/external/webrtc/webrtc/modules/bitrate_controller/include/mock/ |
H A D | mock_bitrate_controller.h | 25 int64_t rtt_ms));
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/external/webrtc/webrtc/modules/bitrate_controller/ |
H A D | send_side_bandwidth_estimation_unittest.cc | 61 int64_t rtt_ms; local 62 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 65 EXPECT_EQ(0, rtt_ms); 73 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 80 EXPECT_EQ(kRttMs, rtt_ms); 90 bwe.CurrentEstimate(&bitrate_bps, &fraction_loss, &rtt_ms); 95 EXPECT_EQ(kRttMs, rtt_ms);
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/estimators/ |
H A D | send_side.cc | 59 int64_t rtt_ms = local 61 rbe_->OnRttUpdate(rtt_ms, rtt_ms); 62 BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms); 79 report_blocks, rtt_ms, clock_->TimeInMilliseconds());
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H A D | nada.cc | 194 int64_t rtt_ms = now_ms - fb.latest_send_time_ms(); local 195 min_round_trip_time_ms_ = std::min(min_round_trip_time_ms_, rtt_ms); 233 observer_->OnNetworkChanged(1000 * bitrate_kbps_, 0, rtt_ms);
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/external/webrtc/webrtc/modules/bitrate_controller/include/ |
H A D | bitrate_controller.h | 38 int64_t rtt_ms) = 0;
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/external/webrtc/webrtc/modules/video_coding/ |
H A D | session_info.h | 25 int64_t rtt_ms; member in struct:webrtc::FrameData 47 int rtt_ms);
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H A D | 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; 466 frame_data.rtt_ms = 0; 509 frame_data.rtt_ms = 0; 564 frame_data.rtt_ms = 0;
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H A D | jitter_buffer.cc | 738 frame_data.rtt_ms = rtt_ms_; 926 void VCMJitterBuffer::UpdateRtt(int64_t rtt_ms) { argument 928 rtt_ms_ = rtt_ms; 929 jitter_estimate_.UpdateRtt(rtt_ms);
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/external/webrtc/webrtc/ |
H A D | audio_send_stream.h | 43 int64_t rtt_ms = -1; member in struct:webrtc::AudioSendStream::Stats
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H A D | call.h | 96 int64_t rtt_ms = -1; member in struct:webrtc::Call::Stats
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/external/webrtc/webrtc/modules/video_coding/test/ |
H A D | vcm_payload_sink_factory.h | 35 int64_t rtt_ms,
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H A D | rtp_player.cc | 76 LostPackets(Clock* clock, int64_t rtt_ms) argument 82 rtt_ms_(rtt_ms) { 325 int64_t rtt_ms, 334 lost_packets_(clock, rtt_ms), 473 int64_t rtt_ms, 488 &packet_source, loss_rate, rtt_ms, reordering)); 320 RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory, const PayloadTypes& payload_types, Clock* clock, rtc::scoped_ptr<test::RtpFileReader>* packet_source, float loss_rate, int64_t rtt_ms, bool reordering) argument 468 Create(const std::string& input_filename, PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock, const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms, bool reordering) argument
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H A D | vcm_payload_sink_factory.cc | 103 int64_t rtt_ms, 110 rtt_ms_(rtt_ms), 98 VcmPayloadSinkFactory( const std::string& base_out_filename, Clock* clock, bool protection_enabled, VCMVideoProtection protection_method, int64_t rtt_ms, uint32_t render_delay_ms, uint32_t min_playout_delay_ms) argument
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/external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
H A D | bwe_test.h | 94 int64_t rtt_ms, 104 int64_t rtt_ms,
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H A D | bwe_test.cc | 250 int64_t rtt_ms, 254 capacity_kbps, max_delay_ms, rtt_ms, max_jitter_ms, 264 int64_t rtt_ms, 307 int64_t one_way_delay_ms = rtt_ms / 2; 640 int64_t rtt_ms = 2 * kOneWayDelayMs; local 649 // rtt_ms = 2 * 100; 658 kLinkCapacity, max_delay_ms, rtt_ms, kMaxJitterMs, offsets_ms, 752 int64_t rtt_ms = 2 * kOneWayDelayMs; local 755 // rtt_ms = 2 * 100; 764 kCapacityKbps, max_delay_ms, rtt_ms, kMaxJitterM 244 RunFairnessTest(BandwidthEstimatorType bwe_type, size_t num_media_flows, size_t num_tcp_flows, int64_t run_time_seconds, uint32_t capacity_kbps, int64_t max_delay_ms, int64_t rtt_ms, int64_t max_jitter_ms, const int64_t* offsets_ms) argument 258 RunFairnessTest(BandwidthEstimatorType bwe_type, size_t num_media_flows, size_t num_tcp_flows, int64_t run_time_seconds, uint32_t capacity_kbps, int64_t max_delay_ms, int64_t rtt_ms, int64_t max_jitter_ms, const int64_t* offsets_ms, const std::string& title, const std::string& flow_name) argument [all...] |
/external/webrtc/webrtc/video/ |
H A D | receive_statistics_proxy.h | 57 int64_t rtt_ms);
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H A D | receive_statistics_proxy.cc | 104 int64_t rtt_ms) { 116 delay_counter_.Add(target_delay_ms + rtt_ms / 2); 97 OnDecoderTiming(int decode_ms, int max_decode_ms, int current_delay_ms, int target_delay_ms, int jitter_buffer_ms, int min_playout_delay_ms, int render_delay_ms, int64_t rtt_ms) argument
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H A D | video_send_stream.cc | 531 int64_t rtt_ms; local 534 &jitter, &rtt_ms) == 0) { 535 return rtt_ms;
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_rtcp_impl_unittest.cc | 46 void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; } 323 EXPECT_EQ(0, sender_.impl_->rtt_ms()); 326 EXPECT_EQ(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms()); 347 EXPECT_EQ(0, receiver_.impl_->rtt_ms()); 350 EXPECT_EQ(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms());
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H A D | rtp_rtcp_impl.cc | 172 int64_t rtt_ms; local 173 if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) { 174 rtt_stats_->OnRttUpdate(rtt_ms); 538 *rtt = rtt_ms(); 734 int64_t rtt = rtt_ms(); 920 int64_t rtt = rtt_ms(); 979 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) { argument 981 rtt_ms_ = rtt_ms; 984 int64_t ModuleRtpRtcpImpl::rtt_ms() const { function in class:webrtc::ModuleRtpRtcpImpl
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H A D | rtp_rtcp_impl.h | 353 void set_rtt_ms(int64_t rtt_ms); 354 int64_t rtt_ms() const;
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
H A D | conference_transport.h | 54 * rtt_ms : RTT in milliseconds. 56 void SetRtt(unsigned int rtt_ms);
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H A D | conference_transport.cc | 220 void ConferenceTransport::SetRtt(unsigned int rtt_ms) { argument 221 rtt_ms_ = rtt_ms;
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/external/webrtc/webrtc/call/ |
H A D | call.cc | 95 int64_t rtt_ms) override; 514 int rtt_ms = kv.second->GetRtt(); local 515 if (rtt_ms > 0) 516 stats.rtt_ms = rtt_ms; 573 int64_t rtt_ms) { 575 target_bitrate_bps, fraction_loss, rtt_ms); 572 OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss, int64_t rtt_ms) argument
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/external/webrtc/webrtc/audio/ |
H A D | audio_send_stream.cc | 139 stats.rtt_ms = call_stats.rttMs;
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