/external/webrtc/webrtc/modules/remote_bitrate_estimator/tools/ |
bwe_rtp.cc | 60 uint32_t ssrc; local 61 while (ss >> ssrc) { 62 ssrcs.insert(ssrc); 81 fprintf(stderr, "Filter on SSRC: ");
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
voip_metric.h | 38 void To(uint32_t ssrc) { ssrc_ = ssrc; } 43 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::VoipMetric
|
app.h | 36 void From(uint32_t ssrc) { ssrc_ = ssrc; } 42 uint32_t ssrc() const { return ssrc_; } function in class:webrtc::rtcp::App
|
dlrr.h | 27 uint32_t ssrc; member in struct:webrtc::rtcp::Dlrr::SubBlock 51 bool WithDlrrItem(uint32_t ssrc, uint32_t last_rr, uint32_t delay_last_rr);
|
/external/webrtc/webrtc/modules/rtp_rtcp/source/ |
ssrc_database.cc | 35 while (true) { // Try until get a new ssrc. 36 // 0 and 0xffffffff are invalid values for SSRC. 37 uint32_t ssrc = random_.Rand(1u, 0xfffffffe); local 38 if (ssrcs_.insert(ssrc).second) { 39 return ssrc; 44 void SSRCDatabase::RegisterSSRC(uint32_t ssrc) { 46 ssrcs_.insert(ssrc); 49 void SSRCDatabase::ReturnSSRC(uint32_t ssrc) { 51 ssrcs_.erase(ssrc);
|
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/webrtc/webrtc/video/ |
vie_remb_unittest.cc | 52 unsigned int ssrc = 1234; local 53 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 77 unsigned int ssrc = 1234; local 78 std::vector<unsigned int> ssrcs(&ssrc, &ssrc + 1); 103 unsigned int ssrc[] = { 1234, 5678 }; local 104 std::vector<unsigned int> ssrcs(ssrc, ssrc + sizeof(ssrc) / sizeof(ssrc[0])) 134 unsigned int ssrc[] = { 1234, 5678 }; local 168 unsigned int ssrc[] = { 1234, 5678 }; local 202 unsigned int ssrc = 1234; local 235 unsigned int ssrc = 1234; local [all...] |
encoder_state_feedback_unittest.cc | 41 void(uint32_t ssrc, uint8_t picture_id)); 43 void(uint32_t ssrc, uint64_t picture_id)); 68 const int ssrc = 1234; local 70 encoder_state_feedback_->AddEncoder(std::vector<uint32_t>(1, ssrc), &encoder); 72 EXPECT_CALL(encoder, OnReceivedIntraFrameRequest(ssrc)) 75 OnReceivedIntraFrameRequest(ssrc); 78 EXPECT_CALL(encoder, OnReceivedSLI(ssrc, sli_picture_id)) 81 ssrc, sli_picture_id); 84 EXPECT_CALL(encoder, OnReceivedRPSI(ssrc, rpsi_picture_id)) 87 ssrc, rpsi_picture_id) [all...] |
/external/webrtc/talk/media/base/ |
rtpdump_unittest.cc | 50 uint32_t ssrc; local 61 EXPECT_TRUE(rtp_packet.GetRtpSsrc(&ssrc)); 62 EXPECT_EQ(kTestSsrc, ssrc); 132 uint32_t ssrc; local 133 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 134 EXPECT_EQ(kTestSsrc, ssrc); 139 // Rewind the stream and read again with a specified ssrc. 148 uint32_t ssrc; local 149 EXPECT_TRUE(GetRtpSsrc(&packet.data[0], packet.data.size(), &ssrc)); 150 EXPECT_EQ(send_ssrc, ssrc); [all...] |
rtputils.h | 43 uint32_t 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 95 uint32_t ssrc; local 96 EXPECT_TRUE(GetRtpSsrc(kPcmuFrame, sizeof(kPcmuFrame), &ssrc)); 97 EXPECT_EQ(1u, ssrc); 104 EXPECT_EQ(1u, header.ssrc); 109 EXPECT_FALSE(GetRtpSsrc(kInvalidPacket, sizeof(kInvalidPacket), &ssrc)); 129 EXPECT_EQ(3333u, header.ssrc); 160 uint32_t ssrc; local [all...] |
streamparams_unittest.cc | 83 const uint32_t ssrc = 7; local 84 cricket::StreamParams one_sp = cricket::StreamParams::CreateLegacy(ssrc); 86 EXPECT_EQ(ssrc, one_sp.first_ssrc()); 88 EXPECT_TRUE(one_sp.has_ssrc(ssrc)); 89 EXPECT_FALSE(one_sp.has_ssrc(ssrc+1)); 260 // stream1 has extra non-sim, non-fid ssrc.
|
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_t ssrc) { 82 stream.ssrcs.push_back(ssrc); 110 bool has_ssrc(uint32_t ssrc) const { 111 return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end(); 113 void add_ssrc(uint32_t ssrc) { ssrcs.push_back(ssrc); } 130 // Convenience function to add an FID ssrc for a primary_ssr 197 uint32_t ssrc; member in struct:cricket::StreamSelector [all...] |
testutils.h | 80 uint32_t ssrc; member in struct:cricket::RawRtpPacket 100 // depending on the flag rtcp. If it is RTP, use the specified SSRC. Return 114 uint32_t ssrc); 161 uint32_t ssrc() const { return ssrc_; } function in class:cricket::ScreencastEventCatcher 163 void OnEvent(uint32_t ssrc, rtc::WindowEvent ev) { 164 ssrc_ = ssrc; 175 uint32_t ssrc() const { return ssrc_; } function in class:cricket::VideoMediaErrorCatcher 177 void OnError(uint32_t ssrc, VideoMediaChannel::Error error) { 178 ssrc_ = ssrc;
|
/external/webrtc/webrtc/call/ |
rtc_event_log2rtp_dump.cc | 46 DEFINE_string(ssrc, 48 "Store only packets with this SSRC (decimal or hex, the latter " 51 // Parses the input string for a valid SSRC. If a valid SSRC is found, it is 52 // written to the output variable |ssrc|, and true is returned. Otherwise, 57 bool ParseSsrc(std::string str, uint32_t* ssrc) { 66 ss >> read_mode >> *ssrc; local
|
/external/webrtc/talk/session/media/ |
currentspeakermonitor.cc | 91 uint32_t ssrc = stream_list_it->first; local 92 active_ssrc_to_level_map[ssrc] = stream_list_it->second; 96 if (ssrc_to_speaking_state_map_.find(ssrc) == 98 ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
|
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/ |
rtp_rtcp_test.cc | 28 unsigned int SSRC); 33 void SetIncomingSsrc(unsigned int ssrc) { 35 incoming_ssrc_ = ssrc; 44 unsigned int SSRC) { 46 sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel, 47 SSRC); 52 if (incoming_ssrc_ == SSRC) 75 // We'll set up the RTCP CNAME and SSRC to something arbitrary here. 112 unsigned int ssrc; local 113 EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc)); [all...] |
/external/webrtc/talk/app/webrtc/ |
peerconnection.h | 147 TrackInfo() : ssrc(0) {} 150 uint32_t ssrc) 151 : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {} 154 this->track_id == other.track_id && this->ssrc == other.ssrc; 158 uint32_t ssrc; member in struct:webrtc::PeerConnection::TrackInfo 167 uint32_t ssrc); 170 uint32_t ssrc); 177 uint32_t ssrc); [all...] |
/external/webrtc/webrtc/audio/ |
audio_send_stream.cc | 33 ss << "{ssrc: " << ssrc; local 76 channel_proxy_->SetLocalSSRC(config.rtp.ssrc); 128 stats.local_ssrc = config_.rtp.ssrc; 152 // Lookup report for send ssrc only.
|
/external/webrtc/webrtc/ |
audio_send_stream.h | 64 // Sender SSRC. 65 uint32_t ssrc = 0; member in struct:webrtc::AudioSendStream::Config::Rtp
|
/external/webrtc/webrtc/modules/remote_bitrate_estimator/ |
remote_bitrate_estimator_single_stream.cc | 74 uint32_t ssrc = header.ssrc; local 79 SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc); 81 // This is a new SSRC. Adding to map. 82 // TODO(holmer): If the channel changes SSRC the old SSRC will still be 84 // callback will no longer be called for the old SSRC. This will be 89 ssrc, new Detector(now_ms, OverUseDetectorOptions(), true))); 193 void RemoteBitrateEstimatorSingleStream::RemoveStream(unsigned int ssrc) { 195 SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc); [all...] |
/external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
conference_transport.cc | 29 unsigned int ssrc = 0; local 30 if (len >= (ssrc_pos + sizeof(ssrc))) { 31 ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos); 33 return ssrc; 152 if (rtp_header.ssrc == kLocalSsrc) { 157 destination = GetReceiverChannelForSsrc(rtp_header.ssrc); 243 // Receive channels have to have the same SSRC in order to send receiver 244 // reports with this SSRC. 251 return remote_ssrc; // remote ssrc used as stream id.
|
/external/webrtc/webrtc/voice_engine/test/auto_test/fixtures/ |
after_initialization_fixture.h | 64 void AddChannel(uint32_t ssrc, int channel) { 66 channels_[ssrc] = channel; 120 uint32_t ssrc = local 122 if (channels_[ssrc] != 0) 123 channel = channels_[ssrc]; 125 // TODO(pbos): Add RTCP SSRC muxing/demuxing if anything requires it.
|
/external/webrtc/talk/media/webrtc/ |
fakewebrtccall.cc | 274 const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) { 276 if (p->GetConfig().rtp.ssrc == ssrc) { 287 const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) { 289 if (p->GetConfig().rtp.remote_ssrc == ssrc) { 392 uint32_t ssrc; local 393 if (!GetRtpSsrc(packet, length, &ssrc)) 399 if (receiver->GetConfig().rtp.remote_ssrc == ssrc) 406 if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
|