| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtcp_utility.cc | 588 _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24;
589 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16;
590 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8;
591 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++;
723 uint32_t SSRC = *_ptrRTCPData++ << 24;
724 SSRC += *_ptrRTCPData++ << 16;
725 SSRC += *_ptrRTCPData++ << 8;
726 SSRC += *_ptrRTCPData++;
731 _packet.CName.SenderSSRC = SSRC; // Add SSRC
[all...] |
| rtcp_sender.h | 97 void SetSSRC( const uint32_t ssrc);
99 void SetRemoteSSRC(uint32_t ssrc);
106 int32_t AddMixedCNAME(const uint32_t SSRC,
109 int32_t RemoveMixedCNAME(const uint32_t SSRC);
128 uint32_t SSRC,
131 int32_t RemoveExternalReportBlock(uint32_t SSRC);
142 const uint32_t* SSRC);
202 uint32_t SSRC,
296 uint32_t _remoteSSRC; // SSRC that we receive on our RTP channel
|
| rtcp_utility.h | 49 uint32_t SSRC;
88 uint32_t SSRC;
95 uint32_t SSRC;
138 uint32_t SSRC;
151 uint32_t SSRC; // "Owner"
164 uint32_t SSRC;
|
| rtcp_format_remb_unittest.cc | 122 uint32_t SSRC = 456789;
124 EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC));
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| rtp_utility.cc | 319 uint32_t SSRC = *ptr++ << 24;
320 SSRC += *ptr++ << 16;
321 SSRC += *ptr++ << 8;
322 SSRC += *ptr++;
325 header->ssrc = SSRC;
359 uint32_t SSRC = *ptr++ << 24;
360 SSRC += *ptr++ << 16;
361 SSRC += *ptr++ << 8;
362 SSRC += *ptr++
[all...] |
| rtp_rtcp_impl.cc | 104 // Make sure that RTCP objects are aware of our SSRC.
105 uint32_t SSRC = rtp_sender_.SSRC();
106 rtcp_sender_.SetSSRC(SSRC);
107 SetRtcpReceiverSsrcs(SSRC);
245 uint32_t* ssrc,
247 rtp_sender_.RTXStatus(mode, ssrc, payload_type);
250 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
251 rtp_sender_.SetRtxSsrc(ssrc);
336 uint32_t ModuleRtpRtcpImpl::SSRC() const
[all...] |
| rtp_receiver_impl.h | 64 uint32_t SSRC() const;
71 void SetRTXStatus(bool enable, uint32_t ssrc);
73 void RTXStatus(bool* enable, uint32_t* ssrc, int* payload_type) const;
|
| rtcp_sender.cc | 329 const uint32_t* SSRC)
344 _rembSSRC[i] = SSRC[i];
400 RTCPSender::SetSSRC( const uint32_t ssrc)
411 _SSRC = ssrc;
414 void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
417 _remoteSSRC = ssrc;
452 int32_t RTCPSender::AddMixedCNAME(const uint32_t SSRC,
462 _csrcCNAMEs[SSRC] = ptr;
466 int32_t RTCPSender::RemoveMixedCNAME(const uint32_t SSRC) {
469 _csrcCNAMEs.find(SSRC);
[all...] |
| rtcp_receiver.cc | 128 RTCPReceiver::SetRemoteSSRC( const uint32_t ssrc)
132 // new SSRC reset old reports
137 _remoteSSRC = ssrc;
178 LOG(LS_WARNING) << "Failed to reset rtt for ssrc " << remoteSSRC;
430 "ssrc", main_ssrc_);
461 "ssrc", main_ssrc_);
489 // |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to
493 if (registered_ssrcs_.find(rtcpPacket.ReportBlockItem.SSRC) ==
517 reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/ |
| 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
|
| vie_receiver.cc | 103 void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
104 rtp_payload_registry_->SetRtxSsrc(ssrc);
108 return rtp_receiver_->SSRC();
278 rtp_receive_statistics_->FecPacketReceived(header.ssrc);
302 &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
340 ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(), rtp_rtcp_);
399 rtp_receive_statistics_->GetStatistician(header.ssrc);
411 rtp_receive_statistics_->GetStatistician(header.ssrc);
416 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
|
| vie_rtp_rtcp_impl.cc | 111 const unsigned int SSRC,
114 LOG_F(LS_INFO) << "channel: " << video_channel << " ssrc: " << SSRC << "";
121 if (vie_channel->SetSSRC(SSRC, usage, simulcast_idx) != 0) {
130 const unsigned int SSRC) const {
132 << " usage: " << static_cast<int>(usage) << " ssrc: " << SSRC;
141 if (ptrViEChannel->SetRemoteSSRCType(usage, SSRC) != 0) {
149 unsigned int& SSRC) const {
157 if (vie_channel->GetLocalSSRC(idx, &SSRC) != 0)
[all...] |
| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
| rtp_rtcp_test.cc | 28 unsigned int SSRC);
34 void SetIncomingSsrc(unsigned int ssrc) {
36 incoming_ssrc_ = ssrc;
45 unsigned int SSRC) {
47 sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
48 SSRC);
53 if (incoming_ssrc_ == SSRC)
93 // We'll set up the RTCP CNAME and SSRC to something arbitrary here.
148 unsigned int ssrc; local
149 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_rtcp.h | 88 virtual void SetRemoteSSRC(const uint32_t ssrc) = 0;
207 * Get SSRC
209 virtual uint32_t SSRC() const = 0;
212 * configure SSRC, default is a random number
216 virtual void SetSSRC(const uint32_t ssrc) = 0;
257 // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
258 // only the SSRC is set.
259 virtual void SetRtxSsrc(uint32_t ssrc) = 0;
266 * Get status of sending RTX (RFC 4588) on a specific SSRC.
268 virtual void RTXSendStatus(int* modes, uint32_t* ssrc,
[all...] |
| rtp_receiver.h | 71 // state. This for instance means that any changes in SSRC and payload type is
92 // Returns the remote SSRC of the currently received RTP stream.
93 virtual uint32_t SSRC() const = 0;
|
| /external/chromium_org/third_party/webrtc/video_engine/include/ |
| vie_rtp_rtcp.h | 12 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
13 // - SSRC handling.
58 // This method is called if SSRC of the incoming stream is changed.
60 const unsigned int SSRC) = 0;
106 // identifier (SSRC) explicitly.
108 const unsigned int SSRC,
112 // This function gets the SSRC for the outgoing RTP stream for the specified
115 unsigned int& SSRC) const = 0;
117 // This function map a incoming SSRC to a StreamType so that the engine
121 const unsigned int SSRC) const = 0
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 40 MOCK_METHOD1(SetRemoteSSRC, void(const uint32_t ssrc));
77 MOCK_CONST_METHOD0(SSRC,
80 void(const uint32_t ssrc));
90 void(int* modes, uint32_t* ssrc, int* payload_type));
119 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms,
147 int32_t(const uint32_t SSRC,
150 int32_t(const uint32_t SSRC));
170 int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock));
172 int32_t(const uint32_t SSRC));
188 int32_t(const uint32_t bitrate, const uint8_t numberOfSSRC, const uint32_t* SSRC));
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 82 TEST_F(RtpRtcpAPITest, SSRC) {
84 EXPECT_EQ(test_ssrc, module->SSRC());
118 unsigned int ssrc = 0; local
125 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
127 EXPECT_EQ(1u, ssrc);
133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
137 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
150 rtx_header.ssrc = kRtxSsrc;
153 rtx_header.ssrc = 0;
155 rtx_header.ssrc = kRtxSsrc
[all...] |
| test_api_rtcp.cc | 58 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {
60 virtual void OnReceivedSLI(uint32_t ssrc,
64 virtual void OnReceivedRPSI(uint32_t ssrc,
79 const uint32_t ssrc) {
80 rtp_rtcp_->SetRemoteSSRC(ssrc);
253 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName));
256 EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
268 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
383 // |test_ssrc+1| is the SSRC of module2 that send the report.
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| rtp_to_text.cc | 83 DataLog::AddColumn(table_name, "ssrc", 1);
109 DataLog::InsertCell(table_name, "ssrc", packet->SSRC());
|
| NETEQTEST_RTPpacket.h | 54 uint32_t SSRC() const;
60 int setSSRC(uint32_t ssrc);
94 uint32_t ssrc, uint8_t markerBit) const;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| TestSenderReceiver.h | 90 const uint32_t SSRC) {};
|
| /external/chromium_org/third_party/webrtc/test/ |
| rtcp_packet_parser.cc | 35 ++report_blocks_per_ssrc_[parser.Packet().ReportBlockItem.SSRC];
|
| /external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/ |
| tb_external_transport.h | 102 void SetSSRCFilter(uint32_t SSRC);
|
| /external/chromium_org/third_party/webrtc/voice_engine/include/ |
| voe_rtp_rtcp.h | 13 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
14 // - SSRC handling.
58 int channel, unsigned int SSRC) = 0;
105 uint32_t sender_SSRC; // SSRC of sender
131 // Sets the local RTP synchronization source identifier (SSRC) explicitly.
132 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
134 // Gets the local RTP SSRC of a specified |channel|.
135 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
137 // Gets the SSRC of the incoming RTP packets.
138 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0
[all...] |
