| /external/u-boot/drivers/rtc/ |
| at91sam9_rtt.c | 11 * The RTT cannot be written to, but only reset.
12 * The actual time is the sum of RTT and one of
34 at91_rtt_t *rtt = (at91_rtt_t *) ATMEL_BASE_RTT; local
41 tim = readl(&rtt->vr);
42 tim2 = readl(&rtt->vr);
52 at91_rtt_t *rtt = (at91_rtt_t *) ATMEL_BASE_RTT; local
59 writel(32768+AT91_RTT_RTTRST, &rtt->mr);
60 writel(~0, &rtt->ar);
63 while (readl(&rtt->vr) != 0)
70 at91_rtt_t *rtt = (at91_rtt_t *) ATMEL_BASE_RTT local
[all...] |
| /external/webrtc/webrtc/video/ |
| call_stats.h | 47 // Helper struct keeping track of the time a rtt value is reported.
50 : rtt(new_rtt), time(rtt_time) {}
51 const int64_t rtt; member in struct:webrtc::CallStats::RttTime
56 void OnRttUpdate(int64_t rtt);
68 // The last RTT in the statistics update (zero if there is no valid estimate).
72 // All Rtt reports within valid time interval, oldest first.
|
| call_stats.cc | 25 // Weight factor to apply to the average rtt.
29 // A rtt report is considered valid for this long.
41 max_rtt_ms = std::max(it->rtt, max_rtt_ms);
53 sum += it->rtt;
79 virtual void OnRttUpdate(int64_t rtt) {
80 owner_->OnRttUpdate(rtt);
83 // Returns the average RTT.
122 // If there is a valid rtt, update all observers with the max rtt.
123 // TODO(asapersson): Consider changing this to report the average rtt
[all...] |
| video_encoder.cc | 155 int64_t rtt) {
158 rtt_ = rtt;
159 int32_t ret = encoder_->SetChannelParameters(packet_loss, rtt);
161 return fallback_encoder_->SetChannelParameters(packet_loss, rtt);
|
| /external/syzkaller/vendor/google.golang.org/grpc/transport/ |
| bdp_estimator.go | 66 rtt float64
70 // network rtt can be calculated when its ack is received.
112 // Bootstrap rtt with an average of first 10 rtt samples.
113 b.rtt += (rttSample - b.rtt) / float64(b.sampleCount)
116 b.rtt += (rttSample - b.rtt) * float64(alpha)
121 bwCurrent := float64(b.sample) / (b.rtt * float64(1.5))
|
| /external/webrtc/webrtc/modules/bitrate_controller/ |
| bitrate_controller_impl.cc | 35 int64_t rtt,
70 owner_->OnReceivedRtcpReceiverReport(fraction_lost_aggregate, rtt,
169 int64_t rtt,
174 bandwidth_estimation_.UpdateReceiverBlock(fraction_loss, rtt,
183 int64_t rtt; local
184 if (GetNetworkParameters(&bitrate, &fraction_loss, &rtt))
185 observer_->OnNetworkChanged(bitrate, fraction_loss, rtt);
190 int64_t* rtt) {
193 bandwidth_estimation_.CurrentEstimate(¤t_bitrate, fraction_loss, rtt);
201 *rtt != last_rtt_ms_ |
216 int64_t rtt; local
[all...] |
| bitrate_controller_impl.h | 56 int64_t rtt,
65 int64_t* rtt);
69 int64_t rtt) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
|
| send_side_bandwidth_estimation.h | 30 void CurrentEstimate(int* bitrate, uint8_t* loss, int64_t* rtt) const;
40 int64_t rtt,
55 void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
|
| send_side_bandwidth_estimation_unittest.cc | 35 int64_t rtt; local
36 bwe.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
44 bwe.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
|
| send_side_bandwidth_estimation.cc | 95 int64_t* rtt) const {
98 *rtt = last_round_trip_time_ms_;
108 int64_t rtt,
114 // Update RTT.
115 last_round_trip_time_ms_ = rtt;
139 UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
143 int64_t rtt,
161 RTC_HISTOGRAM_COUNTS_SPARSE("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
217 // rtt.
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| remote_ntp_time_estimator.h | 32 // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
34 bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator_unittest.cc | 57 void UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
60 estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac,
64 void ReceiveRtcpSr(int64_t rtt,
68 UpdateRtcpTimestamp(rtt, ntp_seconds, ntp_fractions, rtcp_timestamp, true);
|
| remote_ntp_time_estimator.cc | 31 bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(int64_t rtt,
48 int64_t sender_arrival_time_90k = (sender_send_time_ms + rtt / 2) * 90;
|
| 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...] |
| /external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
| reference_picture_selection.h | 55 // Set the round-trip time between the sender and the receiver to |rtt|
57 void SetRtt(int64_t rtt);
|
| reference_picture_selection.cc | 118 void ReferencePictureSelection::SetRtt(int64_t rtt) {
120 rtt_ = 90 * rtt;
|
| /bionic/libc/dns/include/ |
| resolv_stats.h | 36 uint16_t rtt; // round-trip time in ms member in struct:__res_sample
55 _res_stats_set_sample(struct __res_sample* sample, time_t now, int rcode, int rtt);
|
| /external/libnl/lib/idiag/ |
| idiag_vegasinfo_obj.c | 69 void idiagnl_vegasinfo_set_rtt(struct idiagnl_vegasinfo *vinfo, uint32_t rtt)
71 vinfo->tcpv_rtt = rtt;
|
| /external/webrtc/webrtc/modules/video_coding/test/ |
| tester_main.cc | 22 DEFINE_int32(rtt, 0, "RTT (round-trip time), in milliseconds.");
60 args->rtt = FLAGS_rtt;
|
| test_util.h | 81 int rtt; member in class:CmdArgs
|
| /bionic/libc/dns/resolv/ |
| res_stats.c | 39 _res_stats_set_sample(struct __res_sample* sample, time_t now, int rcode, int rtt)
42 async_safe_format_log(ANDROID_LOG_INFO, "libc", "rcode = %d, sec = %d", rcode, rtt);
46 sample->rtt = rtt;
81 rtt_sum += stats->samples[i].rtt;
102 /* If there was at least one successful sample, calculate average RTT. */
133 "= %d, rtt = %d, min_samples = %d\n", successes, errors, timeouts, internal_errors,
|
| /external/webrtc/webrtc/modules/video_coding/codecs/h264/ |
| h264_video_toolbox_encoder.h | 45 int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
|
| /external/webrtc/webrtc/test/ |
| configurable_frame_size_encoder.h | 40 int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
|
| /external/webrtc/webrtc/modules/video_coding/ |
| media_opt_util.h | 49 : rtt(0),
62 int64_t rtt; member in struct:webrtc::media_optimization::VCMProtectionParameters
240 // - rtt : Round-trip time in seconds.
241 void UpdateRtt(int64_t rtt);
|
| /external/webrtc/webrtc/modules/video_coding/include/ |
| video_coding.h | 153 // - rtt : Current round-trip time in ms.
159 int64_t rtt) = 0;
166 // - rtt : Current round-trip time in ms.
173 virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
|
