Searched refs:rtt (Results 1 - 25 of 116) sorted by relevance

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/external/chromium_org/net/quic/congestion_control/
H A Drtt_stats.h5 // A convenience class to store rtt samples and calculate smoothed rtt.
40 // Forces RttStats to sample a new recent min rtt within the next
67 return recent_min_rtt_.rtt;
74 // Sets how old a recent min rtt sample can be.
84 RttSample() : rtt(QuicTime::Delta::Zero()), time(QuicTime::Zero()) { }
85 RttSample(QuicTime::Delta rtt, QuicTime time) : rtt(rtt), time(time) { } argument
87 QuicTime::Delta rtt; member in struct:net::RttStats::RttSample
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H A Drtt_stats.cc15 // Default initial rtt used before any samples are received.
67 DVLOG(1) << "Ignoring rtt, because it's "
100 if (new_min_rtt_.rtt.IsZero() || rtt_sample <= new_min_rtt_.rtt) {
108 // Update the three recent rtt samples.
109 if (recent_min_rtt_.rtt.IsZero() || rtt_sample <= recent_min_rtt_.rtt) {
112 } else if (rtt_sample <= half_window_rtt_.rtt) {
115 } else if (rtt_sample <= quarter_window_rtt_.rtt) {
119 // Expire old min rtt sample
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H A Dhybrid_slow_start.h36 // rtt: the RTT for this ack packet.
39 bool ShouldExitSlowStart(QuicTime::Delta rtt,
82 uint32 rtt_sample_count_; // Number of rtt samples in the current round.
83 QuicTime::Delta current_min_rtt_; // The minimum rtt of current round.
/external/chromium_org/third_party/webrtc/video_engine/
H A Dcall_stats.h47 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
H A Dcall_stats_unittest.cc111 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))
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H A Dcall_stats.cc21 // A rtt report is considered valid for this long.
31 virtual void OnRttUpdate(uint32_t rtt) { argument
32 owner_->OnRttUpdate(rtt);
66 // Remove invalid, as in too old, rtt values.
77 if (it->rtt > max_rtt)
78 max_rtt = it->rtt;
81 // If there is a valid rtt, update all observers.
123 void CallStats::OnRttUpdate(uint32_t rtt) { argument
126 reports_.push_back(RttTime(rtt, time_now));
/external/chromium_org/third_party/webrtc/modules/bitrate_controller/
H A Dbitrate_controller_impl.cc36 uint16_t rtt,
71 owner_->OnReceivedRtcpReceiverReport(fraction_lost_aggregate, rtt,
153 uint32_t rtt; local
154 bandwidth_estimation_.CurrentEstimate(&current_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;
246 OnReceivedRtcpReceiverReport( const uint8_t fraction_loss, const uint32_t rtt, const int number_of_packets, const uint32_t now_ms) argument
280 OnNetworkChanged(const uint32_t bitrate, const uint8_t fraction_loss, const uint32_t rtt) argument
298 NormalRateAllocation(uint32_t bitrate, uint8_t fraction_loss, uint32_t rtt, uint32_t sum_min_bitrates) argument
338 LowRateAllocation(uint32_t bitrate, uint8_t fraction_loss, uint32_t rtt, uint32_t sum_min_bitrates) argument
371 uint32_t rtt; local
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H A Dsend_side_bandwidth_estimation.cc26 uint32_t CalcTfrcBps(uint16_t rtt, uint8_t loss) { argument
27 if (rtt == 0 || loss == 0) {
31 double R = static_cast<double>(rtt) / 1000; // RTT in seconds.
83 uint32_t* rtt) const {
86 *rtt = last_round_trip_time_ms_;
95 uint32_t rtt,
99 last_round_trip_time_ms_ = rtt;
154 // rtt.
94 UpdateReceiverBlock(uint8_t fraction_loss, uint32_t rtt, int number_of_packets, uint32_t now_ms) argument
H A Dsend_side_bandwidth_estimation.h27 void CurrentEstimate(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const;
37 uint32_t rtt,
H A Dbitrate_controller_impl.h86 const uint32_t rtt,
94 const uint32_t rtt)
99 uint32_t rtt,
105 uint32_t rtt,
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Dremote_ntp_time_estimator.cc32 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;
/external/chromium_org/third_party/mesa/src/src/mesa/state_tracker/
H A Dst_cb_fbo.h61 struct st_texture_object *rtt; /**< GL render to texture's texture */ member in struct:st_renderbuffer
H A Dst_atom_framebuffer.c56 struct pipe_resource *resource = strb->rtt ? strb->rtt->pt : strb->texture;
123 /*printf("--------- framebuffer surface rtt %p\n", strb->rtt);*/
124 if (strb->rtt ||
147 if (strb->rtt) {
/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/vp8/
H A Dreference_picture_selection.h55 // Set the round-trip time between the sender and the receiver to |rtt|
57 void SetRtt(int rtt);
H A Dreference_picture_selection.cc117 void ReferencePictureSelection::SetRtt(int rtt) { argument
119 rtt_ = 90 * rtt;
/external/mesa3d/src/mesa/state_tracker/
H A Dst_cb_fbo.h61 struct st_texture_object *rtt; /**< GL render to texture's texture */ member in struct:st_renderbuffer
H A Dst_atom_framebuffer.c56 struct pipe_resource *resource = strb->rtt ? strb->rtt->pt : strb->texture;
123 /*printf("--------- framebuffer surface rtt %p\n", strb->rtt);*/
124 if (strb->rtt ||
147 if (strb->rtt) {
/external/chromium_org/media/cast/sender/
H A Dcongestion_control.h22 // Called with latest measured rtt value.
23 virtual void UpdateRtt(base::TimeDelta rtt) = 0;
H A Dcongestion_control_unittest.cc38 base::TimeDelta rtt,
42 congestion_control_->UpdateRtt(rtt);
36 Run(uint32 frames, size_t frame_size, base::TimeDelta rtt, base::TimeDelta frame_delay, base::TimeDelta ack_time) argument
/external/chromium_org/third_party/webrtc/modules/bitrate_controller/include/
H A Dbitrate_controller.h26 * to get the target bitrate. It also get the fraction loss and rtt to
34 const uint32_t rtt) = 0;
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dtester_main.cc33 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
76 args.rtt = FLAGS_rtt;
/external/chromium_org/net/dns/
H A Ddns_session.cc160 void DnsSession::RecordRTT(unsigned server_index, base::TimeDelta rtt) { argument
166 UMA_HISTOGRAM_TIMES("AsyncDNS.TimeoutErrorJacobson", rtt - timeout_jacobson);
168 rtt - timeout_histogram);
170 timeout_jacobson - rtt);
172 timeout_histogram - rtt);
178 base::TimeDelta current_error = rtt - estimate;
186 ->Accumulate(rtt.InMilliseconds(), 1);
/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/
H A Dpacket_loss_test.cc169 int rtt = 0; local
171 rtt = _rttFrames * (1000 / _inst.maxFramerate);
173 rtt);
/external/chromium_org/chrome/renderer/media/
H A Dcast_ipc_dispatcher.cc105 base::TimeDelta rtt) {
108 sender->OnRtt(ssrc, rtt);
103 OnRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt) argument
H A Dcast_ipc_dispatcher.h52 void OnRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt);

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