/external/openscreen/cast/streaming/ |
D | sender_packet_router_unittest.cc | 88 Clock::time_point send_time, in MakeFakePacketWithFlag() 100 absl::Span<uint8_t> MakeFakePacket(Clock::time_point send_time, in MakeFakePacket() 126 absl::Span<uint8_t> ToEmptyPacketBuffer(Clock::time_point send_time, in ToEmptyPacketBuffer() 410 Invoke([&](Clock::time_point send_time, absl::Span<uint8_t> buffer) { in TEST_F() 523 Invoke([](Clock::time_point send_time, absl::Span<uint8_t> buffer) { in TEST_F() 529 Invoke([&](Clock::time_point send_time, absl::Span<uint8_t> buffer) { in TEST_F() 540 Invoke([](Clock::time_point send_time, absl::Span<uint8_t> buffer) { in TEST_F() 546 Invoke([&](Clock::time_point send_time, absl::Span<uint8_t> buffer) { in TEST_F()
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D | compound_rtcp_builder_unittest.cc | 64 const auto send_time = Clock::now(); in TEST_F() local 92 const auto send_time = Clock::now(); in TEST_F() local 137 auto send_time = Clock::now(); in TEST_F() local 188 const auto send_time = Clock::now(); in TEST_F() local 235 auto send_time = Clock::now(); in TEST_F() local
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D | compound_rtcp_builder.cc | 87 Clock::time_point send_time, in BuildPacket() 137 Clock::time_point send_time, in AppendReceiverReferenceTimeReportPacket()
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D | sender_packet_router.cc | 172 int SenderPacketRouter::SendJustTheRtcpPackets(Clock::time_point send_time) { in SendJustTheRtcpPackets() 198 int SenderPacketRouter::SendJustTheRtpPackets(Clock::time_point send_time, in SendJustTheRtpPackets()
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D | sender.cc | 191 Clock::time_point send_time, in GetRtcpPacketForImmediateSend() 213 Clock::time_point send_time, in GetRtpPacketForImmediateSend()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | absolute_capture_time_sender.cc | 48 const Timestamp send_time = clock_->CurrentTime(); in OnSendPacket() local 73 Timestamp send_time, in ShouldSendExtension()
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/external/webrtc/modules/audio_coding/neteq/ |
D | neteq_stereo_unittest.cc | 143 virtual int GetArrivalTime(int send_time) { in GetArrivalTime() 253 virtual int GetArrivalTime(int send_time) { in GetArrivalTime() 286 virtual int GetArrivalTime(int send_time) { in GetArrivalTime()
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D | neteq_network_stats_unittest.cc | 177 bool Lost(uint32_t send_time) { in Lost()
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/external/webrtc/call/ |
D | fake_network_pipe.h | 58 int64_t send_time() const { return send_time_; } in send_time() function
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D | fake_network_pipe.cc | 33 int64_t send_time, in NetworkPacket()
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/external/webrtc/modules/congestion_controller/rtp/ |
D | transport_feedback_adapter.cc | 115 auto send_time = Timestamp::Millis(sent_packet.send_time_ms); in ProcessSentPacket() local
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/external/webrtc/modules/remote_bitrate_estimator/ |
D | remote_bitrate_estimator_unittest_helper.h | 50 int64_t send_time; member
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/external/webrtc/modules/congestion_controller/goog_cc/ |
D | probe_bitrate_estimator_unittest.cc | 98 int64_t send_time = 0; in TEST_F() local
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/external/webrtc/modules/audio_coding/codecs/isac/fix/test/ |
D | kenny.cc | 54 uint32_t send_time; /* samples */ member
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/external/webrtc/modules/pacing/ |
D | pacing_controller.cc | 689 Timestamp send_time) { in OnPacketSent()
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/external/webrtc/media/base/ |
D | rtp_utils.cc | 71 uint32_t send_time = ((time_us << 18) / 1000000) & 0x00FFFFFF; in UpdateAbsSendTimeExtensionValue() local
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/external/webrtc/video/ |
D | rtp_video_stream_receiver2.cc | 653 uint32_t send_time; in OnRtpPacket() local
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D | rtp_video_stream_receiver.cc | 686 uint32_t send_time; in OnRtpPacket() local
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/external/webrtc/rtc_base/ |
D | virtual_socket_unittest.cc | 130 uint32_t send_time = *reinterpret_cast<const uint32_t*>(data); in OnReadPacket() local
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