| /external/webrtc/webrtc/modules/bitrate_controller/include/mock/ |
| mock_bitrate_controller.h | 25 int64_t rtt_ms));
|
| /external/webrtc/webrtc/modules/bitrate_controller/ |
| send_side_bandwidth_estimation_unittest.cc | 61 int64_t rtt_ms; local
62 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
65 EXPECT_EQ(0, rtt_ms);
73 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
80 EXPECT_EQ(kRttMs, rtt_ms);
90 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms);
95 EXPECT_EQ(kRttMs, rtt_ms);
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| /external/webrtc/webrtc/modules/bitrate_controller/include/ |
| bitrate_controller.h | 38 int64_t rtt_ms) = 0;
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| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/estimators/ |
| send_side.cc | 59 int64_t rtt_ms = local
61 rbe_->OnRttUpdate(rtt_ms, rtt_ms);
62 BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
79 report_blocks, rtt_ms, clock_->TimeInMilliseconds());
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| nada.cc | 194 int64_t rtt_ms = now_ms - fb.latest_send_time_ms(); local
195 min_round_trip_time_ms_ = std::min(min_round_trip_time_ms_, rtt_ms);
233 observer_->OnNetworkChanged(1000 * bitrate_kbps_, 0, rtt_ms);
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| /external/webrtc/webrtc/ |
| audio_send_stream.h | 43 int64_t rtt_ms = -1; member in struct:webrtc::AudioSendStream::Stats
|
| call.h | 96 int64_t rtt_ms = -1; member in struct:webrtc::Call::Stats
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| /external/webrtc/webrtc/modules/video_coding/test/ |
| vcm_payload_sink_factory.h | 35 int64_t rtt_ms,
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| rtp_player.cc | 76 LostPackets(Clock* clock, int64_t rtt_ms)
82 rtt_ms_(rtt_ms) {
325 int64_t rtt_ms,
334 lost_packets_(clock, rtt_ms),
473 int64_t rtt_ms,
488 &packet_source, loss_rate, rtt_ms, reordering));
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| vcm_payload_sink_factory.cc | 103 int64_t rtt_ms,
110 rtt_ms_(rtt_ms),
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| /external/webrtc/webrtc/modules/video_coding/ |
| session_info.h | 25 int64_t rtt_ms; member in struct:webrtc::FrameData
47 int rtt_ms);
|
| decoding_state_unittest.cc | 42 frame_data.rtt_ms = 0;
171 frame_data.rtt_ms = 0;
221 frame_data.rtt_ms = 0;
375 frame_data.rtt_ms = 0;
404 frame_data.rtt_ms = 0;
428 frame_data.rtt_ms = 0;
466 frame_data.rtt_ms = 0;
509 frame_data.rtt_ms = 0;
564 frame_data.rtt_ms = 0;
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| jitter_buffer.cc | 738 frame_data.rtt_ms = rtt_ms_;
926 void VCMJitterBuffer::UpdateRtt(int64_t rtt_ms) {
928 rtt_ms_ = rtt_ms;
929 jitter_estimate_.UpdateRtt(rtt_ms);
[all...] |
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| bwe_test.h | 94 int64_t rtt_ms,
104 int64_t rtt_ms,
|
| bwe_test.cc | 250 int64_t rtt_ms,
254 capacity_kbps, max_delay_ms, rtt_ms, max_jitter_ms,
264 int64_t rtt_ms,
307 int64_t one_way_delay_ms = rtt_ms / 2;
640 int64_t rtt_ms = 2 * kOneWayDelayMs; local
649 // rtt_ms = 2 * 100;
658 kLinkCapacity, max_delay_ms, rtt_ms, kMaxJitterMs, offsets_ms,
752 int64_t rtt_ms = 2 * kOneWayDelayMs; local
755 // rtt_ms = 2 * 100;
764 kCapacityKbps, max_delay_ms, rtt_ms, kMaxJitterMs, kOffSetsMs
[all...] |
| /external/webrtc/webrtc/video/ |
| receive_statistics_proxy.h | 57 int64_t rtt_ms);
|
| receive_statistics_proxy.cc | 104 int64_t rtt_ms) {
116 delay_counter_.Add(target_delay_ms + rtt_ms / 2);
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| video_send_stream.cc | 531 int64_t rtt_ms; local
534 &jitter, &rtt_ms) == 0) {
535 return rtt_ms;
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl_unittest.cc | 46 void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; }
323 EXPECT_EQ(0, sender_.impl_->rtt_ms());
326 EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms());
347 EXPECT_EQ(0, receiver_.impl_->rtt_ms());
350 EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
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| rtp_rtcp_impl.cc | 172 int64_t rtt_ms; local
173 if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
174 rtt_stats_->OnRttUpdate(rtt_ms);
538 *rtt = rtt_ms();
734 int64_t rtt = rtt_ms();
920 int64_t rtt = rtt_ms();
979 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
981 rtt_ms_ = rtt_ms;
984 int64_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
|
| rtp_rtcp_impl.h | 353 void set_rtt_ms(int64_t rtt_ms);
354 int64_t rtt_ms() const;
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| conference_transport.h | 54 * rtt_ms : RTT in milliseconds.
56 void SetRtt(unsigned int rtt_ms);
|
| conference_transport.cc | 220 void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
221 rtt_ms_ = rtt_ms;
|
| /external/webrtc/webrtc/call/ |
| call.cc | 95 int64_t rtt_ms) override;
514 int rtt_ms = kv.second->GetRtt(); local
515 if (rtt_ms > 0)
516 stats.rtt_ms = rtt_ms;
573 int64_t rtt_ms) {
575 target_bitrate_bps, fraction_loss, rtt_ms);
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| /external/webrtc/webrtc/audio/ |
| audio_send_stream.cc | 139 stats.rtt_ms = call_stats.rttMs;
|
