| /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 | 665 frame_data.rtt_ms = rtt_ms_;
843 void VCMJitterBuffer::UpdateRtt(uint32_t rtt_ms) { argument
845 rtt_ms_ = rtt_ms;
846 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 | 75 LostPackets(Clock* clock, uint32_t rtt_ms) argument
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));
324 RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory, const PayloadTypes& payload_types, Clock* clock, scoped_ptr<RtpPacketSourceInterface>* packet_source, float loss_rate, uint32_t rtt_ms, bool reordering) argument
477 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/modules/rtp_rtcp/source/ |
| H A D | rtp_rtcp_impl_unittest.cc | 38 virtual void OnRttUpdate(uint32_t rtt_ms) { argument
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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| H A D | rtcp_receiver_unittest.cc | 503 uint16_t rtt_ms; local
504 EXPECT_FALSE(rtcp_receiver_->GetAndResetXrRrRtt(&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 | 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();
1239 uint16_t rtt = rtt_ms();
1301 void ModuleRtpRtcpImpl::set_rtt_ms(uint32_t rtt_ms) { argument
1303 rtt_ms_ = rtt_ms;
1306 uint32_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
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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/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/webrtc/video_engine/ |
| H A D | vie_channel.cc | 993 int32_t* rtt_ms) {
1038 *rtt_ms = rtt;
1060 int32_t* rtt_ms) {
1077 *rtt_ms = rtt;
989 GetSendRtcpStatistics(uint16_t* fraction_lost, uint32_t* cumulative_lost, uint32_t* extended_max, uint32_t* jitter_samples, int32_t* rtt_ms) argument
1056 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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| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| H A D | mediachannel.h | 729 rtt_ms(0) {
764 int rtt_ms; member in struct:cricket::MediaSenderInfo
|