| /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 | 665 frame_data.rtt_ms = rtt_ms_;
843 void VCMJitterBuffer::UpdateRtt(uint32_t rtt_ms) {
845 rtt_ms_ = rtt_ms;
846 jitter_estimate_.UpdateRtt(rtt_ms);
[all...] |
| session_info.cc | 193 if (frame_data.rtt_ms < kRttThreshold
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| /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 | 75 LostPackets(Clock* clock, uint32_t rtt_ms)
81 rtt_ms_(rtt_ms) {
327 float loss_rate, uint32_t rtt_ms, bool reordering)
335 lost_packets_(clock, rtt_ms),
479 const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms,
491 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/video_engine/include/ |
| vie_rtp_rtcp.h | 293 int& rtt_ms) const = 0;
299 int& rtt_ms) const = 0;
308 int& rtt_ms) const {
312 rtt_ms);
324 int& rtt_ms) const {
328 rtt_ms);
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_rtp_rtcp_impl.h | 100 int& rtt_ms) const;
103 int& rtt_ms) const;
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| vie_channel.h | 176 int32_t* rtt_ms);
187 int32_t* rtt_ms);
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| vie_rtp_rtcp_impl.cc | 688 int& rtt_ms) const {
703 &rtt_ms) != 0) {
713 int& rtt_ms) const {
728 &rtt_ms) != 0) {
[all...] |
| vie_channel.cc | [all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_rtcp_impl_unittest.cc | 38 virtual void OnRttUpdate(uint32_t rtt_ms) {
39 rtt_ms_ = rtt_ms;
167 EXPECT_EQ(0U, sender_.impl_->rtt_ms());
170 EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms());
191 EXPECT_EQ(0U, receiver_.impl_->rtt_ms());
194 EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
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| rtp_rtcp_impl.cc | 212 uint16_t rtt_ms; local
213 if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
214 rtt_stats_->OnRttUpdate(rtt_ms);
725 *rtt = static_cast<uint16_t>(rtt_ms());
875 uint16_t rtt = rtt_ms();
1306 uint32_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
[all...] |
| rtp_rtcp_impl.h | 409 void set_rtt_ms(uint32_t rtt_ms);
410 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 | 503 uint16_t rtt_ms; local
504 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 | 729 rtt_ms(0) {
764 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(talk_base::ToString<int>(sinfo.rtt_ms), value_in_report);
346 EXPECT_EQ(talk_base::ToString<int>(sinfo.rtt_ms), value_in_report);
386 voice_sender_info->rtt_ms = 102;
[all...] |
| statscollector.cc | 331 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);
417 report->AddValue(StatsReport::kStatsValueNameRtt, info.rtt_ms);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/interface/ |
| module_common_types.h | 289 virtual void OnRttUpdate(uint32_t rtt_ms) = 0;
|
