| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_packet_history_unittest.cc | 142 hist_.SetRtt(kRttMs); in TEST_P()
173 hist_.SetRtt(kMinRetransmitIntervalMs); in TEST_P()
214 hist_.SetRtt(kMinRetransmitIntervalMs); in TEST_P()
305 hist_.SetRtt(1); in TEST_P()
397 hist_.SetRtt(kRttMs); in TEST_P()
447 hist_.SetRtt(kRttMs); in TEST_P()
508 hist_.SetRtt(1); // Trigger culling of old packets. in TEST_P()
515 hist_.SetRtt(1); // Trigger culling of old packets. in TEST_P()
523 hist_.SetRtt(kRttMs); in TEST_P()
565 hist_.SetRtt(kRttMs); in TEST_P()
[all …]
|
| D | rtp_packet_history.h | 75 void SetRtt(int64_t rtt_ms);
|
| D | rtp_packet_history.cc | 110 void RtpPacketHistory::SetRtt(int64_t rtt_ms) { in SetRtt() function in webrtc::RtpPacketHistory
|
| D | rtp_rtcp_impl2.cc | 719 rtp_sender_->packet_history.SetRtt(rtt_ms); in set_rtt_ms()
|
| D | rtp_rtcp_impl.cc | 811 rtp_sender_->packet_history.SetRtt(rtt_ms); in set_rtt_ms()
|
| D | rtp_sender.cc | 366 packet_history_->SetRtt(5 + avg_rtt); in OnReceivedNack()
|
| D | rtp_sender_unittest.cc | 2775 rtp_sender_context_->packet_history_.SetRtt(kRtt); in TEST_P()
2804 rtp_sender_context_->packet_history_.SetRtt(kRtt); in TEST_P()
|
| /external/webrtc/modules/audio_coding/audio_network_adaptor/include/ |
| D | audio_network_adaptor.h | 32 virtual void SetRtt(int rtt_ms) = 0;
|
| /external/webrtc/modules/audio_coding/audio_network_adaptor/mock/ |
| D | mock_audio_network_adaptor.h | 31 MOCK_METHOD(void, SetRtt, (int rtt_ms), (override));
|
| /external/webrtc/modules/audio_coding/audio_network_adaptor/ |
| D | audio_network_adaptor_impl.h | 51 void SetRtt(int rtt_ms) override;
|
| D | audio_network_adaptor_impl_unittest.cc | 145 states.audio_network_adaptor->SetRtt(kRtt); in TEST()
232 states.audio_network_adaptor->SetRtt(kRtt); in TEST()
|
| D | audio_network_adaptor_impl.cc | 75 void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) { in SetRtt() function in webrtc::AudioNetworkAdaptorImpl
|
| /external/webrtc/modules/remote_bitrate_estimator/ |
| D | aimd_rate_control.h | 55 void SetRtt(TimeDelta rtt);
|
| D | remote_bitrate_estimator_single_stream.cc | 221 GetRemoteRate()->SetRtt(TimeDelta::Millis(avg_rtt_ms)); in OnRttUpdate()
|
| D | aimd_rate_control_unittest.cc | 87 states.aimd_rate_control->SetRtt(TimeDelta::Millis(100)); in TEST()
|
| D | aimd_rate_control.cc | 179 void AimdRateControl::SetRtt(TimeDelta rtt) { in SetRtt() function in webrtc::AimdRateControl
|
| D | remote_bitrate_estimator_abs_send_time.cc | 395 remote_rate_.SetRtt(TimeDelta::Millis(avg_rtt_ms)); in OnRttUpdate()
|
| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | delay_based_bwe.cc | 336 rate_control_.SetRtt(avg_rtt); in OnRttUpdate()
|
| /external/webrtc/modules/audio_coding/codecs/opus/ |
| D | audio_encoder_opus.cc | 549 audio_network_adaptor_->SetRtt(rtt_ms); in OnReceivedRtt()
|
| D | audio_encoder_opus_unittest.cc | 312 EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt)); in TEST_P()
|