| /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/third_party/libjingle/source/talk/p2p/base/ |
| sessionmessages.h | 186 uint32 ssrc; member in struct:cricket::VideoViewRequest
191 VideoViewRequest(const std::string& nick_name, uint32 ssrc, uint32 width,
193 nick_name(nick_name), ssrc(ssrc), width(width), height(height),
|
| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.h | 35 int ssrc; member in struct:CastRtpPayloadParams
|
| cast_rtp_stream.cc | 69 config->sender_ssrc = payload_params.ssrc;
87 config->sender_ssrc = payload_params.ssrc;
215 ssrc(0),
|
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtputils.h | 43 uint32 ssrc; member in struct:cricket::RtpHeader
|
| streamparams.cc | 147 bool GetStreamBySsrc(const StreamParamsVec& streams, uint32 ssrc,
149 return GetStream(streams, StreamSelector(ssrc), stream_out);
174 bool RemoveStreamBySsrc(StreamParamsVec* streams, uint32 ssrc) {
175 return RemoveStream(streams, StreamSelector(ssrc));
|
| 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.
|
| rtpdump.h | 85 // Get the payload type, sequence number, timestampe, and SSRC of the RTP
90 bool GetRtpSsrc(uint32* ssrc) const;
115 // Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
116 void SetSsrc(uint32 ssrc);
|
| rtpdump.cc | 91 bool RtpDumpPacket::GetRtpSsrc(uint32* ssrc) const {
93 cricket::GetRtpSsrc(&data[0], data.size(), ssrc);
110 void RtpDumpReader::SetSsrc(uint32 ssrc) {
111 ssrc_override_ = ssrc;
151 // If the packet is RTP and we have specified a ssrc, replace the RTP ssrc
152 // with the specified ssrc.
|
| videoengine_unittest.h | 122 uint32 ssrc, cricket::VideoFrame* frame, bool* drop_frame) {
123 T::SignalMediaFrame(ssrc, frame, drop_frame);
588 int NumRtpBytes(uint32 ssrc) {
589 return network_interface_.NumRtpBytes(ssrc);
594 int NumRtpPackets(uint32 ssrc) {
595 return network_interface_.NumRtpPackets(ssrc);
615 int* seqnum, uint32* tstamp, uint32* ssrc,
640 // Read SSRC field.
642 if (ssrc) *ssrc = u32
939 uint32 ssrc = 0; local
960 uint32 ssrc = 0; local
1009 uint32 ssrc = 0; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| call.h | 78 void SetVideoRenderer(Session* session, uint32 ssrc,
95 const std::string& stream_name, uint32 ssrc,
98 const std::string& stream_name, uint32 ssrc);
192 void OnSpeakerMonitor(CurrentSpeakerMonitor* monitor, uint32 ssrc);
227 bool StopScreencastWithoutSendingUpdate(Session* session, uint32 ssrc);
|
| mediasession.h | 233 // Legacy streams have an ssrc, but nothing else.
234 void AddLegacyStream(uint32 ssrc) {
235 streams_.push_back(StreamParams::CreateLegacy(ssrc));
237 void AddLegacyStream(uint32 ssrc, uint32 fid_ssrc) {
238 StreamParams sp = StreamParams::CreateLegacy(ssrc);
239 sp.AddFidSsrc(ssrc, fid_ssrc);
475 uint32 ssrc, StreamParams* stream_out);
|
| typingmonitor.cc | 63 void TypingMonitor::OnVoiceChannelError(uint32 ssrc,
73 // multiple sending audio streams. SSRC 0 means the default sending audio
|
| /external/chromium_org/chrome/renderer/extensions/ |
| cast_streaming_native_handler.cc | 59 cast_params->ssrc = ext_params.ssrc ? *ext_params.ssrc : 0;
80 if (cast_params.ssrc)
81 ext_params->ssrc.reset(new int(cast_params.ssrc));
|
| /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/media/cast/net/rtp_sender/ |
| rtp_sender.cc | 30 config_.ssrc = audio_config->sender_ssrc;
38 config_.ssrc = video_config->sender_ssrc;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| peerconnection.cc | 692 uint32 ssrc) {
693 stream_handler_container_->AddRemoteAudioTrack(stream, audio_track, ssrc);
698 uint32 ssrc) {
699 stream_handler_container_->AddRemoteVideoTrack(stream, video_track, ssrc);
715 uint32 ssrc) {
716 stream_handler_container_->AddLocalAudioTrack(stream, audio_track, ssrc);
720 uint32 ssrc) {
721 stream_handler_container_->AddLocalVideoTrack(stream, video_track, ssrc);
|
| datachannel.cc | 194 send_params.ssrc = config_.id;
253 if (params.ssrc != expected_ssrc) {
435 send_params.ssrc = config_.id;
437 send_params.ssrc = send_ssrc_;
|
| 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...] |
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_receiver.h | 54 void SetRemoteSSRC(uint32 ssrc);
|
| 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/third_party/libjingle/source/talk/session/phone/ |
| mediasessionclient.h | 183 uint32 ssrc() const { return ssrc_; } function in class:cricket::MediaContentDescription
185 void set_ssrc(uint32 ssrc) {
186 ssrc_ = ssrc;
|
| /external/chromium_org/media/cast/audio_receiver/ |
| audio_decoder_unittest.cc | 63 rtp_header.webrtc.header.ssrc = 0x12345678;
113 rtp_header.webrtc.header.ssrc = 0x12345678;
182 rtp_header.webrtc.header.ssrc = 0x12345678;
|
| /external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
| callclient.cc | 265 // TODO: Use a random ssrc
781 params.ssrc = stream.first_ssrc();
867 if (data_streams && GetStreamBySsrc(*data_streams, params.ssrc, &stream)) {
869 "Received data from '%s' on stream '%s' (ssrc=%u): %s",
871 params.ssrc, text.c_str());
874 "Received data (ssrc=%u): %s",
875 params.ssrc, text.c_str());
1474 uint32 ssrc = stream.first_ssrc(); local
[all...] |
| /external/bluetooth/bluedroid/stack/avdt/ |
| avdt_scb_act.c | 67 ** Description This function generates a SSRC number unique to the stream.
69 ** Returns SSRC value.
243 UINT32 ssrc; local
255 BE_STREAM_TO_UINT32(ssrc, p);
330 UINT32 ssrc; local
344 BE_STREAM_TO_UINT32(ssrc, p);
373 AVDT_TRACE_WARNING5( " - SDES SSRC=0x%08x sc=%d %d len=%d %s",
374 ssrc, o_cc, *p, *(p+1), p+2);
414 UINT32 ssrc; local
567 BE_STREAM_TO_UINT32(ssrc, p_payload)
1204 UINT32 ssrc; local
1253 UINT32 ssrc; local
[all...] |
