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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_renderer_host.h" | 15 #include "content/browser/renderer_host/media/audio_input_renderer_host.h" |
16 #include "content/browser/renderer_host/media/audio_renderer_host.h" | 16 #include "content/browser/renderer_host/media/audio_renderer_host.h" |
17 #include "content/browser/renderer_host/media/media_stream_manager.h" | 17 #include "content/browser/renderer_host/media/media_stream_manager.h" |
18 #include "content/browser/renderer_host/media/mock_media_observer.h" | 18 #include "content/browser/renderer_host/media/mock_media_observer.h" |
19 #include "content/common/view_messages.h" | 19 #include "content/common/view_messages.h" |
20 #include "content/public/browser/browser_thread.h" | 20 #include "content/public/browser/browser_thread.h" |
21 #include "content/public/common/content_paths.h" | 21 #include "content/public/common/content_paths.h" |
22 #include "content/public/test/mock_resource_context.h" | 22 #include "content/public/test/mock_resource_context.h" |
23 #include "content/public/test/test_browser_thread.h" | 23 #include "content/public/test/test_browser_thread.h" |
| 24 #include "content/renderer/media/audio_device.h" |
| 25 #include "content/renderer/media/audio_device_factory.h" |
24 #include "content/renderer/media/audio_hardware.h" | 26 #include "content/renderer/media/audio_hardware.h" |
| 27 #include "content/renderer/media/audio_message_filter.h" |
25 #include "content/renderer/media/webrtc_audio_device_impl.h" | 28 #include "content/renderer/media/webrtc_audio_device_impl.h" |
26 #include "content/renderer/render_process.h" | 29 #include "content/renderer/render_process.h" |
27 #include "content/renderer/render_thread_impl.h" | 30 #include "content/renderer/render_thread_impl.h" |
28 #include "content/renderer/renderer_webkitplatformsupport_impl.h" | 31 #include "content/renderer/renderer_webkitplatformsupport_impl.h" |
29 #include "net/url_request/url_request_test_util.h" | 32 #include "net/url_request/url_request_test_util.h" |
30 #include "testing/gmock/include/gmock/gmock.h" | 33 #include "testing/gmock/include/gmock/gmock.h" |
31 #include "testing/gtest/include/gtest/gtest.h" | 34 #include "testing/gtest/include/gtest/gtest.h" |
32 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h" | 35 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h" |
33 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h" | 36 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h" |
34 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h" | 37 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h" |
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134 | 137 |
135 static const char kThreadName[] = "RenderThread"; | 138 static const char kThreadName[] = "RenderThread"; |
136 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, | 139 ChildProcess::current()->io_message_loop()->PostTask(FROM_HERE, |
137 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, | 140 base::Bind(&WebRTCAudioDeviceTest::InitializeIOThread, |
138 base::Unretained(this), kThreadName)); | 141 base::Unretained(this), kThreadName)); |
139 WaitForIOThreadCompletion(); | 142 WaitForIOThreadCompletion(); |
140 | 143 |
141 sandbox_was_enabled_ = | 144 sandbox_was_enabled_ = |
142 RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(false); | 145 RendererWebKitPlatformSupportImpl::SetSandboxEnabledForTesting(false); |
143 render_thread_ = new RenderThreadImpl(kThreadName); | 146 render_thread_ = new RenderThreadImpl(kThreadName); |
| 147 |
| 148 audio_device_factory_.reset( |
| 149 new AudioDeviceFactory<AudioDevice, AudioMessageFilter>( |
| 150 render_thread_->audio_message_filter())); |
144 } | 151 } |
145 | 152 |
146 void WebRTCAudioDeviceTest::TearDown() { | 153 void WebRTCAudioDeviceTest::TearDown() { |
147 SetAudioUtilCallback(NULL); | 154 SetAudioUtilCallback(NULL); |
148 | 155 |
149 // Run any pending cleanup tasks that may have been posted to the main thread. | 156 // Run any pending cleanup tasks that may have been posted to the main thread. |
150 ChildProcess::current()->main_thread()->message_loop()->RunAllPending(); | 157 ChildProcess::current()->main_thread()->message_loop()->RunAllPending(); |
151 | 158 |
152 // Kick of the cleanup process by closing the channel. This queues up | 159 // Kick of the cleanup process by closing the channel. This queues up |
153 // OnStreamClosed calls to be executed on the audio thread. | 160 // OnStreamClosed calls to be executed on the audio thread. |
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354 WebRTCTransportImpl::~WebRTCTransportImpl() {} | 361 WebRTCTransportImpl::~WebRTCTransportImpl() {} |
355 | 362 |
356 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 363 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
357 return network_->ReceivedRTPPacket(channel, data, len); | 364 return network_->ReceivedRTPPacket(channel, data, len); |
358 } | 365 } |
359 | 366 |
360 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 367 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
361 int len) { | 368 int len) { |
362 return network_->ReceivedRTCPPacket(channel, data, len); | 369 return network_->ReceivedRTCPPacket(channel, data, len); |
363 } | 370 } |
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