/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
D | rtpdataengine.cc | 178 if (GetStreamBySsrc(send_streams_, stream.first_ssrc(), &found_stream)) { in AddSendStream() 180 << "' with ssrc=" << stream.first_ssrc() in AddSendStream() 188 rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock( in AddSendStream() 193 << "' with ssrc=" << stream.first_ssrc(); in AddSendStream() 215 if (GetStreamBySsrc(recv_streams_, stream.first_ssrc(), &found_stream)) { in AddRecvStream() 217 << "' with ssrc=" << stream.first_ssrc() in AddRecvStream() 224 << "' with ssrc=" << stream.first_ssrc(); in AddRecvStream()
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D | fakemediaengine.h | 151 return muted_streams_.find(send_streams_[0].first_ssrc()) != in IsStreamMuted() 173 return send_streams_[0].first_ssrc(); in send_ssrc() 289 output_scalings_[sp.first_ssrc()] = OutputScaling(); in AddRecvStream() 513 SetSendStreamDefaultFormat(sp.first_ssrc()); in AddSendStream() 538 SetSendStreamDefaultFormat(it->first_ssrc()); in SetSendCodecs() 582 renderers_[sp.first_ssrc()] = NULL; in AddRecvStream()
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D | streamparams_unittest.cc | 83 EXPECT_EQ(ssrc, one_sp.first_ssrc()); in TEST() 95 EXPECT_EQ(kSsrcs2[0], sp.first_ssrc()); in TEST()
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D | streamparams.h | 100 uint32 first_ssrc() const { in first_ssrc() function
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D | streamparams.cc | 103 ssrcs->push_back(first_ssrc()); in GetPrimarySsrcs()
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
D | channel.cc | 1040 LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w() 1043 desc << "Failed to add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w() 1048 if (!media_channel()->RemoveSendStream(existing_stream.first_ssrc())) { in UpdateLocalStreams_w() 1051 << it->first_ssrc() << "."; in UpdateLocalStreams_w() 1055 RemoveStreamBySsrc(&local_streams_, existing_stream.first_ssrc()); in UpdateLocalStreams_w() 1068 if (!GetStreamBySsrc(streams, it->first_ssrc(), NULL)) { in UpdateLocalStreams_w() 1069 if (!media_channel()->RemoveSendStream(it->first_ssrc())) { in UpdateLocalStreams_w() 1072 << it->first_ssrc() << "."; in UpdateLocalStreams_w() 1081 if (!GetStreamBySsrc(local_streams_, it->first_ssrc(), NULL)) { in UpdateLocalStreams_w() 1086 desc << "Failed to add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w() [all …]
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D | currentspeakermonitor.cc | 204 ssrc_to_speaking_state_map_.erase(it->first_ssrc()); in OnMediaStreamsUpdate() 209 ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING; in OnMediaStreamsUpdate()
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D | bundlefilter.cc | 87 if (GetStreamBySsrc(streams_, stream.first_ssrc(), NULL)) { in AddStream()
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D | mediasession_unittest.cc | 470 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 497 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 504 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F() 605 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 612 EXPECT_NE(0U, dcd->first_ssrc()); // a random nonzero ssrc in TEST_F() 712 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 742 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 747 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F() 775 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F() 780 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F() [all …]
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D | mediasession.h | 259 uint32 first_ssrc() const { in first_ssrc() function 263 return streams_[0].first_ssrc(); in first_ssrc()
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D | mediasessionclient_unittest.cc | 1695 ASSERT_EQ(kAudioSsrc, audio->first_ssrc()); in CheckAudioSsrcForIncomingAccept() 1702 ASSERT_EQ(kVideoSsrc, video->first_ssrc()); in CheckVideoSsrcForIncomingAccept() 1709 ASSERT_EQ(kDataSsrc, data->first_ssrc()); in CheckDataSsrcForIncomingAccept() 2530 ASSERT_EQ(1234U, last_streams_added_.audio()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2538 ASSERT_EQ(1234U, last_streams_added_.audio()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2553 ASSERT_EQ(2468U, last_streams_added_.audio()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2562 ASSERT_EQ(5678U, last_streams_added_.video()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2581 ASSERT_EQ(5679U, last_streams_added_.video()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2612 EXPECT_EQ(1234U, last_streams_removed_.audio()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() 2620 EXPECT_EQ(5678U, last_streams_removed_.video()[0].first_ssrc()); in TestStreamsUpdateAndViewRequests() [all …]
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D | mediasessionclient.cc | 851 QN_GINGLE_AUDIO_SRCID, audio->first_ssrc())); in CreateGingleAudioContentElem() 875 QN_GINGLE_VIDEO_SRCID, video->first_ssrc())); in CreateGingleVideoContentElem() 974 AddXmlAttr(elem, QN_SSRC, media->first_ssrc()); in WriteLegacyJingleSsrc()
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D | call.cc | 1045 channel->RemoveRecvStream(stream.first_ssrc()); in RemoveRecvStream()
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
D | mediastreamsignaling.cc | 622 uint32 ssrc = it->first_ssrc(); in UpdateRemoteStreamsList() 637 OnRemoteTrackSeen(stream_label, track_id, it->first_ssrc(), media_type); in UpdateRemoteStreamsList() 810 uint32 ssrc = it->first_ssrc(); in UpdateLocalTracks() 816 OnLocalTrackSeen(stream_label, track_id, it->first_ssrc(), in UpdateLocalTracks() 906 data_channel_it->second->SetSendSsrc(it->first_ssrc()); in UpdateLocalRtpDataChannels() 923 rtc::ToString(it->first_ssrc()) : it->sync_label; in UpdateRemoteRtpDataChannels() 928 CreateRemoteDataChannel(label, it->first_ssrc()); in UpdateRemoteRtpDataChannels() 930 data_channel_it->second->SetReceiveSsrc(it->first_ssrc()); in UpdateRemoteRtpDataChannels()
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D | webrtcsession_unittest.cc | 2546 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc(); in TEST_F() 2571 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() 2601 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() 2626 uint32 receive_ssrc = channel->recv_streams()[0].first_ssrc(); in TEST_F() 2641 uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F() 2675 const uint32 send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F()
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D | peerconnectioninterface_unittest.cc | 105 *ssrc = media_desc->streams().begin()->first_ssrc(); in GetFirstSsrc()
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D | webrtcsession.cc | 216 *ssrc = stream.first_ssrc(); in GetAudioSsrcByTrackId()
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/external/chromium_org/third_party/libjingle/source/talk/media/webrtc/ |
D | webrtcvideoengine.cc | 1886 if (sp.first_ssrc() == 0) { in AddSendStream() 1899 if (!CreateSendChannelSsrcKey(sp.first_ssrc(), &ssrc_key)) { in AddSendStream() 1900 LOG(LS_ERROR) << "Trying to register duplicate ssrc: " << sp.first_ssrc(); in AddSendStream() 1931 SetReceiverReportSsrc(sp.first_ssrc()); in AddSendStream() 1996 if (sp.first_ssrc() == 0) { in AddRecvStream() 2005 LOG(LS_INFO) << "Recv stream " << sp.first_ssrc() in AddRecvStream() 2008 first_receive_ssrc_ = sp.first_ssrc(); in AddRecvStream() 2021 uint32 ssrc = sp.first_ssrc(); in AddRecvStream() 2036 if (!CreateChannel(sp.first_ssrc(), MD_RECV, &channel_id)) { in AddRecvStream() 2052 LOG(LS_INFO) << "New video stream " << sp.first_ssrc() in AddRecvStream() [all …]
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D | webrtcvoiceengine.cc | 2530 int channel = GetSendChannelNum(sp.first_ssrc()); in SetSendCodecs() 2532 LOG(LS_ERROR) << "Stream already exists with ssrc " << sp.first_ssrc(); in SetSendCodecs() 2562 sp.first_ssrc(), in SetSendCodecs() 2569 if (engine()->voe()->rtp()->SetLocalSSRC(channel, sp.first_ssrc()) == -1) { in SetSendCodecs() 2570 LOG_RTCERR2(SetSendSSRC, channel, sp.first_ssrc()); in SetSendCodecs() 2584 sp.first_ssrc()) != 0) { in SetSendCodecs() 2585 LOG_RTCERR2(SetLocalSSRC, it->second->channel(), sp.first_ssrc()); in SetSendCodecs() 2643 uint32 ssrc = sp.first_ssrc(); in SetSendCodecs() 2660 LOG(LS_INFO) << "Recv stream " << sp.first_ssrc() in SetSendCodecs() 2662 default_receive_ssrc_ = sp.first_ssrc(); in SetSendCodecs()
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D | webrtcvideoengine2.cc | 883 uint32 ssrc = sp.first_ssrc(); in AddSendStream() 959 uint32 ssrc = sp.first_ssrc(); in AddRecvStream() 979 uint32 ssrc = sp.first_ssrc(); in ConfigureReceiverRtp()
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/external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
D | callclient.cc | 630 session->id().c_str(), speaker.first_ssrc()); in OnSpeakerChanged() 634 speaker.first_ssrc()); in OnSpeakerChanged() 783 params.ssrc = stream.first_ssrc(); in SendData() 1434 RemoveStaticRenderedView(it->first_ssrc()); in OnMediaStreamsUpdate() 1475 uint32 ssrc = stream.first_ssrc(); in RenderStream()
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/external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
D | sctpdataengine.cc | 649 const uint32 ssrc = stream.first_ssrc(); in AddStream()
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