| /external/webrtc/webrtc/video/ |
| report_block_stats.h | 36 uint32_t remote_ssrc,
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| report_block_stats.cc | 34 uint32_t remote_ssrc,
41 block.remoteSSRC = remote_ssrc;
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| video_receive_stream.cc | 73 ss << "{remote_ssrc: " << remote_ssrc; local
181 RTC_DCHECK(config_.rtp.remote_ssrc != 0);
184 RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
252 new ReceiveStatisticsProxy(config_.rtp.remote_ssrc, clock_));
299 uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC(); local
302 RemoveStream(remote_ssrc);
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| vie_channel.cc | 760 uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
763 if (it->remoteSSRC == remote_ssrc)
772 remote_ssrc = report_blocks[0].remoteSSRC;
786 if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) !=
[all...] |
| video_send_stream.cc | 270 uint32_t remote_ssrc = vie_channel_->GetRemoteSSRC(); local
272 remote_ssrc);
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| replay.cc | 221 receive_config.rtp.remote_ssrc = flags::Ssrc();
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| /external/webrtc/webrtc/ |
| audio_receive_stream.h | 37 uint32_t remote_ssrc = 0; member in struct:webrtc::AudioReceiveStream::Stats
71 uint32_t remote_ssrc = 0; member in struct:webrtc::AudioReceiveStream::Config::Rtp
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| video_receive_stream.h | 91 uint32_t remote_ssrc = 0; member in struct:webrtc::VideoReceiveStream::Config::Rtp
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| /external/webrtc/webrtc/call/ |
| call_unittest.cc | 57 config.rtp.remote_ssrc = 42;
94 config.rtp.remote_ssrc = ssrc;
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| bitrate_estimator_tests.cc | 133 receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
186 receive_config.rtp.remote_ssrc = test_->video_send_config_.rtp.ssrcs[0];
205 test_->receive_config_.rtp.remote_ssrc =
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| call.cc | 342 RTC_DCHECK(audio_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
344 audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
360 audio_receive_stream->config().rtp.remote_ssrc);
445 RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
447 video_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
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| rampup_tests.cc | 159 recv_config.rtp.remote_ssrc = video_ssrcs_[i];
206 recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
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| rtc_event_log_unittest.cc | 131 EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
375 config->rtp.remote_ssrc = prng->Rand<uint32_t>();
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| /external/webrtc/webrtc/audio/ |
| audio_receive_stream.cc | 51 ss << "{remote_ssrc: " << remote_ssrc; local
138 remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
193 stats.remote_ssrc = config_.rtp.remote_ssrc;
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| audio_receive_stream_unittest.cc | 105 stream_config_.rtp.remote_ssrc = kRemoteSsrc;
127 RemoveStream(stream_config_.rtp.remote_ssrc));
206 config.rtp.remote_ssrc = kRemoteSsrc;
213 "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
292 EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
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| /external/webrtc/webrtc/voice_engine/test/auto_test/fakes/ |
| conference_transport.cc | 229 const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++; local
230 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
249 streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
251 return remote_ssrc; // remote ssrc used as stream id.
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| /external/webrtc/talk/media/webrtc/ |
| fakewebrtccall.cc | 289 if (p->GetConfig().rtp.remote_ssrc == ssrc) {
399 if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
406 if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
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| /external/webrtc/webrtc/modules/rtp_rtcp/include/ |
| rtp_rtcp_defines.h | 136 RTCPReportBlock(uint32_t remote_ssrc,
144 : remoteSSRC(remote_ssrc),
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| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_receiver.h | 264 uint32_t remote_ssrc, uint32_t source_ssrc)
267 uint32_t remote_ssrc, uint32_t source_ssrc) const
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| rtp_rtcp_impl.h | 140 int32_t RemoteCNAME(uint32_t remote_ssrc,
155 int32_t RTT(uint32_t remote_ssrc,
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| rtp_rtcp_impl.cc | 509 const uint32_t remote_ssrc,
511 return rtcp_receiver_.CNAME(remote_ssrc, c_name);
530 int32_t ModuleRtpRtcpImpl::RTT(const uint32_t remote_ssrc,
535 int32_t ret = rtcp_receiver_.RTT(remote_ssrc, rtt, avg_rtt, min_rtt, max_rtt);
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| rtcp_receiver.cc | 425 "remote_ssrc", remoteSSRC, "ssrc", main_ssrc_);
455 "remote_ssrc", remoteSSRC, "ssrc", main_ssrc_);
588 uint32_t remote_ssrc,
591 GetReportBlockInformation(remote_ssrc, source_ssrc);
594 _receivedReportBlockMap[source_ssrc][remote_ssrc] = info;
600 uint32_t remote_ssrc,
607 ReportBlockInfoMap::const_iterator it_info = info_map->find(remote_ssrc);
[all...] |
| /external/webrtc/webrtc/modules/bitrate_controller/ |
| bitrate_controller_unittest.cc | 32 uint32_t remote_ssrc, uint32_t source_ssrc,
34 return webrtc::RTCPReportBlock(remote_ssrc, source_ssrc, fraction_lost, 0,
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| /external/webrtc/webrtc/test/ |
| call_test.cc | 222 video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
234 audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
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| /external/webrtc/talk/app/webrtc/ |
| peerconnection.h | 301 uint32_t remote_ssrc);
|
