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Searched refs:SSRC (Results 1 – 25 of 52) sorted by relevance

123

/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtcp_utility.cc632 _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24; in ParseReportBlockItem()
633 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16; in ParseReportBlockItem()
634 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8; in ParseReportBlockItem()
635 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++; in ParseReportBlockItem()
767 uint32_t SSRC = *_ptrRTCPData++ << 24; in ParseSDESChunk() local
768 SSRC += *_ptrRTCPData++ << 16; in ParseSDESChunk()
769 SSRC += *_ptrRTCPData++ << 8; in ParseSDESChunk()
770 SSRC += *_ptrRTCPData++; in ParseSDESChunk()
775 _packet.CName.SenderSSRC = SSRC; // Add SSRC in ParseSDESChunk()
1050 _packet.XRDLRRReportBlockItem.SSRC = *_ptrRTCPData++ << 24; in ParseXrDlrrItem()
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Drtp_rtcp_impl.cc104 uint32_t SSRC = rtp_sender_.SSRC(); in ModuleRtpRtcpImpl() local
105 rtcp_sender_.SetSSRC(SSRC); in ModuleRtpRtcpImpl()
106 SetRtcpReceiverSsrcs(SSRC); in ModuleRtpRtcpImpl()
290 if (rtp_sender_.SSRC() == ssrc) { in SetRtpStateForSsrc()
302 if (rtp_sender_.SSRC() == ssrc) { in GetRtpStateForSsrc()
313 uint32_t ModuleRtpRtcpImpl::SSRC() const { in SSRC() function in webrtc::ModuleRtpRtcpImpl
314 return rtp_sender_.SSRC(); in SSRC()
379 uint32_t SSRC = rtp_sender_.SSRC(); in SetSendingStatus() local
380 rtcp_sender_.SetSSRC(SSRC); in SetSendingStatus()
381 SetRtcpReceiverSsrcs(SSRC); in SetSendingStatus()
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Drtcp_utility.h72 uint32_t SSRC; member
105 uint32_t SSRC; member
111 uint32_t SSRC; member
150 uint32_t SSRC; member
161 uint32_t SSRC; // "Owner" member
172 uint32_t SSRC; member
Drtp_utility.cc173 uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr); in ParseRtcp() local
177 header->ssrc = SSRC; in ParseRtcp()
209 uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr); in Parse() local
226 header->ssrc = SSRC; in Parse()
Drtcp_format_remb_unittest.cc116 uint32_t SSRC = 456789; in TEST_F() local
118 rtcp_sender_->SetREMBData(1234, std::vector<uint32_t>(1, SSRC)); in TEST_F()
Drtcp_receiver.cc484 if (registered_ssrcs_.find(rtcpPacket.ReportBlockItem.SSRC) == in HandleReportBlock()
499 rtcpPacket.ReportBlockItem.SSRC); in HandleReportBlock()
509 reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC; in HandleReportBlock()
581 TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR_RTT", rb.SSRC, in HandleReportBlock()
929 if (registered_ssrcs_.find(packet.XRDLRRReportBlockItem.SSRC) == in HandleXrDlrrReportBlockItem()
969 if(rtcpPacket.XRVOIPMetricItem.SSRC == main_ssrc_) in HandleXRVOIPMetric()
1057 if (main_ssrc_ == rtcpPacket.TMMBRItem.SSRC && in HandleTMMBRItem()
1107 rtcpPacket.TMMBNItem.SSRC); in HandleTMMBNItem()
1211 if (main_ssrc_ != rtcpPacket.FIRItem.SSRC) { in HandleFIRItem()
Drtp_receiver_impl.h59 uint32_t SSRC() const override;
/external/webrtc/tools/matlab/
DrtpAnalyze.m18 [SeqNo,TimeStamp,ArrTime,Size,PT,M,SSRC] = importfile(input_file);
30 SSRC = SSRC(ix); variable
33 [uSSRC, ~, uix] = unique(SSRC);
60 SSRC = SSRC(ix); variable
69 fprintf('SSRC: %s\n', SSRC{1});
151 title(sprintf('SSRC: %s', SSRC{1}));
178 function [SeqNo,TimeStamp,SendTime,Size,PT,M,SSRC] = ...
181 % [SEQNO,TIMESTAMP,SENDTIME,SIZE,PT,M,SSRC] = IMPORTFILE(FILENAME) Reads
184 % [SEQNO,TIMESTAMP,SENDTIME,SIZE,PT,M,SSRC] = IMPORTFILE(FILENAME,
189 % [SeqNo,TimeStamp,SendTime,Size,PT,M,SSRC] =
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/external/webrtc/webrtc/modules/pacing/
Dpacket_router_unittest.cc48 EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1)); in TEST_F()
64 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); in TEST_F()
83 EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1)); in TEST_F()
85 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); in TEST_F()
96 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2)); in TEST_F()
109 EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1)); in TEST_F()
111 EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2)); in TEST_F()
Dpacket_router.cc48 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) { in TimeToSendPacket()
96 packet->WithPacketSenderSsrc(rtp_module->SSRC()); in SendFeedback()
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/
Drtp_rtcp_test.cc28 unsigned int SSRC);
44 unsigned int SSRC) { in OnIncomingSSRCChanged() argument
47 SSRC); in OnIncomingSSRCChanged()
52 if (incoming_ssrc_ == SSRC) in OnIncomingSSRCChanged()
/external/webrtc/webrtc/call/
Drtc_event_log.proto103 // required - The SSRC of the audio stream associated with the playout event.
121 // TODO(terelius): Video and audio streams could in principle share SSRC,
122 // so identifying a stream based only on SSRC might not work.
123 // It might be better to use a combination of SSRC and media type
124 // or SSRC and port number, but for now we will rely on SSRC only.
128 // required - Sender SSRC used for sending RTCP (such as receiver reports).
177 // required - SSRC to use for the RTX stream.
228 // required - Sender SSRC used for sending RTCP (such as receiver reports).
/external/srtp/doc/
Dintro.txt218 setting SSRC to 2078917053
233 19 octets received from SSRC 2078917053 word: A
234 19 octets received from SSRC 2078917053 word: a
235 20 octets received from SSRC 2078917053 word: aa
236 21 octets received from SSRC 2078917053 word: aal
256 (SSRC) identifier. Some participants may not send any SRTP traffic;
261 same session. The synchronization source identifier (SSRC) is used to
264 SSRC, sequence number, rollover counter, and other data. A particular
271 streams requires care. When key sharing is used, the SSRC values that
321 the SRTP master key and the SSRC value. The SSRC describes what to
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/external/srtp/
DREADME125 setting SSRC to 2078917053
135 19 octets received from SSRC 2078917053 word: A
136 19 octets received from SSRC 2078917053 word: a
137 20 octets received from SSRC 2078917053 word: aa
138 21 octets received from SSRC 2078917053 word: aal
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/
Dtmmbn.cc58 AssignUWord32(buffer, pos, tmmbr_item.SSRC); in CreateTmmbrItem()
97 tmmbn_item.SSRC = ssrc; in WithTmmbr()
Dtmmbr.h35 tmmbr_item_.SSRC = ssrc; in To()
/external/webrtc/webrtc/video/
Dvie_receiver.cc139 return rtp_receiver_->SSRC(); in GetRemoteSsrc()
370 restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(), in ParseAndHandleEncapsulatingHeader()
428 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL); in InsertRTCPPacket()
479 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); in IsPacketRetransmitted()
Dvie_channel.cc665 int32_t ViEChannel::SetSSRC(const uint32_t SSRC, in SetSSRC() argument
670 rtp_rtcp->SetRtxSsrc(SSRC); in SetSSRC()
672 rtp_rtcp->SetSSRC(SSRC); in SetSSRC()
678 const uint32_t SSRC) { in SetRemoteSSRCType() argument
679 vie_receiver_.SetRtxSsrc(SSRC); in SetRemoteSSRCType()
685 *ssrc = rtp_rtcp_modules_[idx]->SSRC(); in GetLocalSSRC()
862 counter.Add(counter_map[rtp_rtcp->SSRC()]); in GetSendRtcpPacketTypeCounter()
/external/webrtc/webrtc/test/
Drtcp_packet_parser.h85 uint32_t Ssrc() const { return rb_.SSRC; } in Ssrc()
322 uint32_t Ssrc() const { return fir_item_.SSRC; } in Ssrc()
449 uint32_t Ssrc() const { return tmmbr_item_.SSRC; } in Ssrc()
490 return tmmbns_[num].SSRC; in Ssrc()
568 return dlrrs_[num].SSRC; in Ssrc()
596 uint32_t Ssrc() const { return voip_metric_.SSRC; } in Ssrc()
/external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/
Dtest_api_rtcp.cc227 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName)); in TEST_F()
230 EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); in TEST_F()
242 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); in TEST_F()
Dtest_api.cc137 TEST_F(RtpRtcpAPITest, SSRC) { in TEST_F() argument
139 EXPECT_EQ(test_ssrc_, module_->SSRC()); in TEST_F()
/external/webrtc/webrtc/modules/rtp_rtcp/include/
Drtp_rtcp.h212 virtual uint32_t SSRC() const = 0;
372 virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
379 virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
Drtp_receiver.h93 virtual uint32_t SSRC() const = 0;
/external/webrtc/webrtc/modules/rtp_rtcp/mocks/
Dmock_rtp_rtcp.h92 MOCK_CONST_METHOD0(SSRC,
154 int32_t(const uint32_t SSRC,
157 int32_t(const uint32_t SSRC));
/external/webrtc/webrtc/modules/audio_coding/neteq/test/
DNETEQTEST_RTPpacket.cc395 uint32_t NETEQTEST_RTPpacket::SSRC() const in SSRC() function in NETEQTEST_RTPpacket
821 red.header.ssrc = SSRC(); in extractRED()
835 red.header.ssrc = SSRC(); in extractRED()

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