/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/remote_bitrate_estimator/test/ |
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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D | bwe_test.h | 94 int64_t rtt_ms, 104 int64_t rtt_ms,
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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/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()
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