| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_utility.cc | 292 uint32_t SSRC = *ptr++ << 24;
293 SSRC += *ptr++ << 16;
294 SSRC += *ptr++ << 8;
295 SSRC += *ptr++;
298 header->ssrc = SSRC;
332 uint32_t SSRC = *ptr++ << 24;
333 SSRC += *ptr++ << 16;
334 SSRC += *ptr++ << 8;
335 SSRC += *ptr++
[all...] |
| 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_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_sender.h | 93 void SetSSRC( const uint32_t ssrc);
95 void SetRemoteSSRC(uint32_t ssrc);
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,
302 // SSRC that we receive on our RTP channel
|
| rtcp_format_remb_unittest.cc | 121 uint32_t SSRC = 456789;
123 EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC));
|
| rtp_receiver_impl.h | 62 virtual uint32_t SSRC() const OVERRIDE;
|
| rtp_rtcp_impl.cc | 109 // Make sure that RTCP objects are aware of our SSRC.
110 uint32_t SSRC = rtp_sender_.SSRC();
111 rtcp_sender_.SetSSRC(SSRC);
112 SetRtcpReceiverSsrcs(SSRC);
249 uint32_t* ssrc,
251 rtp_sender_.RTXStatus(mode, ssrc, payload_type);
254 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
255 rtp_sender_.SetRtxSsrc(ssrc);
340 void ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
[all...] |
| rtcp_sender.cc | 255 const uint32_t* SSRC)
270 _rembSSRC[i] = SSRC[i];
327 RTCPSender::SetSSRC( const uint32_t ssrc)
338 _SSRC = ssrc;
341 void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
344 _remoteSSRC = ssrc;
371 int32_t RTCPSender::AddMixedCNAME(const uint32_t SSRC,
381 _csrcCNAMEs[SSRC] = ptr;
385 int32_t RTCPSender::RemoveMixedCNAME(const uint32_t SSRC) {
388 _csrcCNAMEs.find(SSRC);
[all...] |
| rtcp_packet.cc | 57 // Unused SSRC of media source, set to 0.
128 // | SSRC of sender |
160 // | SSRC of packet sender |
175 // | SSRC_1 (SSRC of first source) |
193 AssignUWord32(buffer, pos, (*it).SSRC);
236 // chunk | SSRC/CSRC_1 |
241 // chunk | SSRC/CSRC_2 |
263 AssignUWord32(buffer, pos, (*it).ssrc);
279 // | SSRC/CSRC |
306 // | SSRC/CSRC
[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
50 uint32_t ssrc,
53 uint32_t ssrc) OVERRIDE;
|
| vie_receiver.cc | 103 void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
104 rtp_payload_registry_->SetRtxSsrc(ssrc);
108 return rtp_receiver_->SSRC();
279 rtp_receive_statistics_->FecPacketReceived(header.ssrc);
303 &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(),
341 ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(), rtp_rtcp_);
400 rtp_receive_statistics_->GetStatistician(header.ssrc);
412 rtp_receive_statistics_->GetStatistician(header.ssrc);
417 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 | 91 virtual void SetRemoteSSRC(const uint32_t ssrc) = 0;
209 virtual void SetRtpStateForSsrc(uint32_t ssrc,
211 virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
214 * Get SSRC
216 virtual uint32_t SSRC() const = 0;
219 * configure SSRC, default is a random number
223 virtual void SetSSRC(const uint32_t ssrc) = 0;
264 // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
265 // only the SSRC is set.
266 virtual void SetRtxSsrc(uint32_t ssrc) = 0
[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.
59 // This method is called if SSRC of the incoming stream is changed.
61 const unsigned int SSRC) = 0;
107 // identifier (SSRC) explicitly.
109 const unsigned int SSRC,
113 // This function gets the SSRC for the outgoing RTP stream for the specified
116 unsigned int& SSRC) const = 0;
118 // This function map a incoming SSRC to a StreamType so that the engine
122 const unsigned int SSRC) const = 0
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 51 MOCK_METHOD1(SetRemoteSSRC, void(const uint32_t ssrc));
89 void(uint32_t ssrc, const RtpState& rtp_state));
90 MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
91 MOCK_CONST_METHOD0(SSRC,
94 void(const uint32_t ssrc));
104 void(int* modes, uint32_t* ssrc, int* payload_type));
133 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms,
159 int32_t(const uint32_t SSRC,
162 int32_t(const uint32_t SSRC));
182 int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock))
[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());
114 unsigned int ssrc = 0; local
121 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
123 EXPECT_EQ(1u, ssrc);
129 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
146 rtx_header.ssrc = kRtxSsrc;
149 rtx_header.ssrc = 0;
151 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/test/ |
| rtcp_packet_parser.h | 44 uint32_t Ssrc() const { return sr_.SenderSSRC; }
67 uint32_t Ssrc() const { return rr_.SenderSSRC; }
85 uint32_t Ssrc() const { return rb_.SSRC; }
149 uint32_t Ssrc() const { return cname_.SenderSSRC; }
168 uint32_t Ssrc() const { return bye_.SenderSSRC; }
186 uint32_t Ssrc() const { return rpsi_.SenderSSRC; }
246 uint32_t Ssrc() const { return pli_.SenderSSRC; }
265 uint32_t Ssrc() const { return sli_.SenderSSRC; }
304 uint32_t Ssrc() const { return fir_.SenderSSRC;
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/BWEStandAlone/ |
| TestSenderReceiver.h | 93 const uint32_t SSRC) OVERRIDE {}
|
| /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...] |
