/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
H A D | session_info.h | 25 int rtt_ms; member in struct:webrtc::FrameData 47 int rtt_ms);
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H A D | jitter_buffer.cc | 662 frame_data.rtt_ms = rtt_ms_; 840 void VCMJitterBuffer::UpdateRtt(uint32_t rtt_ms) { argument 842 rtt_ms_ = rtt_ms; 843 jitter_estimate_.UpdateRtt(rtt_ms);
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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | vcm_payload_sink_factory.cc | 115 uint32_t rtt_ms, 122 rtt_ms_(rtt_ms), 110 VcmPayloadSinkFactory( const std::string& base_out_filename, Clock* clock, bool protection_enabled, VCMVideoProtection protection_method, uint32_t rtt_ms, uint32_t render_delay_ms, uint32_t min_playout_delay_ms) argument
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H A D | rtp_player.cc | 74 LostPackets(Clock* clock, uint32_t rtt_ms) argument 80 rtt_ms_(rtt_ms) { 326 float loss_rate, uint32_t rtt_ms, bool reordering) 334 lost_packets_(clock, rtt_ms), 475 const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms, 488 payload_types, clock, &packet_source, loss_rate, rtt_ms, reordering)); 323 RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory, const PayloadTypes& payload_types, Clock* clock, scoped_ptr<test::RtpFileReader>* packet_source, float loss_rate, uint32_t rtt_ms, bool reordering) argument 473 Create(const std::string& input_filename, PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock, const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms, bool reordering) argument
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/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
H A D | video_engine_jni.cc | 597 int rtt_ms; local 601 jitter, rtt_ms) != 0) { 611 rtt_ms);
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtcp_receiver_unittest.cc | 599 uint16_t rtt_ms; local 600 EXPECT_FALSE(rtcp_receiver_->GetAndResetXrRrRtt(&rtt_ms));
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H A D | rtp_rtcp_impl_unittest.cc | 42 virtual void OnRttUpdate(uint32_t rtt_ms) { argument 43 rtt_ms_ = rtt_ms; 286 EXPECT_EQ(0U, sender_.impl_->rtt_ms()); 289 EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms()); 310 EXPECT_EQ(0U, receiver_.impl_->rtt_ms()); 313 EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
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H A D | rtcp_receiver.cc | 216 bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) { argument 217 assert(rtt_ms); 222 *rtt_ms = xr_rr_rtt_ms_;
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H A D | rtp_rtcp_impl.cc | 217 uint16_t rtt_ms; local 218 if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) { 219 rtt_stats_->OnRttUpdate(rtt_ms); 776 *rtt = static_cast<uint16_t>(rtt_ms()); 931 uint16_t rtt = rtt_ms(); 1278 uint16_t rtt = rtt_ms(); 1341 void ModuleRtpRtcpImpl::set_rtt_ms(uint32_t rtt_ms) { argument 1343 rtt_ms_ = rtt_ms; 1346 uint32_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
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/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
H A D | vie_autotest_custom_call.cc | 1513 int rtt_ms = 0; local 1522 rtt_ms); 1533 rtt_ms); 1548 << rtt_ms << std::endl;
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | mediachannel.h | 702 rtt_ms(0) { 737 int rtt_ms; member in struct:cricket::MediaSenderInfo
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/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_channel.cc | 1011 int32_t* rtt_ms) { 1058 *rtt_ms = rtt; 1080 int32_t* rtt_ms) { 1097 *rtt_ms = rtt; 1007 GetSendRtcpStatistics(uint16_t* fraction_lost, uint32_t* cumulative_lost, uint32_t* extended_max, uint32_t* jitter_samples, int32_t* rtt_ms) argument 1076 GetReceivedRtcpStatistics(uint16_t* fraction_lost, uint32_t* cumulative_lost, uint32_t* extended_max, uint32_t* jitter_samples, int32_t* rtt_ms) argument
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