| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| mediachannel.h | 93 // Sets the SSRC to be used for outgoing data.
115 uint32 ssrc; member in struct:cricket::VoiceSenderInfo
127 uint32 ssrc; member in struct:cricket::VoiceReceiverInfo
138 uint32 ssrc; member in struct:cricket::VideoSenderInfo
154 uint32 ssrc; member in struct:cricket::VideoReceiverInfo
218 // Adds a new receive-only stream with the specified SSRC.
219 virtual bool AddStream(uint32 ssrc) = 0;
221 virtual bool RemoveStream(uint32 ssrc) = 0;
235 virtual void GetLastMediaError(uint32* ssrc,
242 // ssrc(uint32), and error(VoiceMediaChannel::Error)
[all...] |
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_receiver.cc | 74 void RtcpReceiver::SetRemoteSSRC(uint32 ssrc) {
75 remote_ssrc_ = ssrc;
154 VLOG(1) << "Cast RTCP received SR from SSRC " << remote_ssrc;
187 VLOG(1) << "Cast RTCP received RR from SSRC " << remote_ssrc;
204 // |rtcp_field.ReportBlockItem.ssrc| is the ssrc identifier of the source to
210 if (rb.ssrc != ssrc_) {
214 VLOG(1) << "Cast RTCP received RB from SSRC " << remote_ssrc;
222 report_block.media_ssrc = rb.ssrc;
232 rtt_feedback_->OnReceivedDelaySinceLastReport(rb.ssrc,
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
| sctpdataengine_unittest.cc | 223 bool SendData(cricket::SctpDataMediaChannel* chan, uint32 ssrc,
227 params.ssrc = ssrc;
231 bool ReceivedData(const SctpFakeDataReceiver* recv, uint32 ssrc,
234 recv->last_params().ssrc == ssrc &&
307 << "recv2.last_params.ssrc="
308 << receiver2()->last_params().ssrc
310 << receiver2()->last_params().ssrc
320 << "recv1.last_params.ssrc=
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideoengine.cc | 177 FlushBlackFrameData(uint32 s, int64 t) : ssrc(s), timestamp(t) {
179 uint32 ssrc; member in struct:cricket::FlushBlackFrameData
2387 unsigned int ssrc; local
2493 uint32 ssrc = 0; local
2519 uint32 ssrc = 0; local
3376 const uint32 ssrc = send_channel->stream_params()->first_ssrc(); local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| 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...] |
| 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);
82 virtual bool RemoveRecvStream(uint32 ssrc);
83 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
109 void OnMediaError(uint32 ssrc, Error error);
|
| streamparams_unittest.cc | 80 const uint32 ssrc = 7; local
81 cricket::StreamParams one_sp = cricket::StreamParams::CreateLegacy(ssrc);
83 EXPECT_EQ(ssrc, one_sp.first_ssrc());
85 EXPECT_TRUE(one_sp.has_ssrc(ssrc));
86 EXPECT_FALSE(one_sp.has_ssrc(ssrc+1));
225 // stream1 has extra non-sim, non-fid ssrc.
|
| mediaengine.h | 139 virtual bool RegisterVoiceProcessor(uint32 ssrc,
142 virtual bool UnregisterVoiceProcessor(uint32 ssrc,
256 virtual bool RegisterVoiceProcessor(uint32 ssrc,
259 return voice_.RegisterProcessor(ssrc, processor, direction);
261 virtual bool UnregisterVoiceProcessor(uint32 ssrc,
264 return voice_.UnregisterProcessor(ssrc, processor, direction);
312 bool RegisterProcessor(uint32 ssrc,
315 bool UnregisterProcessor(uint32 ssrc,
|
| rtputils.cc | 159 GetRtpSsrc(data, len, &(header->ssrc)));
169 // This method returns SSRC first of RTCP packet, except if packet is SDES.
173 // Packet should be at least of 8 bytes, to get SSRC from a RTCP packet.
223 SetRtpSsrc(data, len, header.ssrc));
|
| /external/srtp/include/ |
| srtp.h | 138 * @brief ssrc_type_t describes the type of an SSRC.
140 * An ssrc_type_t enumeration is used to indicate a type of SSRC. See
145 ssrc_undefined = 0, /**< Indicates an undefined SSRC type. */
146 ssrc_specific = 1, /**< Indicates a specific SSRC value */
147 ssrc_any_inbound = 2, /**< Indicates any inbound SSRC value
150 ssrc_any_outbound = 3 /**< Indicates any outbound SSRC value
156 * @brief An ssrc_t represents a particular SSRC value, or a `wildcard' SSRC.
158 * An ssrc_t represents a particular SSRC value (if its type is
159 * ssrc_specific), or a wildcard SSRC value that will match al
211 ssrc_t ssrc; \/**< The SSRC value of stream, or the member in struct:srtp_policy_t
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| webrtcsession.cc | 210 const std::string& track_id, uint32 *ssrc) {
226 *ssrc = stream.first_ssrc();
231 uint32 ssrc, std::string* track_id) {
244 if (cricket::GetStreamBySsrc(audio_content->streams(), ssrc, &stream_out)) {
258 if (cricket::GetStreamBySsrc(video_content->streams(), ssrc, &stream_out)) {
638 // Update state and SSRC of local MediaStreams and DataChannels based on the
794 bool WebRtcSession::GetTrackIdBySsrc(uint32 ssrc, std::string* id) {
795 if (GetLocalTrackId(ssrc, id)) {
796 if (GetRemoteTrackId(ssrc, id)) {
797 LOG(LS_WARNING) << "SSRC " << ssr
[all...] |
| webrtcsession.h | 163 // Get the id used as a media stream track's "id" field from ssrc.
164 virtual bool GetTrackIdBySsrc(uint32 ssrc, std::string* id);
167 virtual void SetAudioPlayout(uint32 ssrc, bool enable,
169 virtual void SetAudioSend(uint32 ssrc, bool enable,
174 virtual bool SetCaptureDevice(uint32 ssrc,
176 virtual void SetVideoPlayout(uint32 ssrc,
179 virtual void SetVideoSend(uint32 ssrc, bool enable,
281 bool GetLocalTrackId(uint32 ssrc, std::string* track_id);
282 bool GetRemoteTrackId(uint32 ssrc, std::string* track_id);
|
| datachannel_unittest.cc | 148 EXPECT_EQ(provider_.last_send_data_params().ssrc,
165 // Tests that messages are sent with the right ssrc.
171 EXPECT_EQ(1U, provider_.last_send_data_params().ssrc);
183 params.ssrc = 0;
197 params.ssrc = 1;
|
| mediastreamsignaling.cc | 302 OnLocalTrackSeen(info.stream_label, info.track_id, info.ssrc,
314 OnLocalTrackSeen(info.stream_label, info.track_id, info.ssrc,
478 const std::string& track_id, uint32* ssrc) const {
484 *ssrc = it->second.ssrc;
489 const std::string& track_id, uint32* ssrc) const {
495 *ssrc = it->second.ssrc;
553 // Find removed tracks. Ie tracks where the track id or ssrc don't match the
559 if (!cricket::GetStreamBySsrc(streams, info.ssrc, ¶ms) |
575 uint32 ssrc = it->first_ssrc(); local
761 uint32 ssrc = it->first_ssrc(); local
[all...] |
| statscollector.cc | 132 const char StatsReport::kStatsValueNameSsrc[] = "ssrc";
149 const char StatsReport::kStatsReportTypeSsrc[] = "ssrc";
374 uint32 ssrc = it->ssrc(); local
377 StatsReport* report = collector->PrepareLocalReport(ssrc, transport_id);
383 report = collector->PrepareRemoteReport(ssrc, transport_id);
472 uint32 ssrc,
474 std::string ssrc_id = talk_base::ToString<uint32>(ssrc);
480 if (!session()->GetTrackIdBySsrc(ssrc, &track_id)) {
481 LOG(LS_WARNING) << "The SSRC " << ssr
[all...] |
| datachannel.h | 75 // kOpen: The channel have a local SSRC set by a call to UpdateSendSsrc
76 // and a remote SSRC set by call to UpdateReceiveSsrc and the transport
79 // has been called with SSRC==0
81 // SSRC==0.
140 // Set the SSRC this channel should use to send data on the
144 // Set the SSRC this channel should use to receive data from the
|
| peerconnection.h | 118 uint32 ssrc) OVERRIDE;
121 uint32 ssrc) OVERRIDE;
130 uint32 ssrc) OVERRIDE;
133 uint32 ssrc) OVERRIDE;
|
| /external/srtp/srtp/ |
| srtp.c | 229 * the SSRC
234 uint32_t ssrc,
239 debug_print(mod_srtp, "cloning stream (SSRC: 0x%08x)", ssrc);
266 /* set ssrc to that provided */
267 str->ssrc = ssrc;
475 debug_print(mod_srtp, "initializing stream (SSRC: 0x%08x)",
476 p->ssrc.value);
493 /* set the SSRC value *
[all...] |
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
| rtp_packetizer_unittest.cc | 47 EXPECT_EQ(config_.ssrc, rtp_header.ssrc);
104 config_.ssrc = kSsrc;
|
| /frameworks/av/libvideoeditor/vss/common/inc/ |
| SSRC.h | 20 /* %name: SSRC.h % */
125 /* used as input and as output of the SSRC module. */
192 /* This function is used to initialize the SSRC module instance. */
233 /* This function sets headroom gain and the post gain of the SSRC */
237 /* SSRC. The default value is LVM_MODE_ON. LVM_MODE_OFF */
274 /* Process function for the SSRC module. */
|
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/test/ |
| rtp_packet_builder.cc | 67 void RtpPacketBuilder::SetSsrc(uint32 ssrc) {
68 ssrc_ = ssrc;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| testsdpstrings.h | 114 "a=ssrc:4227871655 cname:GeAAgb6XCPNLVMX5\r\n"
115 "a=ssrc:4227871655 msid:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4 a0\r\n"
116 "a=ssrc:4227871655 mslabel:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4\r\n"
117 "a=ssrc:4227871655 label:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4a0\r\n"
139 "a=ssrc:4227871655 cname:TsmD02HRfhkJBm4m\r\n"
140 "a=ssrc:4227871655 msid:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3 a0\r\n"
141 "a=ssrc:4227871655 mslabel:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3\r\n"
142 "a=ssrc:4227871655 label:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3a0\r\n"
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| srtpfilter.cc | 483 uint32 ssrc; local
484 if (GetRtpSsrc(p, in_len, &ssrc)) {
485 srtp_stat_->AddProtectRtpResult(ssrc, err);
530 uint32 ssrc; local
531 if (GetRtpSsrc(p, in_len, &ssrc)) {
532 srtp_stat_->AddUnprotectRtpResult(ssrc, err);
593 policy.ssrc.type = static_cast<ssrc_type_t>(type);
594 policy.ssrc.value = 0;
647 LOG(LS_INFO) << "SRTP event: SSRC collision";
731 void SrtpStat::AddProtectRtpResult(uint32 ssrc, int result)
[all...] |
| /external/srtp/test/ |
| 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...] |
| /frameworks/av/libvideoeditor/vss/mcs/inc/ |
| M4MCS_InternalTypes.h | 49 #include "SSRC.h" /**< SSRC */
473 * Sample Rate Convertor (SSRC) stuff */
474 SSRC_Instance_t SsrcInstance; /**< Context of the Ssrc */
475 SSRC_Scratch_t* SsrcScratch; /**< Working memory of the Ssrc */
476 short iSsrcNbSamplIn; /**< Number of sample the Ssrc needs as input */
477 short iSsrcNbSamplOut; /**< Number of sample the Ssrc outputs */
478 M4OSA_MemAddr8 pSsrcBufferIn; /**< Input of the SSRC */
479 M4OSA_MemAddr8 pSsrcBufferOut; /**< Output of the SSRC */
480 M4OSA_MemAddr8 pPosInSsrcBufferIn; /**< Position into the SSRC in buffer *
[all...] |
