/external/webrtc/talk/session/media/ |
D | channel_unittest.cc | 127 enum Flags { RTCP = 0x1, RTCP_MUX = 0x2, SECURE = 0x4, SSRC_MUX = 0x8, enumerator 157 (flags1 & RTCP) != 0)); in CreateChannels() 160 (flags2 & RTCP) != 0)); in CreateChannels() 524 CreateChannels(RTCP, RTCP); in TestSetContentsRtcpMux() 544 CreateChannels(RTCP, RTCP); in TestSetContentsRtcpMuxWithPrAnswer() 568 CreateContent(RTCP | RTCP_MUX | SECURE, in TestSetRemoteContentUpdate() 967 RTCP | RTCP_MUX, RTCP | RTCP_MUX, in TestCallTeardownRtcpMux() 1008 CreateChannels(0, RTCP); in SendNoRtcpToRtcp() 1023 CreateChannels(RTCP, 0); in SendRtcpToNoRtcp() 1038 CreateChannels(RTCP, RTCP); in SendRtcpToRtcp() [all …]
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/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
D | test_api.cc | 142 TEST_F(RtpRtcpAPITest, RTCP) { in TEST_F() argument 143 EXPECT_EQ(RtcpMode::kOff, module_->RTCP()); in TEST_F() 145 EXPECT_EQ(RtcpMode::kCompound, module_->RTCP()); in TEST_F()
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/external/webrtc/webrtc/call/ |
D | rtc_event_log.proto | 128 // required - Sender SSRC used for sending RTCP (such as receiver reports). 132 // RTCP mode is described by RFC 5506. 137 // required - RTCP mode to use. 228 // required - Sender SSRC used for sending RTCP (such as receiver reports).
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/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_unittest.cc | 212 ASSERT_FALSE(rtp_parser.RTCP()); in VerifyCVOPacket() 341 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 376 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 416 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 443 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 480 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 508 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 536 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F() 595 ASSERT_FALSE(rtp_parser.RTCP()); in TEST_F()
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D | rtp_utility.h | 52 bool RTCP() const;
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D | rtp_header_parser.cc | 46 return rtp_parser.RTCP(); in IsRtcp()
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D | rtp_utility.cc | 83 bool RtpHeaderParser::RTCP() const { in RTCP() function in webrtc::RtpUtility::RtpHeaderParser
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D | rtp_rtcp_impl.h | 131 RtcpMode RTCP() const override;
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D | rtp_rtcp_impl.cc | 483 RtcpMode ModuleRtpRtcpImpl::RTCP() const { in RTCP() function in webrtc::ModuleRtpRtcpImpl
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/external/webrtc/webrtc/test/ |
D | rtp_file_reader_unittest.cc | 88 if (!rtp_header_parser.RTCP() && in CountRtpPacketsPerSsrc()
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D | rtp_file_reader.cc | 457 if (rtp_parser.RTCP()) { in ReadPacket()
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/external/webrtc/tools/matlab/ |
D | rtpAnalyze.m | 20 %% Filter out RTCP packets. 23 fprintf('Removing %i RTCP packets\n', length(SeqNo) - sum(ix));
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/external/webrtc/webrtc/modules/rtp_rtcp/include/ |
D | rtp_rtcp.h | 331 virtual RtcpMode RTCP() const = 0;
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/external/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
D | mock_rtp_rtcp.h | 140 MOCK_CONST_METHOD0(RTCP, RtcpMode());
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/external/webrtc/webrtc/voice_engine/ |
D | channel.cc | 2961 RtcpMode method = _rtpRtcpModule->RTCP(); in GetRTCPStatus() 3108 RtcpMode status = _rtpRtcpModule->RTCP(); in SendApplicationDefinedRTCPPacket() 3139 if (_rtpRtcpModule->RTCP() == RtcpMode::kOff) { in GetRTPStatistics() 3212 &statistics, _rtpRtcpModule->RTCP() == RtcpMode::kOff)) { in GetRTPStatistics() 4069 RtcpMode method = _rtpRtcpModule->RTCP(); in GetRTT()
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/external/webrtc/webrtc/video/ |
D | vie_channel.cc | 514 if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff) in ProcessNACKRequest()
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/external/tcpdump/ |
D | CHANGES | 245 RTCP: Add some bounds checks
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