| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtcp_utility.cc | 588 _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24; in ParseReportBlockItem()
589 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16; in ParseReportBlockItem()
590 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8; in ParseReportBlockItem()
591 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++; in ParseReportBlockItem()
723 uint32_t SSRC = *_ptrRTCPData++ << 24; in ParseSDESChunk() local
724 SSRC += *_ptrRTCPData++ << 16; in ParseSDESChunk()
725 SSRC += *_ptrRTCPData++ << 8; in ParseSDESChunk()
726 SSRC += *_ptrRTCPData++; in ParseSDESChunk()
731 _packet.CName.SenderSSRC = SSRC; // Add SSRC in ParseSDESChunk()
1006 _packet.XRDLRRReportBlockItem.SSRC = *_ptrRTCPData++ << 24; in ParseXrDlrrItem()
[all …]
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| D | rtp_utility.cc | 292 uint32_t SSRC = *ptr++ << 24; in ParseRtcp() local
293 SSRC += *ptr++ << 16; in ParseRtcp()
294 SSRC += *ptr++ << 8; in ParseRtcp()
295 SSRC += *ptr++; in ParseRtcp()
298 header->ssrc = SSRC; in ParseRtcp()
332 uint32_t SSRC = *ptr++ << 24; in Parse() local
333 SSRC += *ptr++ << 16; in Parse()
334 SSRC += *ptr++ << 8; in Parse()
335 SSRC += *ptr++; in Parse()
351 header.ssrc = SSRC; in Parse()
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| D | rtcp_utility.h | 49 uint32_t SSRC; member
88 uint32_t SSRC; member
95 uint32_t SSRC; member
138 uint32_t SSRC; member
151 uint32_t SSRC; // "Owner" member
164 uint32_t SSRC; member
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| D | rtcp_sender.h | 101 int32_t AddMixedCNAME(const uint32_t SSRC,
104 int32_t RemoveMixedCNAME(const uint32_t SSRC);
123 uint32_t SSRC,
126 int32_t RemoveExternalReportBlock(uint32_t SSRC);
137 const uint32_t* SSRC);
196 uint32_t SSRC,
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| D | rtp_rtcp_impl.cc | 110 uint32_t SSRC = rtp_sender_.SSRC(); in ModuleRtpRtcpImpl() local
111 rtcp_sender_.SetSSRC(SSRC); in ModuleRtpRtcpImpl()
112 SetRtcpReceiverSsrcs(SSRC); in ModuleRtpRtcpImpl()
342 if (rtp_sender_.SSRC() == ssrc) { in SetRtpStateForSsrc()
358 if (rtp_sender_.SSRC() == ssrc) { in GetRtpStateForSsrc()
376 uint32_t ModuleRtpRtcpImpl::SSRC() const { in SSRC() function in webrtc::ModuleRtpRtcpImpl
377 return rtp_sender_.SSRC(); in SSRC()
468 uint32_t SSRC = rtp_sender_.SSRC(); in SetSendingStatus() local
469 rtcp_sender_.SetSSRC(SSRC); in SetSendingStatus()
470 SetRtcpReceiverSsrcs(SSRC); in SetSendingStatus()
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| D | rtcp_sender.cc | 255 const uint32_t* SSRC) in SetREMBData() argument
270 _rembSSRC[i] = SSRC[i]; in SetREMBData()
371 int32_t RTCPSender::AddMixedCNAME(const uint32_t SSRC, in AddMixedCNAME() argument
381 _csrcCNAMEs[SSRC] = ptr; in AddMixedCNAME()
385 int32_t RTCPSender::RemoveMixedCNAME(const uint32_t SSRC) { in RemoveMixedCNAME() argument
388 _csrcCNAMEs.find(SSRC); in RemoveMixedCNAME()
540 uint32_t SSRC, in AddExternalReportBlock() argument
543 return AddReportBlock(SSRC, &external_report_blocks_, reportBlock); in AddExternalReportBlock()
547 uint32_t SSRC, in AddReportBlock() argument
557 report_blocks->find(SSRC); in AddReportBlock()
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| D | rtcp_format_remb_unittest.cc | 121 uint32_t SSRC = 456789; in TEST_F() local
123 EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC)); in TEST_F()
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| D | rtcp_packet.cc | 193 AssignUWord32(buffer, pos, (*it).SSRC); in CreateReportBlocks()
461 AssignUWord32(buffer, pos, fir_item.SSRC); in CreateFir()
474 AssignUWord32(buffer, pos, tmmbr_item.SSRC); in CreateTmmbrItem()
661 AssignUWord32(buffer, pos, (*it_block).SSRC); in CreateDlrr()
699 AssignUWord32(buffer, pos, (*it).SSRC); in CreateVoipMetric()
1008 tmmbn_item.SSRC = ssrc; in WithTmmbr()
1083 dlrr.SSRC = ssrc; in WithDlrrItem()
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| D | rtcp_receiver.cc | 493 if (registered_ssrcs_.find(rtcpPacket.ReportBlockItem.SSRC) == in HandleReportBlock()
517 reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC; in HandleReportBlock()
589 TRACE_COUNTER_ID1("webrtc_rtp", "RR_RTT", rb.SSRC, RTT); in HandleReportBlock()
941 if (registered_ssrcs_.find(packet.XRDLRRReportBlockItem.SSRC) == in HandleXrDlrrReportBlockItem()
984 if(rtcpPacket.XRVOIPMetricItem.SSRC == main_ssrc_) in HandleXRVOIPMetric()
1083 if (main_ssrc_ == rtcpPacket.TMMBRItem.SSRC && in HandleTMMBRItem()
1145 rtcpPacket.TMMBNItem.SSRC); in HandleTMMBNItem()
1266 if (main_ssrc_ != rtcpPacket.FIRItem.SSRC) { in HandleFIRItem()
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| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
| D | rtp_rtcp_test.cc | 28 unsigned int SSRC);
45 unsigned int SSRC) { in OnIncomingSSRCChanged() argument
48 SSRC); in OnIncomingSSRCChanged()
53 if (incoming_ssrc_ == SSRC) in OnIncomingSSRCChanged()
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| D | rtp_rtcp.h | 216 virtual uint32_t SSRC() const = 0;
408 const uint32_t SSRC,
416 virtual int32_t RemoveMixedCNAME(const uint32_t SSRC) = 0;
493 const uint32_t SSRC,
501 virtual int32_t RemoveRTCPReportBlock(const uint32_t SSRC) = 0;
544 const uint32_t* SSRC) = 0;
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| D | rtp_receiver.h | 93 virtual uint32_t SSRC() const = 0;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 91 MOCK_CONST_METHOD0(SSRC,
159 int32_t(const uint32_t SSRC,
162 int32_t(const uint32_t SSRC));
182 int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock));
184 int32_t(const uint32_t SSRC));
200 int32_t(const uint32_t bitrate, const uint8_t numberOfSSRC, const uint32_t* SSRC));
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| /external/chromium_org/third_party/webrtc/video_engine/include/ |
| D | vie_rtp_rtcp.h | 61 const unsigned int SSRC) = 0;
109 const unsigned int SSRC,
116 unsigned int& SSRC) const = 0;
122 const unsigned int SSRC) const = 0;
127 unsigned int& SSRC) const = 0;
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | vie_rtp_rtcp_impl.h | 30 const unsigned int SSRC,
34 unsigned int& SSRC) const; // NOLINT
37 const unsigned int SSRC) const;
39 unsigned int& SSRC) const; // NOLINT
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| D | vie_rtp_rtcp_impl.cc | 111 const unsigned int SSRC, in SetLocalSSRC() argument
114 LOG_F(LS_INFO) << "channel: " << video_channel << " ssrc: " << SSRC << ""; in SetLocalSSRC()
121 if (vie_channel->SetSSRC(SSRC, usage, simulcast_idx) != 0) { in SetLocalSSRC()
130 const unsigned int SSRC) const { in SetRemoteSSRCType()
132 << " usage: " << static_cast<int>(usage) << " ssrc: " << SSRC; in SetRemoteSSRCType()
141 if (ptrViEChannel->SetRemoteSSRCType(usage, SSRC) != 0) { in SetRemoteSSRCType()
149 unsigned int& SSRC) const { in GetLocalSSRC()
157 if (vie_channel->GetLocalSSRC(idx, &SSRC) != 0) { in GetLocalSSRC()
165 unsigned int& SSRC) const { in GetRemoteSSRC()
172 if (vie_channel->GetRemoteSSRC(&SSRC) != 0) { in GetRemoteSSRC()
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| D | vie_receiver.cc | 108 return rtp_receiver_->SSRC(); in GetRemoteSsrc()
303 &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(), in ParseAndHandleEncapsulatingHeader()
341 ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(), rtp_rtcp_); in InsertRTCPPacket()
417 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); in IsPacketRetransmitted()
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| /external/chromium_org/third_party/libsrtp/srtp/doc/ |
| D | intro.txt | 218 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/doc/ |
| D | intro.txt | 218 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
[all …]
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| /external/chromium_org/third_party/libsrtp/srtp/ |
| D | README | 125 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
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| /external/srtp/ |
| D | README | 125 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
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| /external/chromium_org/third_party/webrtc/test/ |
| D | rtcp_packet_parser.h | 85 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()
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| D | test_api.cc | 82 TEST_F(RtpRtcpAPITest, SSRC) { in TEST_F() argument
84 EXPECT_EQ(test_ssrc, module->SSRC()); in TEST_F()
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| D | test_api_rtcp.cc | 253 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName)); in TEST_F()
256 EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); in TEST_F()
268 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName)); in TEST_F()
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| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| D | NETEQTEST_RTPpacket.cc | 401 uint32_t NETEQTEST_RTPpacket::SSRC() const in SSRC() function in NETEQTEST_RTPpacket
845 red.header.ssrc = SSRC(); in extractRED()
859 red.header.ssrc = SSRC(); in extractRED()
|