| /external/webrtc/test/ |
| 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 | direct_transport.cc | 76 bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) { in SendRtcp() function in webrtc::test::DirectTransport
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| D | rtp_rtcp_observer.h | 121 bool SendRtcp(const uint8_t* packet, size_t length) override { in SendRtcp() function
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| /external/webrtc/video/ |
| D | transport_adapter.cc | 34 bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::internal::TransportAdapter
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| D | video_analyzer.cc | 313 bool VideoAnalyzer::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::VideoAnalyzer
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| /external/webrtc/media/base/ |
| D | media_channel.cc | 80 bool MediaChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, in SendRtcp() function in cricket::MediaChannel
209 void MediaChannel::SendRtcp(const uint8_t* data, size_t len) { in SendRtcp() function in cricket::MediaChannel
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| D | fake_network_interface.h | 141 virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet, in SendRtcp() function
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| D | fake_media_engine.h | 79 bool SendRtcp(const void* data, size_t len) { in SendRtcp() function
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| /external/webrtc/test/scenario/ |
| D | network_node.cc | 96 bool NetworkNodeTransport::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::test::NetworkNodeTransport
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| /external/webrtc/call/ |
| D | degraded_call.cc | 41 void DegradedCall::FakeNetworkPipeOnTaskQueue::SendRtcp(const uint8_t* packet, in SendRtcp() function in webrtc::DegradedCall::FakeNetworkPipeOnTaskQueue
124 bool DegradedCall::FakeNetworkPipeTransportAdapter::SendRtcp( in SendRtcp() function in webrtc::DegradedCall::FakeNetworkPipeTransportAdapter
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| D | fake_network_pipe.cc | 151 bool FakeNetworkPipe::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::FakeNetworkPipe
168 bool FakeNetworkPipe::SendRtcp(const uint8_t* packet, in SendRtcp() function in webrtc::FakeNetworkPipe
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| /external/webrtc/video/end_to_end_tests/ |
| D | network_state_tests.cc | 46 bool SendRtcp(const uint8_t* packet, size_t length) override { in SendRtcp() function in webrtc::NetworkStateEndToEndTest::UnusedTransport
73 bool SendRtcp(const uint8_t* packet, size_t length) override { in SendRtcp() function in webrtc::NetworkStateEndToEndTest::RequiredTransport
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| /external/webrtc/rtc_tools/rtp_generator/ |
| D | rtp_generator.cc | 283 bool RtpGenerator::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc::RtpGenerator
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender_audio_unittest.cc | 57 bool SendRtcp(const uint8_t* data, size_t len) override { return false; } in SendRtcp() function in webrtc::__anon9fc9e02b0111::LoopbackTransportTest
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| D | nack_rtx_unittest.cc | 105 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in webrtc::RtxLoopBackTransport
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| D | rtp_rtcp_impl_unittest.cc | 83 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in webrtc::__anonfb8e48c30111::SendTransport
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| D | rtp_sender_egress_unittest.cc | 108 bool SendRtcp(const uint8_t*, size_t) override { RTC_CHECK_NOTREACHED(); } in SendRtcp() function in webrtc::__anon6d3bba620111::TestTransport
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| D | rtcp_sender_unittest.cc | 62 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in webrtc::TestTransport
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| D | rtp_rtcp_impl2_unittest.cc | 108 bool SendRtcp(const uint8_t* data, size_t len) override { in SendRtcp() function in webrtc::__anon5f325df50111::SendTransport
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| D | rtcp_transceiver_impl_unittest.cc | 134 bool SendRtcp(const uint8_t* data, size_t size) override { in SendRtcp() function in webrtc::__anon922575930111::FakeRtcpTransport
163 bool SendRtcp(const uint8_t* data, size_t size) override { in SendRtcp() function in webrtc::__anon922575930111::RtcpParserTransport
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| D | rtp_sender_video_unittest.cc | 119 bool SendRtcp(const uint8_t* data, size_t len) override { return false; } in SendRtcp() function in webrtc::__anon2c2abd300111::LoopbackTransportTest
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| /external/openscreen/cast/streaming/ |
| D | receiver.cc | 324 void Receiver::SendRtcp() { in SendRtcp() function in openscreen::cast::Receiver
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| /external/webrtc/examples/androidvoip/jni/ |
| D | android_voip_client.cc | 449 bool AndroidVoipClient::SendRtcp(const uint8_t* packet, size_t length) { in SendRtcp() function in webrtc_examples::AndroidVoipClient
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| /external/webrtc/pc/ |
| D | channel.cc | 292 bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet, in SendRtcp() function in cricket::BaseChannel
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| /external/webrtc/media/engine/ |
| D | webrtc_voice_engine.cc | 2508 bool WebRtcVoiceMediaChannel::SendRtcp(const uint8_t* data, size_t len) { in SendRtcp() function in cricket::WebRtcVoiceMediaChannel
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