| /external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
| webrtcvideoengine.cc | 154 FlushBlackFrameData(uint32 s, int64 t) : ssrc(s), timestamp(t) {
156 uint32 ssrc; member in struct:cricket::FlushBlackFrameData
2616 unsigned int ssrc = 0; local
2761 uint32 ssrc = 0; local
2791 uint32 ssrc = 0; local
3735 const uint32 ssrc = send_channel->stream_params()->first_ssrc(); local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| webrtcsession.cc | 198 const std::string& track_id, uint32 *ssrc) {
214 *ssrc = stream.first_ssrc();
219 uint32 ssrc, std::string* track_id) {
232 if (cricket::GetStreamBySsrc(audio_content->streams(), ssrc, &stream_out)) {
246 if (cricket::GetStreamBySsrc(video_content->streams(), ssrc, &stream_out)) {
756 // Update state and SSRC of local MediaStreams and DataChannels based on the
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| rtp_sender.cc | 326 void RTPSender::SetRtxSsrc(uint32_t ssrc) {
328 ssrc_rtx_ = ssrc;
331 void RTPSender::RTXStatus(int* mode, uint32_t* ssrc,
335 *ssrc = ssrc_rtx_;
551 uint32_t ssrc; local
560 ssrc = ssrc_;
564 ssrc = ssrc_rtx_;
570 int header_length = CreateRTPHeader(padding_packet, payload_type, ssrc,
614 header.ssrc,
824 // Get ssrc before taking statistics_crit_ to avoid possible deadlock
825 uint32_t ssrc = SSRC(); local
[all...] |
| rtp_sender_unittest.cc | 114 EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.ssrc);
826 uint32_t ssrc = rtp_sender_->SSRC(); local
882 uint32_t ssrc = rtp_sender_->SSRC(); local
968 uint32_t ssrc = rtp_sender_->SSRC(); local
[all...] |
| rtcp_receiver_unittest.cc | 37 ReportBlock(uint32_t ssrc, uint32_t extended_max, uint8_t fraction_loss,
39 : ssrc(ssrc),
45 uint32_t ssrc; member in struct:webrtc::__anon19898::PacketBuilder::ReportBlock
127 AddReportBlock(it->ssrc, it->extended_max, it->fraction_loss,
162 Add32(remote_ssrc.at(i)); // Receiver SSRC.
172 Add32(0); // Receiver SSRC.
183 Add32(remote_ssrc); // Receiver SSRC.
344 // The parser will note the remote SSRC on a SR from other than his
660 EXPECT_EQ(kMediaRecipientSsrc, candidate_set.Ssrc(0))
[all...] |
| rtp_sender.h | 43 virtual uint32_t SSRC() const = 0;
120 void SetSSRC(const uint32_t ssrc);
190 void RTXStatus(int* mode, uint32_t* ssrc, int* payload_type) const;
192 void SetRtxSsrc(uint32_t ssrc);
211 virtual uint32_t SSRC() const OVERRIDE;
291 uint32_t ssrc, bool marker_bit,
|
| tmmbr_help.cc | 45 _data.at(i).ssrc = 0;
69 _data.at(i).ssrc = ssrcSet;
169 boundingSetToSend->Ssrc(i));
220 _candidateSet.Ssrc(i));
265 candidateSet.Ssrc(i));
344 candidateSet.Ssrc(minIndexTMMBR));
392 curCandidateSSRC = candidateSet.Ssrc(i);
446 bool TMMBRHelp::IsOwner(const uint32_t ssrc,
456 if(_boundingSet.Ssrc(i) == ssrc) {
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| mediasession.h | 241 // Legacy streams have an ssrc, but nothing else.
242 void AddLegacyStream(uint32 ssrc) {
243 streams_.push_back(StreamParams::CreateLegacy(ssrc));
245 void AddLegacyStream(uint32 ssrc, uint32 fid_ssrc) {
246 StreamParams sp = StreamParams::CreateLegacy(ssrc);
247 sp.AddFidSsrc(ssrc, fid_ssrc);
493 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/third_party/webrtc/ |
| common_types.h | 197 uint32_t ssrc) = 0;
242 uint32_t ssrc) = 0;
259 virtual void Notify(const BitrateStatistics& stats, uint32_t ssrc) = 0;
268 const unsigned int ssrc) = 0;
764 ssrc(0),
777 uint32_t ssrc; member in struct:webrtc::RTPHeader
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_encoder.cc | 121 virtual bool TimeToSendPacket(uint32_t ssrc, uint16_t sequence_number,
123 return owner_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
425 bool ViEEncoder::TimeToSendPacket(uint32_t ssrc,
429 return default_rtp_rtcp_->TimeToSendPacket(ssrc, sequence_number,
526 tempCSRC[i] = default_rtp_rtcp_->SSRC();
747 void ViEEncoder::OnReceivedSLI(uint32_t /*ssrc*/,
753 void ViEEncoder::OnReceivedRPSI(uint32_t /*ssrc*/,
759 void ViEEncoder::OnReceivedIntraFrameRequest(uint32_t ssrc) {
766 std::map<unsigned int, int>::iterator stream_it = ssrc_streams_.find(ssrc);
768 LOG_F(LS_WARNING) << "ssrc not found: " << ssrc << ", map size
825 unsigned int ssrc = *it; local
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump.h | 86 // Get the payload type, sequence number, timestampe, and SSRC of the RTP
91 bool GetRtpSsrc(uint32* ssrc) const;
116 // Use the specified ssrc, rather than the ssrc from dump, for RTP packets.
117 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.
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
| packet.cc | 142 destination->ssrc = header_.ssrc;
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/H264/ |
| rtp_sender_h264.h | 83 const uint32_t ssrc);
92 const uint32_t ssrc,
|
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
| test_util.cc | 85 const std::string& base_out_filename, uint32_t ssrc)
102 ssrc << "." << extension;
|
| pcap_file_reader_unittest.cc | 61 pps[header.ssrc]++;
|
| /external/chromium_org/chrome/renderer/media/ |
| cast_rtp_stream.h | 46 int ssrc; member in struct:CastRtpPayloadParams
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
| EncodeDecodeTest.h | 39 int16_t seqNo, uint32_t timeStamp, uint32_t ssrc);
|
| RTPFile.h | 41 uint32_t timeStamp, uint32_t ssrc);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
| RTPencode.cc | 79 void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, uint32_t timestamp, uint32_t ssrc);
81 int seqNo, uint32_t ssrc);
255 uint32_t ssrc=1235412312; local
581 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
586 makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
652 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
668 makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
678 makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
680 makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/pacing/ |
| paced_sender.cc | 38 Packet(uint32_t ssrc, uint16_t seq_number, int64_t capture_time_ms,
40 : ssrc_(ssrc),
172 bool PacedSender::SendPacket(Priority priority, uint32_t ssrc,
201 packet_list->push_back(paced_sender::Packet(ssrc,
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| mock_rtp_rtcp.h | 40 MOCK_METHOD1(SetRemoteSSRC, void(const uint32_t ssrc));
77 MOCK_CONST_METHOD0(SSRC,
80 void(const uint32_t ssrc));
90 void(int* modes, uint32_t* ssrc, int* payload_type));
119 bool(uint32_t ssrc, uint16_t sequence_number, int64_t capture_time_ms,
147 int32_t(const uint32_t SSRC,
150 int32_t(const uint32_t SSRC));
170 int32_t(const uint32_t SSRC, const RTCPReportBlock* receiveBlock));
172 int32_t(const uint32_t SSRC));
188 int32_t(const uint32_t bitrate, const uint8_t numberOfSSRC, const uint32_t* SSRC));
[all...] |
| /external/chromium_org/third_party/webrtc/modules/video_coding/main/source/ |
| video_receiver_unittest.cc | 98 header.header.ssrc = 1;
122 header.header.ssrc = 1;
174 header.header.ssrc = 1;
|
| /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...] |
