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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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