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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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33 #include "testing/gtest/include/gtest/gtest.h" | 33 #include "testing/gtest/include/gtest/gtest.h" |
34 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h" | 34 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h" |
35 #include "third_party/webrtc/voice_engine/include/voe_base.h" | 35 #include "third_party/webrtc/voice_engine/include/voe_base.h" |
36 #include "third_party/webrtc/voice_engine/include/voe_file.h" | 36 #include "third_party/webrtc/voice_engine/include/voe_file.h" |
37 #include "third_party/webrtc/voice_engine/include/voe_network.h" | 37 #include "third_party/webrtc/voice_engine/include/voe_network.h" |
38 | 38 |
39 #if defined(OS_WIN) | 39 #if defined(OS_WIN) |
40 #include "base/win/scoped_com_initializer.h" | 40 #include "base/win/scoped_com_initializer.h" |
41 #endif | 41 #endif |
42 | 42 |
43 #if defined(OS_ANDROID) | |
44 #include "base/android/jni_android.h" | |
45 #include "media/audio/audio_manager_base.h" | |
46 #endif | |
47 | |
48 using testing::_; | 43 using testing::_; |
49 using testing::InvokeWithoutArgs; | 44 using testing::InvokeWithoutArgs; |
50 using testing::Return; | 45 using testing::Return; |
51 using testing::StrEq; | 46 using testing::StrEq; |
52 | 47 |
53 namespace content { | 48 namespace content { |
54 | 49 |
55 // This class is a mock of the child process singleton which is needed | 50 // This class is a mock of the child process singleton which is needed |
56 // to be able to create a RenderThread object. | 51 // to be able to create a RenderThread object. |
57 class WebRTCMockRenderProcess : public RenderProcess { | 52 class WebRTCMockRenderProcess : public RenderProcess { |
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122 } | 117 } |
123 | 118 |
124 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest() | 119 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest() |
125 : render_thread_(NULL), audio_hardware_config_(NULL), | 120 : render_thread_(NULL), audio_hardware_config_(NULL), |
126 has_input_devices_(false), has_output_devices_(false) { | 121 has_input_devices_(false), has_output_devices_(false) { |
127 } | 122 } |
128 | 123 |
129 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {} | 124 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {} |
130 | 125 |
131 void WebRTCAudioDeviceTest::SetUp() { | 126 void WebRTCAudioDeviceTest::SetUp() { |
132 #if defined(OS_ANDROID) | |
133 media::AudioManagerBase::RegisterAudioManager( | |
134 base::android::AttachCurrentThread()); | |
135 #endif | |
136 | |
137 // This part sets up a RenderThread environment to ensure that | 127 // This part sets up a RenderThread environment to ensure that |
138 // RenderThread::current() (<=> TLS pointer) is valid. | 128 // RenderThread::current() (<=> TLS pointer) is valid. |
139 // Main parts are inspired by the RenderViewFakeResourcesTest. | 129 // Main parts are inspired by the RenderViewFakeResourcesTest. |
140 // Note that, the IPC part is not utilized in this test. | 130 // Note that, the IPC part is not utilized in this test. |
141 saved_content_renderer_.reset( | 131 saved_content_renderer_.reset( |
142 new ReplaceContentClientRenderer(&content_renderer_client_)); | 132 new ReplaceContentClientRenderer(&content_renderer_client_)); |
143 mock_process_.reset(new WebRTCMockRenderProcess()); | 133 mock_process_.reset(new WebRTCMockRenderProcess()); |
144 ui_thread_.reset(new TestBrowserThread(BrowserThread::UI, | 134 ui_thread_.reset(new TestBrowserThread(BrowserThread::UI, |
145 MessageLoop::current())); | 135 MessageLoop::current())); |
146 | 136 |
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369 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { | 359 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) { |
370 return network_->ReceivedRTPPacket(channel, data, len); | 360 return network_->ReceivedRTPPacket(channel, data, len); |
371 } | 361 } |
372 | 362 |
373 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, | 363 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data, |
374 int len) { | 364 int len) { |
375 return network_->ReceivedRTCPPacket(channel, data, len); | 365 return network_->ReceivedRTCPPacket(channel, data, len); |
376 } | 366 } |
377 | 367 |
378 } // namespace content | 368 } // namespace content |
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