| /external/autotest/server/cros/ |
| cfm_jmidata_v3_helper.py | 12 SSRC = u'ssrc'
81 jmi_type=SSRC, is_audio=True, key=BYTES_RECEIVED)
85 jmi_type=SSRC, is_audio=True, key=BYTES_SENT)
89 jmi_type=SSRC, is_audio=True, key=AUDIO_OUTPUT)
93 jmi_type=SSRC, is_audio=True, key=AUDIO_INPUT)
97 jmi_type=SSRC, is_audio=False, key=BYTES_SENT)
101 jmi_type=SSRC, is_audio=False, key=BYTES_RECEIVED)
105 jmi_type=SSRC, is_audio=False, key=FRAMERATE_RECEIVED)
109 jmi_type=SSRC, is_audio=False, key=FRAMERATE_SENT
[all...] |
| /external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
| rtp_rtcp_test.cc | 28 unsigned int SSRC);
33 void SetIncomingSsrc(unsigned int ssrc) {
35 incoming_ssrc_ = ssrc;
44 unsigned int SSRC) {
46 sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
47 SSRC);
52 if (incoming_ssrc_ == SSRC)
75 // We'll set up the RTCP CNAME and SSRC to something arbitrary here.
112 unsigned int ssrc; local
113 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
[all...] |
| /external/webrtc/tools/matlab/ |
| rtpAnalyze.m | 18 [SeqNo,TimeStamp,ArrTime,Size,PT,M,SSRC] = importfile(input_file);
30 SSRC = SSRC(ix);
33 [uSSRC, ~, uix] = unique(SSRC);
60 SSRC = SSRC(ix);
69 fprintf('SSRC: %s\n', SSRC{1});
151 title(sprintf('SSRC: %s', SSRC{1}))
[all...] |
| /external/webrtc/webrtc/modules/pacing/ |
| packet_router_unittest.cc | 46 // Send on the first module by letting rtp_1 be sending with correct ssrc.
48 EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
64 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
81 // Add a packet with incorrect ssrc and test it's dropped in the router.
83 EXPECT_CALL(rtp_1, SSRC()).Times(1).WillOnce(Return(kSsrc1));
85 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
93 // rtp_1 has been removed, try sending a packet on that ssrc and make sure
96 EXPECT_CALL(rtp_2, SSRC()).Times(1).WillOnce(Return(kSsrc2));
109 EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
111 EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2))
[all...] |
| packet_router.cc | 42 bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
48 if (rtp_module->SendingMedia() && ssrc == rtp_module->SSRC()) {
49 return rtp_module->TimeToSendPacket(ssrc, sequence_number,
96 packet->WithPacketSenderSsrc(rtp_module->SSRC());
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| tmmbr.h | 31 void From(uint32_t ssrc) {
32 tmmbr_.SenderSSRC = ssrc;
34 void To(uint32_t ssrc) {
35 tmmbr_item_.SSRC = ssrc;
|
| tmmbn.cc | 58 AssignUWord32(buffer, pos, tmmbr_item.SSRC);
73 // | SSRC |
90 bool Tmmbn::WithTmmbr(uint32_t ssrc, uint32_t bitrate_kbps, uint16_t overhead) {
97 tmmbn_item.SSRC = ssrc;
|
| tmmbr.cc | 60 AssignUWord32(buffer, pos, tmmbr_item.SSRC);
75 // | SSRC |
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_utility.cc | 632 _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24;
633 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16;
634 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8;
635 _packet.ReportBlockItem.SSRC += *_ptrRTCPData++;
767 uint32_t SSRC = *_ptrRTCPData++ << 24;
768 SSRC += *_ptrRTCPData++ << 16;
769 SSRC += *_ptrRTCPData++ << 8;
770 SSRC += *_ptrRTCPData++;
775 _packet.CName.SenderSSRC = SSRC; // Add SSRC
[all...] |
| rtcp_utility.h | 72 uint32_t SSRC;
105 uint32_t SSRC;
111 uint32_t SSRC;
150 uint32_t SSRC;
161 uint32_t SSRC; // "Owner"
172 uint32_t SSRC;
|
| rtcp_format_remb_unittest.cc | 116 uint32_t SSRC = 456789;
118 rtcp_sender_->SetREMBData(1234, std::vector<uint32_t>(1, SSRC));
|
| rtp_receiver_impl.h | 59 uint32_t SSRC() const override;
|
| rtp_utility.cc | 173 uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
177 header->ssrc = SSRC;
209 uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
226 header->ssrc = SSRC;
|
| rtp_rtcp_impl.cc | 103 // Make sure that RTCP objects are aware of our SSRC.
104 uint32_t SSRC = rtp_sender_.SSRC();
105 rtcp_sender_.SetSSRC(SSRC);
106 SetRtcpReceiverSsrcs(SSRC);
209 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
210 rtp_sender_.SetRtxSsrc(ssrc);
288 bool ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
290 if (rtp_sender_.SSRC() == ssrc) {
[all...] |
| /external/syslinux/core/ |
| Makefile | 45 SSRC := $(shell find $(SRC) -name '*.S' -print)
48 ALLSRC = $(NASMSRC) $(NASMHDR) $(CSRC) $(SSRC) $(CHDR) $(OTHERSRC)
51 SOBJ := $(subst $(SRC)/,,$(patsubst %.S,%.o,$(SSRC)))
|
| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| 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;
|
| rtp_rtcp.h | 94 virtual void SetRemoteSSRC(uint32_t ssrc) = 0;
204 // Returns true if the ssrc matched this module, false otherwise.
205 virtual bool SetRtpStateForSsrc(uint32_t ssrc,
207 virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
210 * Get SSRC
212 virtual uint32_t SSRC() const = 0;
215 * configure SSRC, default is a random number
217 virtual void SetSSRC(uint32_t ssrc) = 0;
238 // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX,
239 // only the SSRC is set
[all...] |
| /external/webrtc/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 | 53 uint32_t SSRC() const;
59 int setSSRC(uint32_t ssrc);
93 uint32_t ssrc, uint8_t markerBit) const;
|
| /external/webrtc/webrtc/voice_engine/include/ |
| voe_rtp_rtcp.h | 13 // - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
14 // - SSRC handling.
56 virtual void OnIncomingSSRCChanged(int channel, unsigned int SSRC) = 0;
89 uint32_t sender_SSRC; // SSRC of sender
113 // Sets the local RTP synchronization source identifier (SSRC) explicitly.
114 virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
116 // Gets the local RTP SSRC of a specified |channel|.
117 virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
119 // Gets the SSRC of the incoming RTP packets.
120 virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api_rtcp.cc | 38 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
39 virtual void OnReceivedSLI(uint32_t ssrc,
43 virtual void OnReceivedRPSI(uint32_t ssrc,
58 void OnIncomingSSRCChanged(const uint32_t ssrc) override {
59 rtp_rtcp_->SetRemoteSSRC(ssrc);
227 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC() + 1, cName));
230 EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
242 EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
260 // |test_ssrc+1| is the SSRC of module2 that send the report.
|
| /external/webrtc/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/webrtc/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 54 MOCK_METHOD1(SetRemoteSSRC, void(const uint32_t ssrc));
90 bool(uint32_t ssrc, const RtpState& rtp_state));
91 MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
92 MOCK_CONST_METHOD0(SSRC,
95 void(const uint32_t ssrc));
133 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms,
154 int32_t(const uint32_t SSRC,
157 int32_t(const uint32_t SSRC));
|
| /external/webrtc/webrtc/video/ |
| vie_receiver.cc | 126 void ViEReceiver::SetRtxSsrc(uint32_t ssrc) {
127 rtp_payload_registry_->SetRtxSsrc(ssrc);
130 bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const {
131 return rtp_payload_registry_->GetRtxSsrc(ssrc);
139 return rtp_receiver_->SSRC();
289 ss << "Packet received on SSRC: " << header.ssrc << " with payload type: "
370 restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
428 rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL);
462 rtp_receive_statistics_->GetStatistician(header.ssrc);
[all...] |
| vie_sync_module.cc | 82 new StreamSynchronization(video_rtp_rtcp_->SSRC(), voe_channel_id));
|
