| /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);
82 virtual bool RemoveRecvStream(uint32 ssrc);
83 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer);
109 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...] |
| rtpdataengine.h | 107 virtual bool RemoveSendStream(uint32 ssrc);
109 virtual bool RemoveRecvStream(uint32 ssrc);
|
| /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...] |
| /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...] |
| currentspeakermonitor.cc | 88 uint32 ssrc = stream_list_it->first; local
89 active_ssrc_to_level_map[ssrc] = stream_list_it->second;
93 if (ssrc_to_speaking_state_map_.find(ssrc) ==
95 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
|
| call.cc | 174 << it->selector.ssrc << ", '"
199 void Call::SetVideoRenderer(Session* session, uint32 ssrc,
203 video_channel->SetRenderer(ssrc, renderer);
204 LOG(LS_INFO) << "Set renderer of ssrc " << ssrc
207 LOG(LS_INFO) << "Failed to set renderer of ssrc " << ssrc << ".";
343 uint32 ssrc = it->second.started_screencasts.begin()->first; local
344 if (!StopScreencastWithoutSendingUpdate(it->second.session, ssrc)) {
345 LOG(LS_ERROR) << "Unable to stop screencast with ssrc " << ssrc
[all...] |
| /external/chromium/third_party/libjingle/source/talk/session/phone/ |
| filemediaengine.cc | 93 // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
182 uint32 ssrc; local
183 if (!packet->GetRtpSsrc(&ssrc)) {
188 first_ssrc_ = ssrc;
190 if (ssrc == first_ssrc_) {
|
| /external/chromium_org/media/cast/framer/ |
| framer.h | 30 uint32 ssrc,
|
| framer.cc | 16 uint32 ssrc,
21 &frame_id_map_, ssrc, decoder_faster_than_max_frame_rate,
|
| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.h | 31 uint32 ssrc; member in struct:media::cast::RtpCastTestHeader
|
| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/test/ |
| rtp_packet_builder.h | 28 void SetSsrc(uint32 ssrc);
|
| rtp_packet_builder.cc | 67 void RtpPacketBuilder::SetSsrc(uint32 ssrc) {
68 ssrc_ = ssrc;
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| 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;
|
| peerconnection.h | 118 uint32 ssrc) OVERRIDE;
121 uint32 ssrc) OVERRIDE;
130 uint32 ssrc) OVERRIDE;
133 uint32 ssrc) OVERRIDE;
|
| 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...] |
| mediastreamhandler_unittest.cc | 57 MOCK_METHOD3(SetAudioPlayout, void(uint32 ssrc, bool enable,
59 MOCK_METHOD4(SetAudioSend, void(uint32 ssrc, bool enable,
68 MOCK_METHOD2(SetCaptureDevice, bool(uint32 ssrc,
70 MOCK_METHOD3(SetVideoPlayout, void(uint32 ssrc,
73 MOCK_METHOD3(SetVideoSend, void(uint32 ssrc, bool enable,
|
| 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...] |
| mediastreamsignaling_unittest.cc | 78 "a=ssrc:1 cname:stream1\r\n"
79 "a=ssrc:1 mslabel:stream1\r\n"
80 "a=ssrc:1 label:audiotrack0\r\n"
84 "a=ssrc:2 cname:stream1\r\n"
85 "a=ssrc:2 mslabel:stream1\r\n"
86 "a=ssrc:2 label:videotrack0\r\n";
100 "a=ssrc:1 cname:stream1\r\n"
101 "a=ssrc:1 msid:stream1 audiotrack0\r\n"
102 "a=ssrc:3 cname:stream2\r\n"
103 "a=ssrc:3 msid:stream2 audiotrack1\r\n
377 uint32 ssrc; member in struct:MockSignalingObserver::TrackInfo
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/other/ |
| linphonemediaengine.h | 127 virtual bool AddStream(uint32 ssrc) { return true; }
128 virtual bool RemoveStream(uint32 ssrc) { return true; }
131 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
134 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
|
| /external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
| sctpdataengine.cc | 162 packet->params.ssrc = rcv.rcv_sid;
424 << "' with ssrc=" << stream.first_ssrc()
433 bool SctpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
435 if (!GetStreamBySsrc(streams_, ssrc, &found_stream)) {
439 RemoveStreamBySsrc(&streams_, ssrc);
443 // Note: expects exactly one ssrc. If none are given, it will fail. If more
454 << "' with ssrc=" << stream.first_ssrc()
462 << "' with ssrc=" << stream.first_ssrc();
466 bool SctpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
467 RemoveStreamBySsrc(&streams_, ssrc);
[all...] |
| sctpdataengine.h | 138 virtual bool RemoveSendStream(uint32 ssrc);
140 virtual bool RemoveRecvStream(uint32 ssrc);
|
| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcmediaengine.h | 149 uint32 ssrc, VoiceProcessor* video_processor,
151 return delegate_->RegisterVoiceProcessor(ssrc, video_processor, direction);
154 uint32 ssrc, VoiceProcessor* video_processor,
156 return delegate_->UnregisterVoiceProcessor(ssrc, video_processor,
|
| /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;
|
