Searched refs:rtt_ms (Results 1 - 25 of 31) sorted by relevance

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/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/
H A Dvcm_payload_sink_factory.h31 uint32_t rtt_ms, uint32_t render_delay_ms,
H A Drtp_player.cc74 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
H A Dvcm_payload_sink_factory.cc115 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
/external/chromium_org/third_party/webrtc/modules/video_coding/main/source/
H A Dsession_info.h25 int rtt_ms; member in struct:webrtc::FrameData
47 int rtt_ms);
H A Ddecoding_state_unittest.cc42 frame_data.rtt_ms = 0;
171 frame_data.rtt_ms = 0;
221 frame_data.rtt_ms = 0;
375 frame_data.rtt_ms = 0;
404 frame_data.rtt_ms = 0;
428 frame_data.rtt_ms = 0;
H A Djitter_buffer.cc662 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);
H A Djitter_buffer.h153 void UpdateRtt(uint32_t rtt_ms);
H A Dsession_info.cc236 if (frame_data.rtt_ms < kRttThreshold
H A Dsession_info_unittest.cc32 frame_data.rtt_ms = 0;
248 frame_data.rtt_ms = 150;
/external/chromium_org/third_party/webrtc/video_engine/include/
H A Dvie_rtp_rtcp.h306 int& rtt_ms) const = 0;
315 int& rtt_ms) const = 0;
324 int& rtt_ms) const {
328 rtt_ms);
340 int& rtt_ms) const {
344 rtt_ms);
/external/chromium_org/third_party/webrtc/video_engine/
H A Dvie_rtp_rtcp_impl.h103 int& rtt_ms) const;
106 int& rtt_ms) const;
H A Dvie_channel.h177 int32_t* rtt_ms);
188 int32_t* rtt_ms);
H A Dvie_rtp_rtcp_impl.cc697 int& rtt_ms) const {
712 &rtt_ms) != 0) {
722 int& rtt_ms) const {
737 &rtt_ms) != 0) {
H A Dvie_channel.cc1011 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
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/
H A Drtp_rtcp_impl_unittest.cc42 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());
H A Drtp_rtcp_impl.cc217 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
H A Drtp_rtcp_impl.h397 void set_rtt_ms(uint32_t rtt_ms);
398 uint32_t rtt_ms() const;
H A Drtcp_receiver.h85 bool GetAndResetXrRrRtt(uint16_t* rtt_ms);
H A Drtcp_receiver.cc216 bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) { argument
217 assert(rtt_ms);
222 *rtt_ms = xr_rr_rtt_ms_;
H A Drtcp_receiver_unittest.cc599 uint16_t rtt_ms; local
600 EXPECT_FALSE(rtcp_receiver_->GetAndResetXrRrRtt(&rtt_ms));
/external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/
H A Dvideo_engine_jni.cc597 int rtt_ms; local
601 jitter, rtt_ms) != 0) {
611 rtt_ms);
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/
H A Dvie_autotest_custom_call.cc1513 int rtt_ms = 0; local
1522 rtt_ms);
1533 rtt_ms);
1548 << rtt_ms << std::endl;
/external/chromium_org/third_party/libjingle/source/talk/media/base/
H A Dmediachannel.h702 rtt_ms(0) {
737 int rtt_ms; member in struct:cricket::MediaSenderInfo
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/
H A Dstatscollector_unittest.cc343 EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report);
346 EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report);
386 voice_sender_info->rtt_ms = 102;
H A Dstatscollector.cc359 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);
445 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);

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