[all...] |
