/external/chromium_org/third_party/webrtc/video_engine/include/ |
D | vie_rtp_rtcp.h | 306 int& rtt_ms) const = 0; 315 int& rtt_ms) const = 0; 324 int& rtt_ms) const { in GetReceivedRTCPStatistics() argument 328 rtt_ms); in GetReceivedRTCPStatistics() 340 int& rtt_ms) const { in GetSentRTCPStatistics() argument 344 rtt_ms); in GetSentRTCPStatistics()
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
D | session_info.h | 25 int 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()
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D | jitter_buffer.cc | 662 frame_data.rtt_ms = rtt_ms_; in InsertPacket() 840 void VCMJitterBuffer::UpdateRtt(uint32_t rtt_ms) { in UpdateRtt() argument 842 rtt_ms_ = rtt_ms; in UpdateRtt() 843 jitter_estimate_.UpdateRtt(rtt_ms); in UpdateRtt()
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D | jitter_buffer.h | 153 void UpdateRtt(uint32_t rtt_ms);
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D | session_info.cc | 236 if (frame_data.rtt_ms < kRttThreshold in UpdateDecodableSession()
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D | session_info_unittest.cc | 32 frame_data.rtt_ms = 0; in SetUp() 248 frame_data.rtt_ms = 150; in TEST_F()
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
D | rtp_player.cc | 74 LostPackets(Clock* clock, uint32_t rtt_ms) in LostPackets() argument 80 rtt_ms_(rtt_ms) { in LostPackets() 326 float loss_rate, uint32_t rtt_ms, bool reordering) in RtpPlayerImpl() argument 334 lost_packets_(clock, rtt_ms), in RtpPlayerImpl() 475 const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms, in Create() argument 488 payload_types, clock, &packet_source, loss_rate, rtt_ms, reordering)); in Create()
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D | vcm_payload_sink_factory.cc | 115 uint32_t rtt_ms, in VcmPayloadSinkFactory() argument 122 rtt_ms_(rtt_ms), in VcmPayloadSinkFactory()
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D | vcm_payload_sink_factory.h | 31 uint32_t rtt_ms, uint32_t render_delay_ms,
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
D | rtp_rtcp_impl_unittest.cc | 42 virtual void OnRttUpdate(uint32_t rtt_ms) { in OnRttUpdate() argument 43 rtt_ms_ = rtt_ms; in OnRttUpdate() 286 EXPECT_EQ(0U, sender_.impl_->rtt_ms()); in TEST_F() 289 EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms()); in TEST_F() 310 EXPECT_EQ(0U, receiver_.impl_->rtt_ms()); in TEST_F() 313 EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms()); in TEST_F()
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D | rtp_rtcp_impl.cc | 217 uint16_t rtt_ms; in Process() local 218 if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) { in Process() 219 rtt_stats_->OnRttUpdate(rtt_ms); in Process() 776 *rtt = static_cast<uint16_t>(rtt_ms()); in RTT() 931 uint16_t rtt = rtt_ms(); in SendNACK() 1278 uint16_t rtt = rtt_ms(); in OnReceivedNACK() 1341 void ModuleRtpRtcpImpl::set_rtt_ms(uint32_t rtt_ms) { in set_rtt_ms() argument 1343 rtt_ms_ = rtt_ms; in set_rtt_ms() 1346 uint32_t ModuleRtpRtcpImpl::rtt_ms() const { in rtt_ms() function in webrtc::ModuleRtpRtcpImpl
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D | rtp_rtcp_impl.h | 397 void set_rtt_ms(uint32_t rtt_ms); 398 uint32_t rtt_ms() const;
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D | rtcp_receiver.h | 85 bool GetAndResetXrRrRtt(uint16_t* rtt_ms);
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D | rtcp_receiver.cc | 216 bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) { in GetAndResetXrRrRtt() argument 217 assert(rtt_ms); in GetAndResetXrRrRtt() 222 *rtt_ms = xr_rr_rtt_ms_; in GetAndResetXrRrRtt()
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D | rtcp_receiver_unittest.cc | 599 uint16_t rtt_ms; in TEST_F() local 600 EXPECT_FALSE(rtcp_receiver_->GetAndResetXrRrRtt(&rtt_ms)); in TEST_F()
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/external/chromium_org/third_party/webrtc/video_engine/ |
D | vie_rtp_rtcp_impl.h | 103 int& rtt_ms) const; 106 int& rtt_ms) const;
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D | vie_rtp_rtcp_impl.cc | 697 int& rtt_ms) const { in GetReceiveChannelRtcpStatistics() 712 &rtt_ms) != 0) { in GetReceiveChannelRtcpStatistics() 722 int& rtt_ms) const { in GetSendChannelRtcpStatistics() 737 &rtt_ms) != 0) { in GetSendChannelRtcpStatistics()
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D | vie_channel.h | 177 int32_t* rtt_ms); 188 int32_t* rtt_ms);
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D | vie_channel.cc | 1011 int32_t* rtt_ms) { in GetSendRtcpStatistics() argument 1058 *rtt_ms = rtt; in GetSendRtcpStatistics() 1080 int32_t* rtt_ms) { in GetReceivedRtcpStatistics() argument 1097 *rtt_ms = rtt; in GetReceivedRtcpStatistics()
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
D | video_engine_jni.cc | 597 int rtt_ms; in JOWW() local 601 jitter, rtt_ms) != 0) { in JOWW() 611 rtt_ms); in JOWW()
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
D | vie_autotest_custom_call.cc | 1513 int rtt_ms = 0; in PrintRTCCPStatistics() local 1522 rtt_ms); in PrintRTCCPStatistics() 1533 rtt_ms); in PrintRTCCPStatistics() 1548 << rtt_ms << std::endl; in PrintRTCCPStatistics()
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
D | mediachannel.h | 702 rtt_ms(0) { in MediaSenderInfo() 737 int rtt_ms; member
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
D | statscollector_unittest.cc | 343 EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report); in VerifyVoiceSenderInfoReport() 346 EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report); in VerifyVoiceSenderInfoReport() 386 voice_sender_info->rtt_ms = 102; in InitVoiceSenderInfo()
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D | statscollector.cc | 359 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms); in ExtractStats() 445 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms); in ExtractStats()
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