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