/external/chromium_org/net/quic/congestion_control/ |
H A D | rtt_stats.h | 5 // 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 [all...] |
H A D | rtt_stats.cc | 15 // 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 [all...] |
H A D | hybrid_slow_start.h | 36 // 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.
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | 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
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H A D | 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...] |
H A D | call_stats.cc | 21 // 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));
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/ |
H A D | 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; 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 [all...] |
H A D | send_side_bandwidth_estimation.cc | 26 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
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H A D | send_side_bandwidth_estimation.h | 27 void CurrentEstimate(uint32_t* bitrate, uint8_t* loss, uint32_t* rtt) const; 37 uint32_t rtt,
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H A D | bitrate_controller_impl.h | 86 const uint32_t rtt, 94 const uint32_t rtt) 99 uint32_t rtt, 105 uint32_t rtt,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | 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;
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/external/chromium_org/third_party/mesa/src/src/mesa/state_tracker/ |
H A D | 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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H A D | st_atom_framebuffer.c | 56 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) {
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/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/vp8/ |
H A D | reference_picture_selection.h | 55 // Set the round-trip time between the sender and the receiver to |rtt| 57 void SetRtt(int rtt);
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H A D | reference_picture_selection.cc | 117 void ReferencePictureSelection::SetRtt(int rtt) { argument 119 rtt_ = 90 * rtt;
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/external/mesa3d/src/mesa/state_tracker/ |
H A D | 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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H A D | st_atom_framebuffer.c | 56 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) {
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/external/chromium_org/media/cast/sender/ |
H A D | congestion_control.h | 22 // Called with latest measured rtt value. 23 virtual void UpdateRtt(base::TimeDelta rtt) = 0;
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H A D | congestion_control_unittest.cc | 38 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
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/external/chromium_org/third_party/webrtc/modules/bitrate_controller/include/ |
H A D | bitrate_controller.h | 26 * to get the target bitrate. It also get the fraction loss and rtt to 34 const uint32_t rtt) = 0;
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | tester_main.cc | 33 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds."); 76 args.rtt = FLAGS_rtt;
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/external/chromium_org/net/dns/ |
H A D | dns_session.cc | 160 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);
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/external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/ |
H A D | packet_loss_test.cc | 169 int rtt = 0; local 171 rtt = _rttFrames * (1000 / _inst.maxFramerate); 173 rtt);
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/external/chromium_org/chrome/renderer/media/ |
H A D | cast_ipc_dispatcher.cc | 105 base::TimeDelta rtt) { 108 sender->OnRtt(ssrc, rtt); 103 OnRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt) argument
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H A D | cast_ipc_dispatcher.h | 52 void OnRtt(int32 channel_id, uint32 ssrc, base::TimeDelta rtt);
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