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| 1 /* | 1 /* | 
| 2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 2  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | 
| 3  * | 3  * | 
| 4  *  Use of this source code is governed by a BSD-style license | 4  *  Use of this source code is governed by a BSD-style license | 
| 5  *  that can be found in the LICENSE file in the root of the source | 5  *  that can be found in the LICENSE file in the root of the source | 
| 6  *  tree. An additional intellectual property rights grant can be found | 6  *  tree. An additional intellectual property rights grant can be found | 
| 7  *  in the file PATENTS.  All contributing project authors may | 7  *  in the file PATENTS.  All contributing project authors may | 
| 8  *  be found in the AUTHORS file in the root of the source tree. | 8  *  be found in the AUTHORS file in the root of the source tree. | 
| 9  */ | 9  */ | 
| 10 | 10 | 
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| 33     return ssrc; | 33     return ssrc; | 
| 34   } | 34   } | 
| 35 }  // namespace | 35 }  // namespace | 
| 36 | 36 | 
| 37 namespace voetest { | 37 namespace voetest { | 
| 38 | 38 | 
| 39 ConferenceTransport::ConferenceTransport() | 39 ConferenceTransport::ConferenceTransport() | 
| 40     : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 40     : pq_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 
| 41       stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 41       stream_crit_(webrtc::CriticalSectionWrapper::CreateCriticalSection()), | 
| 42       packet_event_(webrtc::EventWrapper::Create()), | 42       packet_event_(webrtc::EventWrapper::Create()), | 
| 43       thread_(webrtc::ThreadWrapper::CreateThread(Run, | 43       thread_(webrtc::PlatformThread::CreateThread(Run, | 
| 44                                                   this, | 44                                                    this, | 
| 45                                                   "ConferenceTransport")), | 45                                                    "ConferenceTransport")), | 
| 46       rtt_ms_(0), | 46       rtt_ms_(0), | 
| 47       stream_count_(0), | 47       stream_count_(0), | 
| 48       rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { | 48       rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { | 
| 49   rtp_header_parser_-> | 49   rtp_header_parser_-> | 
| 50       RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, | 50       RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, | 
| 51                                  kAudioLevelHeaderId); | 51                                  kAudioLevelHeaderId); | 
| 52 | 52 | 
| 53   local_voe_ = webrtc::VoiceEngine::Create(); | 53   local_voe_ = webrtc::VoiceEngine::Create(); | 
| 54   local_base_ = webrtc::VoEBase::GetInterface(local_voe_); | 54   local_base_ = webrtc::VoEBase::GetInterface(local_voe_); | 
| 55   local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); | 55   local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); | 
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| 281 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 281 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 
| 282                                                 webrtc::CallStatistics* stats) { | 282                                                 webrtc::CallStatistics* stats) { | 
| 283   int dst = GetReceiverChannelForSsrc(id); | 283   int dst = GetReceiverChannelForSsrc(id); | 
| 284   if (dst == -1) { | 284   if (dst == -1) { | 
| 285     return false; | 285     return false; | 
| 286   } | 286   } | 
| 287   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 287   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 
| 288   return true; | 288   return true; | 
| 289 } | 289 } | 
| 290 }  // namespace voetest | 290 }  // namespace voetest | 
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