| /external/openscreen/cast/streaming/ |
| D | packet_util_unittest.cc | 33 EXPECT_EQ(ApparentPacketType::RTCP, result.first); in TEST()
64 EXPECT_EQ(ApparentPacketType::RTCP, result.first); in TEST()
70 EXPECT_EQ(ApparentPacketType::RTCP, result.first); in TEST()
85 EXPECT_EQ(ApparentPacketType::RTCP, result.first); in TEST()
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| D | receiver_packet_router.cc | 49 OSP_DCHECK(InspectPacketForRouting(packet).first == ApparentPacketType::RTCP); in SendRtcpPacket()
95 } else if (seems_like.first == ApparentPacketType::RTCP) { in OnReceivedPacket()
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| D | packet_util.cc | 34 ApparentPacketType::RTCP, in InspectPacketForRouting()
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| D | packet_util.h | 55 enum class ApparentPacketType { UNKNOWN, RTP, RTCP }; enumerator
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| D | sender_packet_router.cc | 103 if (seems_like.first != ApparentPacketType::RTCP) { in OnReceivedPacket()
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| /external/libsrtp2/ |
| D | CHANGES | 69 with RTCP an incorrect initialization vector is formed.
71 2.0) of libSRTP when using the AES 256 ICM cipher with OpenSSL for RTCP.
77 2.0) of libSRTP when using the AES GCM cipher for RTCP.
114 PR #304 - Fix (S)RTP and (S)RTCP for big endian machines
126 PR #167 - Additional RTCP and SRTCP tests
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| D | README.md | 106 libSRTP provides functions for protecting RTP and RTCP. RTP packets
111 functions apply security to RTCP packets.
117 needed to protect a particular RTP and RTCP stream. This datatype
123 stream context to protect the RTP and RTCP stream that it is
142 that describe the cryptograhic policies for RTP and RTCP, as well as
152 for RTP and RTCP protection, respectively.
162 ports for RTP and RTCP. RTCP, the RTP control protocol, is used to
173 RTP allows multiple sources to send RTP and RTCP traffic during the
253 * The replay window for (S)RTCP is hardcoded to 128 bits in length.
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| /external/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_utility.h | 37 bool RTCP() const;
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| D | rtp_rtcp_interface.h | 328 virtual RtcpMode RTCP() const = 0;
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| D | rtp_rtcp_impl2.h | 170 RtcpMode RTCP() const override;
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| D | rtp_rtcp_impl.h | 161 RtcpMode RTCP() const override;
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| D | rtp_utility.cc | 60 bool RtpHeaderParser::RTCP() const { in RTCP() function in webrtc::RtpUtility::RtpHeaderParser
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| /external/webrtc/docs/native-code/rtp-hdrext/abs-capture-time/ |
| D | README.md | 6 audio-to-video synchronization when RTCP-terminating intermediate systems (e.g.
60 as the clock used to generate NTP timestamps for RTCP sender reports on the
79 the RTCP sender reports on this stream. The sender system is typically either
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| /external/webrtc/logging/rtc_event_log/ |
| D | rtc_event_log.proto | 169 // required - Sender SSRC used for sending RTCP (such as receiver reports).
173 // RTCP mode is described by RFC 5506.
178 // required - RTCP mode to use.
262 // required - Sender SSRC used for sending RTCP (such as receiver reports).
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| D | rtc_event_log2.proto | 246 // TODO(terelius): Feasible to log parsed RTCP instead?
262 // TODO(terelius): Feasible to log parsed RTCP instead?
375 // required - Sender SSRC used for sending RTCP (such as receiver reports).
415 // required - Sender SSRC used for sending RTCP (such as receiver reports).
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| /external/webrtc/test/ |
| D | rtp_header_parser.cc | 49 return rtp_parser.RTCP(); in IsRtcp()
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| D | rtp_file_reader_unittest.cc | 89 if (!rtp_header_parser.RTCP() && in CountRtpPacketsPerSsrc()
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| D | rtp_file_reader.cc | 437 if (rtp_parser.RTCP()) { in ReadPacket()
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| /external/webrtc/test/fuzzers/corpora/ |
| D | README | 25 This corpus was initially assembled from the unittests. RTCP was
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| /external/webrtc/test/peer_scenario/tests/ |
| D | remote_estimate_test.cc | 32 if (!rtp_parser.RTCP()) { in GetRtpPacketExtensions()
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| /external/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 123 MOCK_METHOD(RtcpMode, RTCP, (), (const, override));
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| /external/webrtc/tools_webrtc/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/modules/pacing/ |
| D | packet_router.cc | 314 if (rtp_module->RTCP() == RtcpMode::kOff) { in SendCombinedRtcpPacket()
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| D | packet_router_unittest.cc | 322 ON_CALL(rtp_1, RTCP()).WillByDefault(Return(RtcpMode::kCompound)); in TEST_F()
323 ON_CALL(rtp_2, RTCP()).WillByDefault(Return(RtcpMode::kCompound)); in TEST_F()
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| /external/webrtc/docs/native-code/rtp-hdrext/playout-delay/ |
| D | README.md | 54 RTCP feedback to RTP sender includes the highest sequence number that was seen on the RTP receiver.…
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