| /external/chromium_org/media/cast/transport/rtp_sender/ |
| rtp_sender.h | 63 uint32 ssrc() const { return config_.ssrc; } function in class:media::cast::transport::RtpSender
|
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| bundlefilter.cc | 45 // For rtcp packets, we check whether the ssrc can be found or is the special
57 // Rtcp packets using ssrc filter.
59 uint32 ssrc = 0; local
66 if (!GetRtcpSsrc(data, len, &ssrc)) return false;
67 if (ssrc == kSsrc01) {
68 // SSRC 1 has a special meaning and indicates generic feedback on
75 return !HasStreams() || FindStream(ssrc);
91 bool BundleFilter::RemoveStream(uint32 ssrc) {
92 return RemoveStreamBySsrc(&streams_, ssrc);
99 bool BundleFilter::FindStream(uint32 ssrc) const
[all...] |
| currentspeakermonitor.cc | 94 uint32 ssrc = stream_list_it->first; local
95 active_ssrc_to_level_map[ssrc] = stream_list_it->second;
99 if (ssrc_to_speaking_state_map_.find(ssrc) ==
101 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| ssrc_database.cc | 58 uint32_t ssrc = GenerateRandom(); local
62 while(_ssrcMap.find(ssrc) != _ssrcMap.end())
64 ssrc = GenerateRandom();
66 _ssrcMap[ssrc] = 0;
86 if (_ssrcVector[i] == ssrc)
89 i = 0; // start over with a new ssrc
90 ssrc = GenerateRandom();
95 _ssrcVector[_numberOfSSRC] = ssrc;
99 return ssrc;
103 SSRCDatabase::RegisterSSRC(const uint32_t ssrc)
201 uint32_t ssrc = 0; local
[all...] |
| forward_error_correction.h | 81 // The ssrc member is needed to ensure we can restore the SSRC field of
91 uint32_t ssrc; // SSRC of the current frame. Must be set for FEC member in class:webrtc::ForwardErrorCorrection::ReceivedPacket
|
| tmmbr_help.h | 41 uint32_t Ssrc(int i) const {
42 return _data.at(i).ssrc;
65 SetElement() : tmmbr(0), packet_oh(0), ssrc(0) {}
68 uint32_t ssrc; member in class:webrtc::TMMBRSet::SetElement
94 bool IsOwner(const uint32_t ssrc, const uint32_t length) const;
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_remb_unittest.cc | 50 unsigned int ssrc = 1234; local
51 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
75 unsigned int ssrc = 1234; local
76 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1);
101 unsigned int ssrc[] = { 1234, 5678 }; local
102 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0]))
132 unsigned int ssrc[] = { 1234, 5678 }; local
166 unsigned int ssrc[] = { 1234, 5678 }; local
200 unsigned int ssrc = 1234; local
233 unsigned int ssrc = 1234; local
[all...] |
| encoder_state_feedback_unittest.cc | 37 void(uint32_t ssrc, uint8_t picture_id));
39 void(uint32_t ssrc, uint64_t picture_id));
57 const int ssrc = 1234; local
59 EXPECT_TRUE(encoder_state_feedback_->AddEncoder(ssrc, &encoder));
61 EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc))
64 OnReceivedIntraFrameRequest(ssrc);
67 EXPECT_CALL(encoder, OnReceivedSLI(ssrc, sli_picture_id))
70 ssrc, sli_picture_id);
73 EXPECT_CALL(encoder, OnReceivedRPSI(ssrc, rpsi_picture_id))
76 ssrc, rpsi_picture_id)
131 const int ssrc = 1234; local
[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));
34 EXPECT_TRUE(pc_dict.GetString(Statistic("packetsLost", ssrc), &value));
42 const std::string& ssrc, const base::DictionaryValue& pc_dict) {
44 if (!pc_dict.GetString(Statistic("audioInputLevel", ssrc), &value)) {
49 EXPECT_TRUE(pc_dict.GetString(Statistic("bytesSent", ssrc), &value));
52 EXPECT_TRUE(pc_dict.GetString(Statistic("googJitterReceived", ssrc), &value));
55 EXPECT_TRUE(pc_dict.GetString(Statistic("googRtt", ssrc), &value))
231 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...] |
| /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.
90 'ssrc': true,
281 if (report.type == 'ssrc') {
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/ |
| rtp_packetizer.h | 37 // SSRC.
38 unsigned int ssrc; member in struct:media::cast::transport::RtpPacketizerConfig
|
| /external/chromium_org/media/cast/transport/rtp_sender/rtp_packetizer/test/ |
| rtp_header_parser.cc | 30 ssrc(0),
66 uint32 rtp_timestamp, ssrc; local
68 big_endian_reader.ReadU32(&ssrc);
76 parsed_packet->ssrc = ssrc;
|
| rtp_header_parser.h | 33 uint32 ssrc; member in struct:media::cast::transport::RtpCastTestHeader
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.cc | 82 TEST_F(RtpRtcpAPITest, SSRC) {
84 EXPECT_EQ(test_ssrc, module->SSRC());
118 unsigned int ssrc = 0; local
125 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
127 EXPECT_EQ(1u, ssrc);
133 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
137 module->RTXSendStatus(&rtx_mode, &ssrc, &payload_type);
150 rtx_header.ssrc = kRtxSsrc;
153 rtx_header.ssrc = 0;
155 rtx_header.ssrc = kRtxSsrc
[all...] |
| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.h | 46 int ssrc; member in struct:CastRtpPayloadParams
|
| /external/chromium_org/media/cast/transport/pacing/ |
| paced_sender.cc | 29 uint32 ssrc,
31 return std::make_pair(ticks, std::make_pair(ssrc, packet_id));
101 bool PacedSender::SendRtcpPacket(uint32 ssrc, PacketRef packet) {
103 packet_list_[PacedPacketSender::MakePacketKey(base::TimeTicks(), ssrc, 0)] =
236 // Get SSRC from packet and compare with the audio_ssrc / video_ssrc to see
240 uint32 ssrc; local
241 bool success = reader.ReadU32(&ssrc);
244 if (ssrc == audio_ssrc_) {
246 } else if (ssrc == video_ssrc_) {
249 DVLOG(3) << "Got unknown ssrc " << ssrc << " when logging packet event"
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| mediastreamhandler.h | 51 TrackHandler(MediaStreamTrackInterface* track, uint32 ssrc);
58 uint32 ssrc() const { return ssrc_; } function in class:webrtc::TrackHandler
99 uint32 ssrc,
125 uint32 ssrc,
148 uint32 ssrc,
168 uint32 ssrc,
191 virtual void AddAudioTrack(AudioTrackInterface* audio_track, uint32 ssrc) = 0;
192 virtual void AddVideoTrack(VideoTrackInterface* video_track, uint32 ssrc) = 0;
214 uint32 ssrc) OVERRIDE;
216 uint32 ssrc) OVERRIDE
[all...] |
| mediastreamsignaling.h | 69 uint32 ssrc) = 0;
74 uint32 ssrc) = 0;
87 uint32 ssrc) = 0;
92 uint32 ssrc) = 0;
97 uint32 ssrc) = 0;
127 // session description. This will set the SSRC used for sending data on
130 // session description. If the DataChannel label and a SSRC is included in
131 // the description, the DataChannel is updated with SSRC that will be used
133 // 4. When both the local and remote SSRC of a DataChannel is set the state of
138 // session description. If a label and a SSRC of a new DataChannel is foun
288 uint32 ssrc; member in struct:webrtc::MediaStreamSignaling::TrackInfo
[all...] |
| /external/chromium_org/third_party/webrtc/ |
| config.h | 26 : ssrc(0),
30 uint32_t ssrc; member in struct:webrtc::RtpStatistics
|
