| /external/webrtc/rtc_base/ |
| D | async_udp_socket.cc | 68 rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(), in Send() local
80 rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(), in SendTo() local
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| D | async_tcp_socket.cc | 279 rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(), in Send() local
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| /external/cronet/net/third_party/quiche/src/quiche/quic/core/congestion_control/ |
| D | bandwidth_sampler.cc | 270 packet_number, [&](const ConnectionStateOnSentPacket& sent_packet) { in OnPacketNeutered()
397 const ConnectionStateOnSentPacket& sent_packet) { in OnPacketAcknowledgedInner()
541 const ConnectionStateOnSentPacket& sent_packet, in SentPacketToSendTimeState()
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| /external/webrtc/modules/congestion_controller/pcc/ |
| D | utility_function_unittest.cc | 49 SentPacket sent_packet; local
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| D | monitor_interval_unittest.cc | 35 SentPacket sent_packet; local
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| D | bitrate_controller_unittest.cc | 49 SentPacket sent_packet; local
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| /external/webrtc/p2p/base/ |
| D | async_stun_tcp_socket.cc | 86 rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis()); in Send() local
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| D | dtls_transport_unittest.cc | 256 const rtc::SentPacket& sent_packet) { in OnTransportSentPacket()
260 rtc::SentPacket sent_packet() const { return sent_packet_; } in sent_packet() function in cricket::DtlsTestClient
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| D | tcp_port.cc | 338 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| D | dtls_transport.cc | 671 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| D | stun_port.cc | 442 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| /external/webrtc/test/ |
| D | direct_transport.cc | 65 rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis()); in SendRtp() local
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| /external/webrtc/test/scenario/ |
| D | network_node.cc | 78 rtc::SentPacket sent_packet; in SendRtp() local
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| /external/webrtc/modules/congestion_controller/rtp/ |
| D | transport_feedback_adapter.cc | 113 const rtc::SentPacket& sent_packet) { in ProcessSentPacket()
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| D | transport_feedback_adapter_unittest.cc | 394 absl::optional<SentPacket> sent_packet = adapter_->ProcessSentPacket( in TEST_F() local
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| /external/webrtc/call/ |
| D | degraded_call.cc | 112 rtc::SentPacket sent_packet; in SendRtp() local
389 void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| D | rtp_transport_controller_send.cc | 402 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| /external/webrtc/rtc_tools/rtc_event_log_visualizer/ |
| D | log_simulation.cc | 95 rtc::SentPacket sent_packet; in OnPacketSent() local
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| /external/webrtc/pc/ |
| D | rtp_transport.cc | 225 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| D | rtp_transport_unittest.cc | 56 const rtc::SentPacket& sent_packet) { in OnSentPacket()
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| D | channel.cc | 808 void BaseChannel::SignalSentPacket_n(const rtc::SentPacket& sent_packet) { in SignalSentPacket_n()
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| /external/webrtc/api/transport/ |
| D | network_types.h | 163 SentPacket sent_packet; member
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| /external/webrtc/modules/congestion_controller/goog_cc/ |
| D | goog_cc_network_control.cc | 258 SentPacket sent_packet) { in OnSentPacket()
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| D | send_side_bandwidth_estimation.cc | 598 void SendSideBandwidthEstimation::OnSentPacket(const SentPacket& sent_packet) { in OnSentPacket()
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| /external/webrtc/media/base/ |
| D | fake_media_engine.h | 282 void OnPacketSent(const rtc::SentPacket& sent_packet) override {} in OnPacketSent()
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