| /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_utility.h | 52 bool RTCP() const;
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| 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_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 | 149 RTCP: Add some bounds checks
|