| /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/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/chromium_org/third_party/libjingle/source/talk/session/media/ |
| 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;
|
| /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_) {
|
| filemediaengine.h | 137 virtual bool AddStream(uint32 ssrc) { return true; }
138 virtual bool RemoveStream(uint32 ssrc) { return true; }
172 virtual bool AddStream(uint32 ssrc, uint32 voice_ssrc) { return true; }
173 virtual bool RemoveStream(uint32 ssrc) { return true; }
174 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| 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,
|
| webrtcsdp_unittest.cc | 161 "a=ssrc:1 cname:stream_1_cname\r\n"
162 "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
163 "a=ssrc:1 mslabel:local_stream_1\r\n"
164 "a=ssrc:1 label:audio_track_id_1\r\n"
165 "a=ssrc:4 cname:stream_2_cname\r\n"
166 "a=ssrc:4 msid:local_stream_2 audio_track_id_2\r\n"
167 "a=ssrc:4 mslabel:local_stream_2\r\n"
168 "a=ssrc:4 label:audio_track_id_2\r\n"
190 "a=ssrc:2 cname:stream_1_cname\r\n"
191 "a=ssrc:2 msid:local_stream_1 video_track_id_1\r\n
[all...] |
| statscollector_unittest.cc | 235 // Adds a track with a given SSRC into the stats.
368 // Set up an SSRC just to test that we get both kinds of stats back: SSRC and
463 // when StatsCollector::UpdateStats is called with ssrc stats.
476 // Constructs an ssrc stats update.
495 // and one ssrc report.
503 // and one ssrc report.
518 // This test verifies that an SSRC object has the identifier of a Transport
534 // Constructs an ssrc stats update.
568 // an outgoing SSRC where remote stats are not returned
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| fakertp.h | 34 // PT=0, SN=1, TS=0, SSRC=1
61 // PT=RR, LN=1, SSRC=1
62 // send SSRC=2, all other fields 0
70 // PT = 97, TS = 0, Seq = 1, SSRC = 2
97 // PT= 101, SN=2, TS=3, SSRC = 4
|
| 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);
|
| 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...] |
| 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));
|
| /external/srtp/ |
| README | 125 setting SSRC to 2078917053
135 19 octets received from SSRC 2078917053 word: A
136 19 octets received from SSRC 2078917053 word: a
137 20 octets received from SSRC 2078917053 word: aa
138 21 octets received from SSRC 2078917053 word: aal
|
| /external/srtp/doc/ |
| intro.txt | 218 setting SSRC to 2078917053
233 19 octets received from SSRC 2078917053 word: A
234 19 octets received from SSRC 2078917053 word: a
235 20 octets received from SSRC 2078917053 word: aa
236 21 octets received from SSRC 2078917053 word: aal
256 (SSRC) identifier. Some participants may not send any SRTP traffic;
261 same session. The synchronization source identifier (SSRC) is used to
264 SSRC, sequence number, rollover counter, and other data. A particular
271 streams requires care. When key sharing is used, the SSRC values that
321 the SRTP master key and the SSRC value. The SSRC describes what t
[all...] |
| /external/chromium_org/media/cast/rtcp/ |
| rtcp_sender.cc | 317 big_endian_writer.WriteU32(ssrc_); // Add our own SSRC.
356 big_endian_writer.WriteU32(ssrc_); // Add our own SSRC.
357 big_endian_writer.WriteU32(remote_ssrc); // Add the remote SSRC.
440 big_endian_writer.WriteU32(ssrc_); // Add our own SSRC.
441 big_endian_writer.WriteU32(0); // Remote SSRC must be 0.
481 big_endian_writer.WriteU32(ssrc_); // Add our own SSRC.
482 big_endian_writer.WriteU32(nack->remote_ssrc); // Add the remote SSRC.
531 big_endian_writer.WriteU32(ssrc_); // Add our own SSRC.
540 | SSRC |
544 | SSRC_1 (SSRC of first receiver) | sub
[all...] |
| rtcp_defines.h | 77 uint32 remote_ssrc; // SSRC of sender of this report.
78 uint32 media_ssrc; // SSRC of the RTP packet sender.
|
| rtcp_receiver.h | 54 void SetRemoteSSRC(uint32 ssrc);
|
| /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));
|
| /frameworks/av/libvideoeditor/vss/inc/ |
| M4VSS3GPP_InternalTypes.h | 51 #include "SSRC.h" /**< SSRC */
[all...] |
| /external/chromium_org/media/cast/framer/ |
| framer.cc | 16 uint32 ssrc,
21 &frame_id_map_, ssrc, decoder_faster_than_max_frame_rate,
|
| framer.h | 30 uint32 ssrc,
|
| /external/srtp/include/ |
| rtp.h | 77 struct sockaddr_in addr, unsigned int ssrc);
81 struct sockaddr_in addr, unsigned int ssrc);
|
| /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/content/browser/resources/media/ |
| stats_graph_helper.js | 8 // e.g. 1234-0-ssrc-abcd123-bytesSent is the graph for the series of bytesSent
9 // for ssrc-abcd123 of PeerConnection 0 in process 1234.
85 'ssrc': true,
210 if (report.type == 'ssrc') {
|
| /external/chromium_org/media/cast/test/ |
| receiver.cc | 77 test::InputBuilder input_tx("Choose audio sender SSRC.",
81 test::InputBuilder input_rx("Choose audio receiver SSRC.",
87 test::InputBuilder input_tx("Choose video sender SSRC.",
91 test::InputBuilder input_rx("Choose video receiver SSRC.",
|
