| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_packet_history_unittest.cc | 140 const int64_t kRttMs = 100; in TEST_P() local
142 hist_.SetRtt(kRttMs); in TEST_P()
162 fake_clock_.AdvanceTimeMilliseconds(kRttMs + 1); in TEST_P()
391 const int64_t kRttMs = RtpPacketHistory::kMinPacketDurationMs * 2; in TEST_P() local
393 kRttMs * RtpPacketHistory::kMinPacketDurationRtt; in TEST_P()
397 hist_.SetRtt(kRttMs); in TEST_P()
444 const int64_t kRttMs = RtpPacketHistory::kMinPacketDurationMs * 2; in TEST_P() local
447 hist_.SetRtt(kRttMs); in TEST_P()
452 int64_t kMaxPacketDurationMs = kRttMs * in TEST_P()
522 const int64_t kRttMs = RtpPacketHistory::kMinPacketDurationMs * 2; in TEST_P() local
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| D | rtp_sender_video_unittest.cc | 409 const int64_t kRttMs = (kFrameIntervalMs * 3) / 2; in TEST_P() local
425 rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs); in TEST_P()
435 rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
438 rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
442 EXPECT_TRUE(rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
449 EXPECT_TRUE(rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
453 EXPECT_TRUE(rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
459 rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs)); in TEST_P()
464 const int64_t kRttMs = (kFrameIntervalMs * 3) / 2; in TEST_P() local
481 rtp_sender_video_.AllowRetransmission(header, kSettings, kRttMs); in TEST_P()
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| D | rtcp_receiver_unittest.cc | 249 const int64_t kRttMs = 123; in TEST() local
258 mocks.clock.AdvanceTimeMilliseconds(kRttMs + kDelayMs); in TEST()
273 EXPECT_NEAR(kRttMs, rtt_ms, 1); in TEST()
281 const int64_t kRttMs = -13; in TEST() local
290 mocks.clock.AdvanceTimeMilliseconds(kRttMs + kDelayMs); in TEST()
315 const int64_t kRttMs = 120; in TEST() local
321 mocks.clock.AdvanceTimeMilliseconds(kRttMs + kDelayMs); in TEST()
336 OnReceivedRtcpReceiverReport(SizeIs(2), kRttMs, _)); in TEST()
972 const int64_t kRttMs = rand.Rand(1, 9 * 3600 * 1000); in TEST() local
978 mocks.clock.AdvanceTimeMilliseconds(kRttMs + kDelayMs); in TEST()
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| /external/webrtc/audio/test/ |
| D | audio_stats_test.cc | 31 const int64_t kRttMs = 100; member in webrtc::test::__anon1afff1ca0111::NoLossTest
37 pipe_config.queue_delay_ms = kRttMs / 2; in GetNetworkPipeConfig()
55 EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms); in OnStreamsStopped()
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | send_side_bandwidth_estimation_unittest.cc | 101 static const int64_t kRttMs = 50; in TEST() local
111 bwe.UpdateRtt(TimeDelta::Millis(kRttMs), Timestamp::Millis(now_ms)); in TEST()
122 EXPECT_EQ(kRttMs, bwe.round_trip_time().ms()); in TEST()
136 EXPECT_EQ(kRttMs, bwe.round_trip_time().ms()); in TEST()
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| /external/webrtc/modules/video_coding/ |
| D | jitter_estimator_tests.cc | 134 constexpr int64_t kRttMs = 250; in TEST_F() local
139 estimator_->UpdateRtt(kRttMs); // To test rtt_mult. in TEST_F()
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| D | frame_buffer2_unittest.cc | 451 constexpr int64_t kRttMs = 200; in TEST_F() local
452 buffer_->UpdateRtt(kRttMs); in TEST_F()
465 EXPECT_LT(timing_.GetCurrentJitter(), kRttMs); in TEST_F()
471 constexpr int64_t kRttMs = 200; in TEST_F() local
473 buffer_->UpdateRtt(kRttMs); in TEST_F()
487 EXPECT_GT(timing_.GetCurrentJitter(), kRttMs); in TEST_F()
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| /external/webrtc/video/ |
| D | receive_statistics_proxy_unittest.cc | 481 const int64_t kRttMs = 8; in TEST_F() local
482 statistics_proxy_->OnRttUpdate(kRttMs, 0); in TEST_F()
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| D | receive_statistics_proxy2_unittest.cc | 511 const int64_t kRttMs = 8; in TEST_F() local
512 statistics_proxy_->OnRttUpdate(kRttMs); in TEST_F()
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