| OLD | NEW |
| 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" | |
| 16 #include "content/browser/renderer_host/media/audio_input_renderer_host.h" | 15 #include "content/browser/renderer_host/media/audio_input_renderer_host.h" |
| 17 #include "content/browser/renderer_host/media/audio_renderer_host.h" | 16 #include "content/browser/renderer_host/media/audio_renderer_host.h" |
| 18 #include "content/browser/renderer_host/media/media_stream_manager.h" | 17 #include "content/browser/renderer_host/media/media_stream_manager.h" |
| 19 #include "content/browser/renderer_host/media/mock_media_observer.h" | 18 #include "content/browser/renderer_host/media/mock_media_observer.h" |
| 20 #include "content/browser/renderer_host/media/video_capture_manager.h" | |
| 21 #include "content/common/view_messages.h" | 19 #include "content/common/view_messages.h" |
| 22 #include "content/public/browser/browser_thread.h" | 20 #include "content/public/browser/browser_thread.h" |
| 23 #include "content/public/common/content_paths.h" | 21 #include "content/public/common/content_paths.h" |
| 24 #include "content/public/test/mock_resource_context.h" | 22 #include "content/public/test/mock_resource_context.h" |
| 25 #include "content/public/test/test_browser_thread.h" | 23 #include "content/public/test/test_browser_thread.h" |
| 26 #include "content/renderer/media/audio_device_factory.h" | 24 #include "content/renderer/media/audio_device_factory.h" |
| 27 #include "content/renderer/media/audio_hardware.h" | 25 #include "content/renderer/media/audio_hardware.h" |
| 28 #include "content/renderer/media/audio_input_message_filter.h" | 26 #include "content/renderer/media/audio_input_message_filter.h" |
| 29 #include "content/renderer/media/audio_message_filter.h" | 27 #include "content/renderer/media/audio_message_filter.h" |
| 30 #include "content/renderer/media/webrtc_audio_device_impl.h" | 28 #include "content/renderer/media/webrtc_audio_device_impl.h" |
| (...skipping 99 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 130 // Main parts are inspired by the RenderViewFakeResourcesTest. | 128 // Main parts are inspired by the RenderViewFakeResourcesTest. |
| 131 // Note that, the IPC part is not utilized in this test. | 129 // Note that, the IPC part is not utilized in this test. |
| 132 saved_content_renderer_.reset( | 130 saved_content_renderer_.reset( |
| 133 new ReplaceContentClientRenderer(&content_renderer_client_)); | 131 new ReplaceContentClientRenderer(&content_renderer_client_)); |
| 134 mock_process_.reset(new WebRTCMockRenderProcess()); | 132 mock_process_.reset(new WebRTCMockRenderProcess()); |
| 135 ui_thread_.reset(new content::TestBrowserThread(content::BrowserThread::UI, | 133 ui_thread_.reset(new content::TestBrowserThread(content::BrowserThread::UI, |
| 136 MessageLoop::current())); | 134 MessageLoop::current())); |
| 137 | 135 |
| 138 // Create our own AudioManager and MediaStreamManager. | 136 // Create our own AudioManager and MediaStreamManager. |
| 139 audio_manager_.reset(media::AudioManager::Create()); | 137 audio_manager_.reset(media::AudioManager::Create()); |
| 140 | 138 media_stream_manager_.reset( |
| 141 scoped_refptr<media_stream::AudioInputDeviceManager> | 139 new media_stream::MediaStreamManager(audio_manager_.get())); |
| 142 audio_input_device_manager(new media_stream::AudioInputDeviceManager( | |
| 143 audio_manager_.get())); | |
| 144 scoped_refptr<media_stream::VideoCaptureManager> video_capture_manager( | |
| 145 new media_stream::VideoCaptureManager()); | |
| 146 media_stream_manager_.reset(new media_stream::MediaStreamManager( | |
| 147 audio_input_device_manager, video_capture_manager)); | |
| 148 | 140 |
| 149 // Construct the resource context on the UI thread. | 141 // Construct the resource context on the UI thread. |
| 150 resource_context_.reset(new MockResourceContext); | 142 resource_context_.reset(new MockResourceContext); |
| 151 | 143 |
| 152 static const char kThreadName[] = "RenderThread"; | 144 static const char kThreadName[] = "RenderThread"; |
| 153 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, | 145 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, |
| 154 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, | 146 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, |
| 155 base::Unretained(this), kThreadName)); | 147 base::Unretained(this), kThreadName)); |
| 156 WaitForIOThreadCompletion(); | 148 WaitForIOThreadCompletion(); |
| 157 | 149 |
| (...skipping 211 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
| 369 WebRTCTransportImpl::~WebRTCTransportImpl() {} | 361 WebRTCTransportImpl::~WebRTCTransportImpl() {} |
| 370 | 362 |
| 371 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 363 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
| 372 return network_->ReceivedRTPPacket(channel, data, len); | 364 return network_->ReceivedRTPPacket(channel, data, len); |
| 373 } | 365 } |
| 374 | 366 |
| 375 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 367 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
| 376 int len) { | 368 int len) { |
| 377 return network_->ReceivedRTCPPacket(channel, data, len); | 369 return network_->ReceivedRTCPPacket(channel, data, len); |
| 378 } | 370 } |
| OLD | NEW |