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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" |
11 #include "base/message_loop.h" | 11 #include "base/message_loop.h" |
12 #include "base/synchronization/waitable_event.h" | 12 #include "base/synchronization/waitable_event.h" |
13 #include "base/test/test_timeouts.h" | 13 #include "base/test/test_timeouts.h" |
14 #include "base/win/scoped_com_initializer.h" | 14 #include "base/win/scoped_com_initializer.h" |
| 15 #include "content/browser/renderer_host/media/audio_input_device_manager.h" |
15 #include "content/browser/renderer_host/media/audio_input_renderer_host.h" | 16 #include "content/browser/renderer_host/media/audio_input_renderer_host.h" |
16 #include "content/browser/renderer_host/media/audio_renderer_host.h" | 17 #include "content/browser/renderer_host/media/audio_renderer_host.h" |
17 #include "content/browser/renderer_host/media/media_stream_manager.h" | 18 #include "content/browser/renderer_host/media/media_stream_manager.h" |
18 #include "content/browser/renderer_host/media/mock_media_observer.h" | 19 #include "content/browser/renderer_host/media/mock_media_observer.h" |
| 20 #include "content/browser/renderer_host/media/video_capture_manager.h" |
19 #include "content/common/view_messages.h" | 21 #include "content/common/view_messages.h" |
20 #include "content/public/browser/browser_thread.h" | 22 #include "content/public/browser/browser_thread.h" |
21 #include "content/public/common/content_paths.h" | 23 #include "content/public/common/content_paths.h" |
22 #include "content/public/test/mock_resource_context.h" | 24 #include "content/public/test/mock_resource_context.h" |
23 #include "content/public/test/test_browser_thread.h" | 25 #include "content/public/test/test_browser_thread.h" |
24 #include "content/renderer/media/audio_device_factory.h" | 26 #include "content/renderer/media/audio_device_factory.h" |
25 #include "content/renderer/media/audio_hardware.h" | 27 #include "content/renderer/media/audio_hardware.h" |
26 #include "content/renderer/media/audio_message_filter.h" | 28 #include "content/renderer/media/audio_message_filter.h" |
27 #include "content/renderer/media/webrtc_audio_device_impl.h" | 29 #include "content/renderer/media/webrtc_audio_device_impl.h" |
28 #include "content/renderer/render_process.h" | 30 #include "content/renderer/render_process.h" |
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43 using testing::StrEq; | 45 using testing::StrEq; |
44 | 46 |
45 // This class is a mock of the child process singleton which is needed | 47 // This class is a mock of the child process singleton which is needed |
46 // to be able to create a RenderThread object. | 48 // to be able to create a RenderThread object. |
47 class WebRTCMockRenderProcess : public RenderProcess { | 49 class WebRTCMockRenderProcess : public RenderProcess { |
48 public: | 50 public: |
49 WebRTCMockRenderProcess() {} | 51 WebRTCMockRenderProcess() {} |
50 virtual ~WebRTCMockRenderProcess() {} | 52 virtual ~WebRTCMockRenderProcess() {} |
51 | 53 |
52 // RenderProcess implementation. | 54 // RenderProcess implementation. |
53 virtual skia::PlatformCanvas* GetDrawingCanvas(TransportDIB** memory, | 55 virtual skia::PlatformCanvas* GetDrawingCanvas( |
54 const gfx::Rect& rect) { | 56 TransportDIB** memory, const gfx::Rect& rect) OVERRIDE { |
55 return NULL; | 57 return NULL; |
56 } | 58 } |
57 virtual void ReleaseTransportDIB(TransportDIB* memory) {} | 59 virtual void ReleaseTransportDIB(TransportDIB* memory) OVERRIDE {} |
58 virtual bool UseInProcessPlugins() const { return false; } | 60 virtual bool UseInProcessPlugins() const OVERRIDE { return false; } |
59 virtual void AddBindings(int bindings) {} | 61 virtual void AddBindings(int bindings) OVERRIDE {} |
60 virtual int GetEnabledBindings() const { return 0; } | 62 virtual int GetEnabledBindings() const { return 0; } |
61 virtual bool HasInitializedMediaLibrary() const { return false; } | 63 virtual bool HasInitializedMediaLibrary() const { return false; } |
62 | 64 |
63 private: | 65 private: |
64 DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess); | 66 DISALLOW_COPY_AND_ASSIGN(WebRTCMockRenderProcess); |
65 }; | 67 }; |
66 | 68 |
67 // Utility scoped class to replace the global content client's renderer for the | 69 // Utility scoped class to replace the global content client's renderer for the |
68 // duration of the test. | 70 // duration of the test. |
69 class ReplaceContentClientRenderer { | 71 class ReplaceContentClientRenderer { |
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190 // We initialize COM (STA) on our IO thread as is done in Chrome. | 192 // We initialize COM (STA) on our IO thread as is done in Chrome. |
191 // See BrowserProcessSubThread::Init. | 193 // See BrowserProcessSubThread::Init. |
192 initialize_com_.reset(new ScopedCOMInitializer()); | 194 initialize_com_.reset(new ScopedCOMInitializer()); |
193 | 195 |
194 // Set the current thread as the IO thread. | 196 // Set the current thread as the IO thread. |
195 io_thread_.reset(new content::TestBrowserThread(content::BrowserThread::IO, | 197 io_thread_.reset(new content::TestBrowserThread(content::BrowserThread::IO, |
196 MessageLoop::current())); | 198 MessageLoop::current())); |
197 | 199 |
198 audio_manager_.reset(media::AudioManager::Create()); | 200 audio_manager_.reset(media::AudioManager::Create()); |
199 | 201 |
| 202 scoped_refptr<media_stream::AudioInputDeviceManager> |
| 203 audio_input_device_manager(new media_stream::AudioInputDeviceManager( |
| 204 audio_manager_.get())); |
| 205 scoped_refptr<media_stream::VideoCaptureManager> video_capture_manager( |
| 206 new media_stream::VideoCaptureManager()); |
| 207 media_stream_manager_.reset(new media_stream::MediaStreamManager( |
| 208 audio_input_device_manager, video_capture_manager)); |
| 209 |
200 // Populate our resource context. | 210 // Populate our resource context. |
201 test_request_context_.reset(new TestURLRequestContext()); | 211 test_request_context_.reset(new TestURLRequestContext()); |
202 MockResourceContext* resource_context = | 212 MockResourceContext* resource_context = |
203 static_cast<MockResourceContext*>(resource_context_.get()); | 213 static_cast<MockResourceContext*>(resource_context_.get()); |
204 resource_context->set_request_context(test_request_context_.get()); | 214 resource_context->set_request_context(test_request_context_.get()); |
205 media_observer_.reset(new MockMediaObserver()); | 215 media_observer_.reset(new MockMediaObserver()); |
206 | 216 |
207 has_input_devices_ = audio_manager_->HasAudioInputDevices(); | 217 has_input_devices_ = audio_manager_->HasAudioInputDevices(); |
208 has_output_devices_ = audio_manager_->HasAudioOutputDevices(); | 218 has_output_devices_ = audio_manager_->HasAudioOutputDevices(); |
209 | 219 |
210 // Create an IPC channel that handles incoming messages on the IO thread. | 220 // Create an IPC channel that handles incoming messages on the IO thread. |
211 CreateChannel(thread_name); | 221 CreateChannel(thread_name); |
212 } | 222 } |
213 | 223 |
214 void WebRTCAudioDeviceTest::UninitializeIOThread() { | 224 void WebRTCAudioDeviceTest::UninitializeIOThread() { |
215 resource_context_.reset(); | 225 resource_context_.reset(); |
216 | 226 |
| 227 media_stream_manager_.reset(); |
217 audio_manager_.reset(); | 228 audio_manager_.reset(); |
218 test_request_context_.reset(); | 229 test_request_context_.reset(); |
219 initialize_com_.reset(); | 230 initialize_com_.reset(); |
220 } | 231 } |
221 | 232 |
222 void WebRTCAudioDeviceTest::CreateChannel(const char* name) { | 233 void WebRTCAudioDeviceTest::CreateChannel(const char* name) { |
223 DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO)); | 234 DCHECK(content::BrowserThread::CurrentlyOn(content::BrowserThread::IO)); |
224 audio_render_host_ = new AudioRendererHost( | 235 audio_render_host_ = new AudioRendererHost( |
225 audio_manager_.get(), media_observer_.get()); | 236 audio_manager_.get(), media_observer_.get()); |
226 audio_render_host_->OnChannelConnected(base::GetCurrentProcId()); | 237 audio_render_host_->OnChannelConnected(base::GetCurrentProcId()); |
227 | 238 |
228 audio_input_renderer_host_ = new AudioInputRendererHost( | 239 audio_input_renderer_host_ = new AudioInputRendererHost( |
229 resource_context_.get(), audio_manager_.get()); | 240 audio_manager_.get(), media_stream_manager_.get()); |
230 audio_input_renderer_host_->OnChannelConnected(base::GetCurrentProcId()); | 241 audio_input_renderer_host_->OnChannelConnected(base::GetCurrentProcId()); |
231 | 242 |
232 channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this)); | 243 channel_.reset(new IPC::Channel(name, IPC::Channel::MODE_SERVER, this)); |
233 ASSERT_TRUE(channel_->Connect()); | 244 ASSERT_TRUE(channel_->Connect()); |
234 | 245 |
235 audio_render_host_->OnFilterAdded(channel_.get()); | 246 audio_render_host_->OnFilterAdded(channel_.get()); |
236 audio_input_renderer_host_->OnFilterAdded(channel_.get()); | 247 audio_input_renderer_host_->OnFilterAdded(channel_.get()); |
237 } | 248 } |
238 | 249 |
239 void WebRTCAudioDeviceTest::DestroyChannel() { | 250 void WebRTCAudioDeviceTest::DestroyChannel() { |
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356 WebRTCTransportImpl::~WebRTCTransportImpl() {} | 367 WebRTCTransportImpl::~WebRTCTransportImpl() {} |
357 | 368 |
358 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 369 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
359 return network_->ReceivedRTPPacket(channel, data, len); | 370 return network_->ReceivedRTPPacket(channel, data, len); |
360 } | 371 } |
361 | 372 |
362 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 373 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
363 int len) { | 374 int len) { |
364 return network_->ReceivedRTCPPacket(channel, data, len); | 375 return network_->ReceivedRTCPPacket(channel, data, len); |
365 } | 376 } |
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