| /external/webrtc/talk/media/base/ |
| D | rtpdataengine.cc | 171 if (GetStreamBySsrc(send_streams_, stream.first_ssrc())) { in AddSendStream()
173 << "' with ssrc=" << stream.first_ssrc() in AddSendStream()
181 rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock( in AddSendStream()
186 << "' with ssrc=" << stream.first_ssrc(); in AddSendStream()
206 if (GetStreamBySsrc(recv_streams_, stream.first_ssrc())) { in AddRecvStream()
208 << "' with ssrc=" << stream.first_ssrc() in AddRecvStream()
215 << "' with ssrc=" << stream.first_ssrc(); in AddRecvStream()
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| D | fakemediaengine.h | 135 return muted_streams_.find(send_streams_[0].first_ssrc()) != in IsStreamMuted()
157 return send_streams_[0].first_ssrc(); in send_ssrc()
293 output_scalings_[sp.first_ssrc()] = 1.0; in AddRecvStream()
491 SetSendStreamDefaultFormat(sp.first_ssrc()); in AddSendStream()
537 renderers_[sp.first_ssrc()] = NULL; in AddRecvStream()
580 SetSendStreamDefaultFormat(it->first_ssrc()); in SetSendCodecs()
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| D | streamparams_unittest.cc | 86 EXPECT_EQ(ssrc, one_sp.first_ssrc()); in TEST()
98 EXPECT_EQ(kSsrcs2[0], sp.first_ssrc()); in TEST()
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| D | streamparams.cc | 167 ssrcs->push_back(first_ssrc()); in GetPrimarySsrcs()
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| D | streamparams.h | 100 uint32_t first_ssrc() const { in first_ssrc() function
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| /external/webrtc/talk/session/media/ |
| D | channel.cc | 1161 LOG(LS_INFO) << "Add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w()
1164 desc << "Failed to add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w()
1169 if (!media_channel()->RemoveSendStream(existing_stream->first_ssrc())) { in UpdateLocalStreams_w()
1172 << it->first_ssrc() << "."; in UpdateLocalStreams_w()
1176 RemoveStreamBySsrc(&local_streams_, existing_stream->first_ssrc()); in UpdateLocalStreams_w()
1189 if (!GetStreamBySsrc(streams, it->first_ssrc())) { in UpdateLocalStreams_w()
1190 if (!media_channel()->RemoveSendStream(it->first_ssrc())) { in UpdateLocalStreams_w()
1193 << it->first_ssrc() << "."; in UpdateLocalStreams_w()
1202 if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) { in UpdateLocalStreams_w()
1207 desc << "Failed to add send stream ssrc: " << it->first_ssrc(); in UpdateLocalStreams_w()
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| D | currentspeakermonitor.cc | 201 ssrc_to_speaking_state_map_.erase(it->first_ssrc()); in OnMediaStreamsUpdate()
206 ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING; in OnMediaStreamsUpdate()
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| D | mediasession_unittest.cc | 496 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
523 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
530 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F()
632 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
639 EXPECT_NE(0U, dcd->first_ssrc()); // a random nonzero ssrc in TEST_F()
784 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
814 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
819 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F()
847 EXPECT_NE(0U, acd->first_ssrc()); // a random nonzero ssrc in TEST_F()
852 EXPECT_NE(0U, vcd->first_ssrc()); // a random nonzero ssrc in TEST_F()
[all …]
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| D | mediasession.h | 270 uint32_t first_ssrc() const { in first_ssrc() function
274 return streams_[0].first_ssrc(); in first_ssrc()
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| /external/webrtc/webrtc/video/ |
| D | vie_encoder.cc | 581 uint32_t first_ssrc; in OnNetworkChanged() local
588 first_ssrc = ssrc_streams_.begin()->first; in OnNetworkChanged()
601 << " for ssrc " << first_ssrc; in OnNetworkChanged()
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| D | video_send_stream.cc | 170 std::vector<uint32_t> first_ssrc(1, ssrcs[0]); in VideoSendStream() local
171 vie_encoder_->SetSsrcs(first_ssrc); in VideoSendStream()
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| /external/webrtc/talk/app/webrtc/ |
| D | peerconnection.cc | 1529 uint32_t ssrc = params.first_ssrc(); in UpdateRemoteStreamsList()
1685 uint32_t ssrc = params.first_ssrc(); in UpdateLocalTracks()
1690 OnLocalTrackSeen(stream_label, track_id, params.first_ssrc(), media_type); in UpdateLocalTracks()
1756 data_channel_it->second->SetSendSsrc(params.first_ssrc()); in UpdateLocalRtpDataChannels()
1772 ? rtc::ToString(params.first_ssrc()) in UpdateRemoteRtpDataChannels()
1777 CreateRemoteRtpDataChannel(label, params.first_ssrc()); in UpdateRemoteRtpDataChannels()
1779 data_channel_it->second->SetReceiveSsrc(params.first_ssrc()); in UpdateRemoteRtpDataChannels()
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| D | webrtcsession_unittest.cc | 3384 uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc(); in TEST_F()
3403 uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F()
3429 uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F()
3454 uint32_t receive_ssrc = channel->recv_streams()[0].first_ssrc(); in TEST_F()
3469 uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F()
3502 const uint32_t send_ssrc = channel->send_streams()[0].first_ssrc(); in TEST_F()
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| D | webrtcsession.cc | 291 *ssrc = stream->first_ssrc(); in GetAudioSsrcByTrackId()
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| D | peerconnectioninterface_unittest.cc | 301 *ssrc = media_desc->streams().begin()->first_ssrc(); in GetFirstSsrc()
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| /external/webrtc/talk/media/sctp/ |
| D | sctpdataengine.cc | 731 const uint32_t ssrc = stream.first_ssrc(); in AddStream()
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| /external/webrtc/talk/media/webrtc/ |
| D | webrtcvideoengine2.cc | 976 uint32_t ssrc = sp.first_ssrc(); in AddSendStream()
1068 uint32_t ssrc = sp.first_ssrc(); in AddRecvStream()
1110 uint32_t ssrc = sp.first_ssrc(); in ConfigureReceiverRtp()
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| D | webrtcvoiceengine.cc | 1862 uint32_t ssrc = sp.first_ssrc(); in AddSendStream()
1947 const uint32_t ssrc = sp.first_ssrc(); in AddRecvStream()
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