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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. | 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "content/test/webrtc_audio_device_test.h" | 5 #include "content/test/webrtc_audio_device_test.h" |
6 | 6 |
7 #include "base/bind.h" | 7 #include "base/bind.h" |
8 #include "base/bind_helpers.h" | 8 #include "base/bind_helpers.h" |
9 #include "base/compiler_specific.h" | 9 #include "base/compiler_specific.h" |
10 #include "base/file_util.h" | 10 #include "base/file_util.h" |
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110 return test_request_context_; | 110 return test_request_context_; |
111 } | 111 } |
112 | 112 |
113 private: | 113 private: |
114 net::URLRequestContext* test_request_context_; | 114 net::URLRequestContext* test_request_context_; |
115 | 115 |
116 DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext); | 116 DISALLOW_COPY_AND_ASSIGN(MockRTCResourceContext); |
117 }; | 117 }; |
118 | 118 |
119 ACTION_P(QuitMessageLoop, loop_or_proxy) { | 119 ACTION_P(QuitMessageLoop, loop_or_proxy) { |
120 loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure()); | 120 loop_or_proxy->PostTask(FROM_HERE, base::MessageLoop::QuitClosure()); |
121 } | 121 } |
122 | 122 |
123 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest() | 123 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest() |
124 : render_thread_(NULL), audio_hardware_config_(NULL), | 124 : render_thread_(NULL), audio_hardware_config_(NULL), |
125 has_input_devices_(false), has_output_devices_(false) { | 125 has_input_devices_(false), has_output_devices_(false) { |
126 } | 126 } |
127 | 127 |
128 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {} | 128 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {} |
129 | 129 |
130 void WebRTCAudioDeviceTest::SetUp() { | 130 void WebRTCAudioDeviceTest::SetUp() { |
131 // This part sets up a RenderThread environment to ensure that | 131 // This part sets up a RenderThread environment to ensure that |
132 // RenderThread::current() (<=> TLS pointer) is valid. | 132 // RenderThread::current() (<=> TLS pointer) is valid. |
133 // Main parts are inspired by the RenderViewFakeResourcesTest. | 133 // Main parts are inspired by the RenderViewFakeResourcesTest. |
134 // Note that, the IPC part is not utilized in this test. | 134 // Note that, the IPC part is not utilized in this test. |
135 saved_content_renderer_.reset( | 135 saved_content_renderer_.reset( |
136 new ReplaceContentClientRenderer(&content_renderer_client_)); | 136 new ReplaceContentClientRenderer(&content_renderer_client_)); |
137 mock_process_.reset(new WebRTCMockRenderProcess()); | 137 mock_process_.reset(new WebRTCMockRenderProcess()); |
138 ui_thread_.reset(new TestBrowserThread(BrowserThread::UI, | 138 ui_thread_.reset( |
139 MessageLoop::current())); | 139 new TestBrowserThread(BrowserThread::UI, base::MessageLoop::current())); |
140 | 140 |
141 // Construct the resource context on the UI thread. | 141 // Construct the resource context on the UI thread. |
142 resource_context_.reset(new MockRTCResourceContext); | 142 resource_context_.reset(new MockRTCResourceContext); |
143 | 143 |
144 static const char kThreadName[] = "RenderThread"; | 144 static const char kThreadName[] = "RenderThread"; |
145 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, | 145 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, |
146 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, | 146 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, |
147 base::Unretained(this), kThreadName)); | 147 base::Unretained(this), kThreadName)); |
148 WaitForIOThreadCompletion(); | 148 WaitForIOThreadCompletion(); |
149 | 149 |
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195 } | 195 } |
196 | 196 |
197 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) { | 197 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) { |
198 #if defined(OS_WIN) | 198 #if defined(OS_WIN) |
199 // We initialize COM (STA) on our IO thread as is done in Chrome. | 199 // We initialize COM (STA) on our IO thread as is done in Chrome. |
200 // See BrowserProcessSubThread::Init. | 200 // See BrowserProcessSubThread::Init. |
201 initialize_com_.reset(new base::win::ScopedCOMInitializer()); | 201 initialize_com_.reset(new base::win::ScopedCOMInitializer()); |
202 #endif | 202 #endif |
203 | 203 |
204 // Set the current thread as the IO thread. | 204 // Set the current thread as the IO thread. |
205 io_thread_.reset(new TestBrowserThread(BrowserThread::IO, | 205 io_thread_.reset( |
206 MessageLoop::current())); | 206 new TestBrowserThread(BrowserThread::IO, base::MessageLoop::current())); |
207 | 207 |
208 // Populate our resource context. | 208 // Populate our resource context. |
209 test_request_context_.reset(new net::TestURLRequestContext()); | 209 test_request_context_.reset(new net::TestURLRequestContext()); |
210 MockRTCResourceContext* resource_context = | 210 MockRTCResourceContext* resource_context = |
211 static_cast<MockRTCResourceContext*>(resource_context_.get()); | 211 static_cast<MockRTCResourceContext*>(resource_context_.get()); |
212 resource_context->set_request_context(test_request_context_.get()); | 212 resource_context->set_request_context(test_request_context_.get()); |
213 media_internals_.reset(new MockMediaInternals()); | 213 media_internals_.reset(new MockMediaInternals()); |
214 | 214 |
215 // Create our own AudioManager, AudioMirroringManager and MediaStreamManager. | 215 // Create our own AudioManager, AudioMirroringManager and MediaStreamManager. |
216 audio_manager_.reset(media::AudioManager::Create()); | 216 audio_manager_.reset(media::AudioManager::Create()); |
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357 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 357 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
358 return network_->ReceivedRTPPacket(channel, data, len); | 358 return network_->ReceivedRTPPacket(channel, data, len); |
359 } | 359 } |
360 | 360 |
361 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 361 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
362 int len) { | 362 int len) { |
363 return network_->ReceivedRTCPPacket(channel, data, len); | 363 return network_->ReceivedRTCPPacket(channel, data, len); |
364 } | 364 } |
365 | 365 |
366 } // namespace content | 366 } // namespace content |
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