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Side by Side Diff: webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

Issue 1476453002: Clean up PlatformThread. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: IsRunning DCHECK Created 5 years ago
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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_);
(...skipping 16 matching lines...) Expand all
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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