/external/webrtc/webrtc/call/ |
D | transport_adapter.cc | 32 bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::internal::TransportAdapter 36 return transport_->SendRtcp(packet, length); in SendRtcp()
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D | transport_adapter.h | 27 bool SendRtcp(const uint8_t* packet, size_t length) override;
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/external/webrtc/webrtc/test/ |
D | rtp_rtcp_observer.h | 113 bool SendRtcp(const uint8_t* packet, size_t length) override { in SendRtcp() function 128 return test::DirectTransport::SendRtcp(packet, length); in SendRtcp()
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D | null_transport.cc | 21 bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::test::NullTransport
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D | mock_transport.h | 25 MOCK_METHOD2(SendRtcp, bool(const uint8_t* data, size_t len));
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D | null_transport.h | 25 bool SendRtcp(const uint8_t* packet, size_t length) override;
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D | direct_transport.h | 47 bool SendRtcp(const uint8_t* data, size_t length) override;
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D | direct_transport.cc | 67 bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) { in SendRtcp() function in webrtc::test::DirectTransport
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/external/webrtc/webrtc/ |
D | transport.h | 33 virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api.h | 44 bool SendRtcp(const uint8_t* data, size_t len) override;
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D | test_api.cc | 62 bool LoopBackTransport::SendRtcp(const uint8_t* data, size_t len) { in SendRtcp() function in webrtc::LoopBackTransport
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/external/webrtc/talk/media/webrtc/ |
D | webrtcvoiceengine.h | 214 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function 217 return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions()); in SendRtcp()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
D | conference_transport.h | 104 bool SendRtcp(const uint8_t *data, size_t len) override;
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D | conference_transport.cc | 116 bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) { in SendRtcp() function in voetest::ConferenceTransport
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/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
D | rtp_rtcp_extensions.cc | 56 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in ExtensionVerifyTransport
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtcp_format_remb_unittest.cc | 37 bool SendRtcp(const uint8_t* packet, size_t packetLength) override { in SendRtcp() function in webrtc::__anon1faa48600111::TestTransport
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D | nack_rtx_unittest.cc | 152 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in webrtc::RtxLoopBackTransport
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D | rtcp_sender.cc | 94 if (transport_->SendRtcp(data, length)) in OnPacketReady() 1049 if (!transport_->SendRtcp(data, length)) in SendFeedbackPacket()
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/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
D | after_initialization_fixture.h | 49 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function
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/external/webrtc/talk/media/base/ |
D | fakenetworkinterface.h | 157 virtual bool SendRtcp(rtc::Buffer* packet, in SendRtcp() function
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D | mediachannel.h | 461 virtual bool SendRtcp(rtc::Buffer* packet, 510 bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) { in SendRtcp() function 548 : network_interface_->SendRtcp(packet, options); in DoSendPacket()
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D | fakemediaengine.h | 81 bool SendRtcp(const void* data, int len) { in SendRtcp() function 84 return Base::SendRtcp(&packet, rtc::PacketOptions()); in SendRtcp()
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/external/webrtc/webrtc/test/channel_transport/ |
D | udp_transport_impl.h | 122 bool SendRtcp(const uint8_t* data, size_t length) override;
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/external/webrtc/talk/session/media/ |
D | channel_unittest.cc | 305 return media_channel1_->SendRtcp(rtcp_packet_.c_str(), in SendRtcp1() 309 return media_channel2_->SendRtcp(rtcp_packet_.c_str(), in SendRtcp2() 325 return media_channel1_->SendRtcp(data.c_str(), in SendCustomRtcp1() 330 return media_channel2_->SendRtcp(data.c_str(), in SendCustomRtcp2() 963 T::MediaChannel::SendRtcp(kRtcpReport, sizeof(kRtcpReport)); in TestCallTeardownRtcpMux()
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/external/webrtc/webrtc/video/ |
D | video_quality_test.cc | 177 bool SendRtcp(const uint8_t* packet, size_t length) override { in SendRtcp() function in webrtc::VideoAnalyzer 178 return transport_->SendRtcp(packet, length); in SendRtcp()
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