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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::PlatformThread::CreateThread(Run, | 43 thread_(Run, this, "ConferenceTransport"), |
| 44 this, | |
| 45 "ConferenceTransport")), | |
| 46 rtt_ms_(0), | 44 rtt_ms_(0), |
| 47 stream_count_(0), | 45 stream_count_(0), |
| 48 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { | 46 rtp_header_parser_(webrtc::RtpHeaderParser::Create()) { |
| 49 rtp_header_parser_-> | 47 rtp_header_parser_-> |
| 50 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, | 48 RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel, |
| 51 kAudioLevelHeaderId); | 49 kAudioLevelHeaderId); |
| 52 | 50 |
| 53 local_voe_ = webrtc::VoiceEngine::Create(); | 51 local_voe_ = webrtc::VoiceEngine::Create(); |
| 54 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); | 52 local_base_ = webrtc::VoEBase::GetInterface(local_voe_); |
| 55 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); | 53 local_network_ = webrtc::VoENetwork::GetInterface(local_voe_); |
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| 72 SetSendAudioLevelIndicationStatus(local_sender_, true, | 70 SetSendAudioLevelIndicationStatus(local_sender_, true, |
| 73 kAudioLevelHeaderId)); | 71 kAudioLevelHeaderId)); |
| 74 | 72 |
| 75 EXPECT_EQ(0, local_base_->StartSend(local_sender_)); | 73 EXPECT_EQ(0, local_base_->StartSend(local_sender_)); |
| 76 | 74 |
| 77 EXPECT_EQ(0, remote_base_->Init()); | 75 EXPECT_EQ(0, remote_base_->Init()); |
| 78 reflector_ = remote_base_->CreateChannel(); | 76 reflector_ = remote_base_->CreateChannel(); |
| 79 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this)); | 77 EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this)); |
| 80 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc)); | 78 EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc)); |
| 81 | 79 |
| 82 thread_->Start(); | 80 thread_.Start(); |
| 83 thread_->SetPriority(webrtc::kHighPriority); | 81 thread_.SetPriority(rtc::kHighPriority); |
| 84 } | 82 } |
| 85 | 83 |
| 86 ConferenceTransport::~ConferenceTransport() { | 84 ConferenceTransport::~ConferenceTransport() { |
| 87 // Must stop sending, otherwise DispatchPackets() cannot quit. | 85 // Must stop sending, otherwise DispatchPackets() cannot quit. |
| 88 EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_)); | 86 EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_)); |
| 89 EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_)); | 87 EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_)); |
| 90 | 88 |
| 91 while (!streams_.empty()) { | 89 while (!streams_.empty()) { |
| 92 auto stream = streams_.begin(); | 90 auto stream = streams_.begin(); |
| 93 RemoveStream(stream->first); | 91 RemoveStream(stream->first); |
| 94 } | 92 } |
| 95 | 93 |
| 96 EXPECT_TRUE(thread_->Stop()); | 94 thread_.Stop(); |
| 97 | 95 |
| 98 remote_file_->Release(); | 96 remote_file_->Release(); |
| 99 remote_rtp_rtcp_->Release(); | 97 remote_rtp_rtcp_->Release(); |
| 100 remote_network_->Release(); | 98 remote_network_->Release(); |
| 101 remote_base_->Release(); | 99 remote_base_->Release(); |
| 102 | 100 |
| 103 local_rtp_rtcp_->Release(); | 101 local_rtp_rtcp_->Release(); |
| 104 local_network_->Release(); | 102 local_network_->Release(); |
| 105 local_base_->Release(); | 103 local_base_->Release(); |
| 106 | 104 |
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| 281 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, | 279 bool ConferenceTransport::GetReceiverStatistics(unsigned int id, |
| 282 webrtc::CallStatistics* stats) { | 280 webrtc::CallStatistics* stats) { |
| 283 int dst = GetReceiverChannelForSsrc(id); | 281 int dst = GetReceiverChannelForSsrc(id); |
| 284 if (dst == -1) { | 282 if (dst == -1) { |
| 285 return false; | 283 return false; |
| 286 } | 284 } |
| 287 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); | 285 EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats)); |
| 288 return true; | 286 return true; |
| 289 } | 287 } |
| 290 } // namespace voetest | 288 } // namespace voetest |
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