| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| vcm_payload_sink_factory.h | 31 uint32_t rtt_ms, uint32_t render_delay_ms,
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| rtp_player.cc | 74 LostPackets(Clock* clock, uint32_t rtt_ms)
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));
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| vcm_payload_sink_factory.cc | 115 uint32_t rtt_ms,
122 rtt_ms_(rtt_ms),
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| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| session_info.h | 25 int rtt_ms; member in struct:webrtc::FrameData
47 int rtt_ms);
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| decoding_state_unittest.cc | 42 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;
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| jitter_buffer.h | 153 void UpdateRtt(uint32_t rtt_ms);
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| jitter_buffer.cc | 662 frame_data.rtt_ms = rtt_ms_;
840 void VCMJitterBuffer::UpdateRtt(uint32_t rtt_ms) {
842 rtt_ms_ = rtt_ms;
843 jitter_estimate_.UpdateRtt(rtt_ms);
[all...] |
| session_info.cc | 236 if (frame_data.rtt_ms < kRttThreshold
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| /external/chromium_org/third_party/webrtc/video_engine/include/ |
| vie_rtp_rtcp.h | 306 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);
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_rtp_rtcp_impl.h | 103 int& rtt_ms) const;
106 int& rtt_ms) const;
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| vie_channel.h | 177 int32_t* rtt_ms);
188 int32_t* rtt_ms);
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| vie_rtp_rtcp_impl.cc | 697 int& rtt_ms) const {
712 &rtt_ms) != 0) {
722 int& rtt_ms) const {
737 &rtt_ms) != 0) {
[all...] |
| vie_channel.cc | [all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl_unittest.cc | 42 virtual void OnRttUpdate(uint32_t rtt_ms) {
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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| 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();
1346 uint32_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
[all...] |
| rtp_rtcp_impl.h | 397 void set_rtt_ms(uint32_t rtt_ms);
398 uint32_t rtt_ms() const;
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| rtcp_receiver.h | 85 bool GetAndResetXrRrRtt(uint16_t* rtt_ms);
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| rtcp_receiver.cc | 216 bool RTCPReceiver::GetAndResetXrRrRtt(uint16_t* rtt_ms) {
217 assert(rtt_ms);
222 *rtt_ms = xr_rr_rtt_ms_;
[all...] |
| rtcp_receiver_unittest.cc | 599 uint16_t rtt_ms; local
600 EXPECT_FALSE(rtcp_receiver_->GetAndResetXrRrRtt(&rtt_ms));
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| /external/chromium_org/third_party/webrtc/examples/android/media_demo/jni/ |
| 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/video_engine/test/auto_test/source/ |
| vie_autotest_custom_call.cc | 1513 int rtt_ms = 0; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| mediachannel.h | 702 rtt_ms(0) {
737 int rtt_ms; member in struct:cricket::MediaSenderInfo
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| statscollector_unittest.cc | 343 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;
[all...] |
| statscollector.cc | 359 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);
445 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/interface/ |
| module_common_types.h | 281 virtual void OnRttUpdate(uint32_t rtt_ms) = 0;
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