| /external/chromium_org/media/cast/net/rtp_sender/rtp_packetizer/ |
| rtp_packetizer_config.h | 25 // SSRC.
26 unsigned int ssrc; member in struct:media::cast::RtpPacketizerConfig
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| /external/chromium_org/media/cast/rtp_receiver/rtp_parser/ |
| rtp_parser.h | 18 ssrc = 0;
23 uint32 ssrc; member in struct:media::cast::RtpParserConfig
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| rtp_parser.cc | 33 rtp_header->webrtc.header.ssrc == parser_config_.ssrc) {
37 // Not a valid payload type / ssrc combination.
52 uint32 rtp_timestamp, ssrc; local
56 big_endian_reader.ReadU32(&ssrc);
58 if (ssrc != parser_config_.ssrc) return false;
64 rtp_header->webrtc.header.ssrc = ssrc;
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| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| ssrcmuxfilter.cc | 50 uint32 ssrc = 0; local
52 GetRtpSsrc(data, len, &ssrc);
61 if (!GetRtcpSsrc(data, len, &ssrc)) return false;
62 if (ssrc == kSsrc01) {
63 // SSRC 1 has a special meaning and indicates generic feedback on
70 return FindStream(ssrc);
82 bool SsrcMuxFilter::RemoveStream(uint32 ssrc) {
83 return RemoveStreamBySsrc(&streams_, ssrc);
86 bool SsrcMuxFilter::FindStream(uint32 ssrc) const {
87 if (ssrc == 0)
[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;
|
| mediamessages.cc | 51 bool ParseSsrc(const std::string& string, uint32* ssrc) {
52 return talk_base::FromString(string, ssrc);
55 bool ParseSsrc(const buzz::XmlElement* element, uint32* ssrc) {
59 return ParseSsrc(element->BodyText(), ssrc);
86 AddXmlAttr(view_elem, QN_SSRC, view.selector.ssrc);
155 uint32 ssrc; local
156 if (!ParseSsrc(view_elem->Attr(QN_SSRC), &ssrc)) {
157 return BadParse("Invalid or missing view ssrc.", error);
159 view->selector = StreamSelector(ssrc);
220 uint32 ssrc; local
237 uint32 ssrc; local
[all...] |
| srtpfilter.h | 240 void AddProtectRtpResult(uint32 ssrc, int result);
242 void AddUnprotectRtpResult(uint32 ssrc, int result);
260 // For each different ssrc and error, we collect statistics separately.
263 : ssrc(0),
269 : ssrc(in_ssrc),
275 (ssrc < key.ssrc) ||
276 (ssrc == key.ssrc && mode < key.mode) ||
277 (ssrc == key.ssrc && mode == key.mode && error < key.error)
279 uint32 ssrc; member in struct:cricket::SrtpStat::FailureKey
[all...] |
| /external/chromium_org/chrome/browser/media/ |
| webrtc_browsertest_perf.cc | 14 // A ssrc stats key will be on the form stats.<bucket>-<key>.values.
24 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
26 if (!pc_dict.GetString(Statistic("audioOutputLevel", ssrc), &value)) {
31 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesReceived", ssrc), &value));
33 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value));
40 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
42 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) {
47 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value));
49 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
52 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), &value))
193 const std::string& ssrc = *ssrc_iterator; 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...] |
| rtputils.h | 43 uint32 ssrc; member in struct:cricket::RtpHeader
|
| rtputils_unittest.cc | 60 // PT = 206, FMT = 1, Sender SSRC = 0x1111, Media SSRC = 0x1111
67 // PT = 204, SSRC = 0x1111
97 uint32 ssrc; local
98 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc));
99 EXPECT_EQ(1u, ssrc);
106 EXPECT_EQ(1u, header.ssrc);
111 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc));
135 EXPECT_EQ(3333u, header.ssrc);
151 EXPECT_EQ(3333u, header.ssrc);
182 uint32 ssrc; local
[all...] |
| 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.
|
| filemediaengine.cc | 134 void SetSendSsrc(uint32 ssrc);
141 // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
219 void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
221 rtp_dump_reader_->SetSsrc(ssrc);
254 uint32 ssrc; local
255 if (!packet->GetRtpSsrc(&ssrc)) {
260 first_ssrc_ = ssrc;
262 if (ssrc == first_ssrc_) {
310 bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
311 if (ssrc != send_ssrc_
[all...] |
| streamparams.h | 35 // Let the simulcast elements have SSRC 10, 20, 30.
37 // SSRC 11,21,31.
39 // StreamParams would then contain ssrc = {10,11,20,21,30,31} and
80 static StreamParams CreateLegacy(uint32 ssrc) {
82 stream.ssrcs.push_back(ssrc);
110 bool has_ssrc(uint32 ssrc) const {
111 return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end();
113 void add_ssrc(uint32 ssrc) {
114 ssrcs.push_back(ssrc);
132 // Convenience function to add an FID ssrc for a primary_ssr
189 uint32 ssrc; member in struct:cricket::StreamSelector
[all...] |
| filemediaengine_unittest.cc | 191 uint32 ssrc; local
192 if (!packet.GetRtpSsrc(&ssrc)) {
195 ssrcs.insert(ssrc);
384 // Test that we can specify the ssrc for outgoing RTP packets.
|
| testutils.h | 79 uint32 ssrc; member in struct:cricket::RawRtpPacket
99 // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return
110 size_t count, talk_base::StreamInterface* stream, uint32 ssrc);
157 uint32 ssrc() const { return ssrc_; } function in class:cricket::ScreencastEventCatcher
159 void OnEvent(uint32 ssrc, talk_base::WindowEvent ev) {
160 ssrc_ = ssrc;
171 uint32 ssrc() const { return ssrc_; } function in class:cricket::VideoMediaErrorCatcher
173 void OnError(uint32 ssrc, VideoMediaChannel::Error error) {
174 ssrc_ = ssrc;
|
| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.h | 35 int ssrc; member in struct:CastRtpPayloadParams
|
| /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/net/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.cc | 29 ssrc(0),
66 uint32 rtp_timestamp, ssrc; local
68 big_endian_reader.ReadU32(&ssrc);
76 parsed_packet->ssrc = ssrc;
|
| rtp_header_parser.h | 31 uint32 ssrc; member in struct:media::cast::RtpCastTestHeader
|
| /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/third_party/libjingle/source/talk/app/webrtc/ |
| mediastreamhandler.h | 50 TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc);
57 uint32 ssrc() const { return ssrc_; } function in class:webrtc::TrackHandler
76 uint32 ssrc,
97 uint32 ssrc,
117 uint32 ssrc,
137 uint32 ssrc,
160 virtual void AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) = 0;
161 virtual void AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) = 0;
183 uint32 ssrc) OVERRIDE;
185 uint32 ssrc) OVERRIDE
[all...] |
| mediastreamsignaling.h | 68 uint32 ssrc) = 0;
73 uint32 ssrc) = 0;
86 uint32 ssrc) = 0;
91 uint32 ssrc) = 0;
125 // session description. This will set the SSRC used for sending data on
128 // session description. If the DataChannel label and a SSRC is included in
129 // the description, the DataChannel is updated with SSRC that will be used
131 // 4. When both the local and remote SSRC of a DataChannel is set the state of
136 // session description. If a label and a SSRC of a new DataChannel is found
137 // MediaStreamSignalingObserver::OnAddDataChannel with the label and SSRC i
288 uint32 ssrc; member in struct:webrtc::MediaStreamSignaling::TrackInfo
[all...] |
| /external/bluetooth/bluedroid/stack/avdt/ |
| avdt_api.c | 1295 UINT32 ssrc; local
[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),
|
