| /external/chromium_org/third_party/libsrtp/srtp/include/ |
| rtp.h | 77 struct sockaddr_in addr, unsigned int ssrc);
81 struct sockaddr_in addr, unsigned int ssrc);
|
| /external/chromium_org/third_party/webrtc/ |
| config.h | 26 : ssrc(0),
30 uint32_t ssrc; member in struct:webrtc::RtpStatistics
|
| /external/chromium_org/third_party/webrtc/modules/pacing/include/ |
| paced_sender.h | 46 virtual bool TimeToSendPacket(uint32_t ssrc,
81 uint32_t ssrc,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| remote_ntp_time_estimator.cc | 29 bool RemoteNtpTimeEstimator::UpdateRtcpTimestamp(uint32_t ssrc,
33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL);
|
| rtp_rtcp_impl.h | 45 virtual void SetRemoteSSRC(const uint32_t ssrc);
76 virtual uint32_t SSRC() const OVERRIDE;
78 // Configure SSRC, default is a random number.
79 virtual void SetSSRC(const uint32_t ssrc) OVERRIDE;
96 virtual void RTXSendStatus(int* mode, uint32_t* ssrc,
99 virtual void SetRtxSsrc(uint32_t ssrc) OVERRIDE;
125 virtual bool TimeToSendPacket(uint32_t ssrc,
161 virtual int32_t AddMixedCNAME(const uint32_t ssrc,
164 virtual int32_t RemoveMixedCNAME(const uint32_t ssrc) OVERRIDE;
195 const uint32_t ssrc, const RTCPReportBlock* receive_block) OVERRIDE
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| peerconnection.h | 125 uint32 ssrc) OVERRIDE;
128 uint32 ssrc) OVERRIDE;
137 uint32 ssrc) OVERRIDE;
140 uint32 ssrc) OVERRIDE;
144 uint32 ssrc) OVERRIDE;
|
| /external/chromium_org/third_party/libsrtp/srtp/test/ |
| dtls_srtp_driver.c | 53 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc);
192 policy.ssrc.type = ssrc_any_inbound;
214 * srtp_create_test_packet(len, ssrc) returns a pointer to a
216 * by pkt_octet_len and the SSRC value ssrc. The total length of the
227 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc) {
247 hdr->ssrc = htonl(ssrc); /* synch. source */
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| /external/srtp/test/ |
| dtls_srtp_driver.c | 53 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc);
185 policy.ssrc.type = ssrc_any_inbound;
201 * srtp_create_test_packet(len, ssrc) returns a pointer to a
203 * by pkt_octet_len and the SSRC value ssrc. The total length of the
214 srtp_create_test_packet(int pkt_octet_len, uint32_t ssrc) {
234 hdr->ssrc = htonl(ssrc); /* synch. source */
|
| rtpw.c | 149 uint32_t ssrc = 0xdeadbeef; /* ssrc value hardcoded for now */ local
310 * using the right SSRC value
329 policy.ssrc.type = ssrc_specific;
330 policy.ssrc.value = ssrc;
373 policy.ssrc.type = ssrc_specific;
374 policy.ssrc.value = ssrc;
413 rtp_sender_init(snd, sock, name, ssrc);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| NETEQTEST_RTPpacket.cc | 292 rtp_header->header.ssrc = _rtpInfo.header.ssrc;
401 uint32_t NETEQTEST_RTPpacket::SSRC() const
414 return tempRTPinfo.header.ssrc;
496 int NETEQTEST_RTPpacket::setSSRC(uint32_t ssrc)
506 _rtpInfo.header.ssrc = ssrc;
509 _datagram[8]=(unsigned char)((ssrc>>24)&0xFF);
510 _datagram[9]=(unsigned char)((ssrc>>16)&0xFF);
511 _datagram[10]=(unsigned char)((ssrc>>8)&0xFF)
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
| sctpdataengine_unittest.cc | 271 void AddStream(int ssrc) {
272 cricket::StreamParams p(cricket::StreamParams::CreateLegacy(ssrc));
291 bool SendData(cricket::SctpDataMediaChannel* chan, uint32 ssrc,
295 params.ssrc = ssrc;
301 bool ReceivedData(const SctpFakeDataReceiver* recv, uint32 ssrc,
304 recv->last_params().ssrc == ssrc &&
367 << ", recv2.last_params.ssrc="
368 << receiver2()->last_params().ssrc
[all...] |
| sctpdataengine.h | 146 virtual bool RemoveSendStream(uint32 ssrc);
148 virtual bool RemoveRecvStream(uint32 ssrc);
213 bool ResetStream(uint32 ssrc);
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| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideoengine.h | 257 virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format);
262 virtual bool RemoveSendStream(uint32 ssrc);
264 virtual bool RemoveRecvStream(uint32 ssrc);
265 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
267 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
276 virtual bool MuteStream(uint32 ssrc, bool on);
294 bool GetRenderer(uint32 ssrc, VideoRenderer** renderer);
295 bool GetVideoAdapter(uint32 ssrc, CoordinatedVideoAdapter** video_adapter);
323 // contain the new channel's ID. If |receiving| is true |ssrc| is the
324 // remote ssrc. If |sending| is true the ssrc is local ssrc. If bot
[all...] |
| /external/chromium_org/third_party/webrtc/video_engine/test/libvietest/testbed/ |
| tb_external_transport.cc | 134 uint32_t ssrc = ptr[8] << 24; local
135 ssrc += ptr[9] << 16;
136 ssrc += ptr[10] << 8;
137 ssrc += ptr[11];
138 if (ssrc != _SSRC)
317 void TbExternalTransport::SetSSRCFilter(uint32_t ssrc)
321 _SSRC = ssrc;
415 unsigned int ssrc = 0; local
418 ssrc = ((packet->packetBuffer[8]) << 24);
419 ssrc += (packet->packetBuffer[9] << 16)
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| hybridvideoengine.h | 64 virtual bool RemoveSendStream(uint32 ssrc);
65 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer);
67 virtual bool MuteStream(uint32 ssrc, bool muted);
75 virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format);
83 virtual bool RemoveRecvStream(uint32 ssrc);
84 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
110 void OnMediaError(uint32 ssrc, Error error);
|
| rtpdump_unittest.cc | 50 uint32 ssrc; local
60 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc));
61 EXPECT_EQ(kTestSsrc, ssrc);
131 uint32 ssrc; local
132 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc));
133 EXPECT_EQ(kTestSsrc, ssrc);
138 // Rewind the stream and read again with a specified ssrc.
147 uint32 ssrc; local
148 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc));
149 EXPECT_EQ(send_ssrc, ssrc);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| RTPFile.cc | 36 rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
43 uint32_t ssrc) {
54 rtpHeader[8] = (unsigned char) ((ssrc >> 24) & 0xFF);
55 rtpHeader[9] = (unsigned char) ((ssrc >> 16) & 0xFF);
57 rtpHeader[10] = (unsigned char) ((ssrc >> 8) & 0xFF);
58 rtpHeader[11] = (unsigned char) (ssrc & 0xFF);
106 rtpInfo->header.ssrc = 0;
|
| /external/chromium_org/third_party/webrtc/modules/remote_bitrate_estimator/ |
| remote_bitrate_estimator_single_stream.cc | 32 // Called for each incoming packet. If this is a new SSRC, a new
49 // Removes all data for |ssrc|.
50 virtual void RemoveStream(unsigned int ssrc) OVERRIDE;
62 // Map from SSRC to over-use detector and last incoming packet time in
112 uint32_t ssrc = header.ssrc; local
117 SsrcOveruseDetectorMap::iterator it = overuse_detectors_.find(ssrc);
119 // This is a new SSRC. Adding to map.
120 // TODO(holmer): If the channel changes SSRC the old SSRC will still b
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| rtp_rtcp_defines.h | 176 uint32_t remoteSSRC; // SSRC of sender of this report.
177 uint32_t sourceSSRC; // SSRC of the RTP packet sender.
234 // Receiving payload change or SSRC change. (return success!)
247 const uint32_t ssrc) = 0;
253 virtual void ResetStatistics(uint32_t ssrc) = 0;
269 virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
271 virtual void OnReceivedSLI(uint32_t ssrc,
274 virtual void OnReceivedRPSI(uint32_t ssrc,
320 const uint32_t ssrc) OVERRIDE {}
326 virtual void ResetStatistics(uint32_t ssrc) OVERRIDE {
[all...] |
| /external/chromium_org/media/cast/transport/pacing/ |
| paced_sender.h | 30 // { capture_time, ssrc, packet_id }
46 virtual bool SendRtcpPacket(uint32 ssrc, PacketRef packet) = 0;
52 uint32 ssrc,
78 virtual bool SendRtcpPacket(uint32 ssrc, PacketRef packet) OVERRIDE;
|
| /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...] |
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/test/ |
| rtp_packet_builder.h | 28 void SetSsrc(uint32 ssrc);
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/ |
| rtp_packetizer.h | 37 // SSRC.
38 unsigned int ssrc; member in struct:media::cast::transport::RtpPacketizerConfig
|
| rtp_packetizer_unittest.cc | 52 EXPECT_EQ(config_.ssrc, rtp_header.ssrc);
106 config_.ssrc = kSsrc;
112 pacer_->RegisterVideoSsrc(config_.ssrc);
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.h | 33 uint32 ssrc; member in struct:media::cast::transport::RtpCastTestHeader
|
