Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(51)

Side by Side Diff: content/test/webrtc_audio_device_test.cc

Issue 13288002: Revert 190475 "Merge 189983 "Add speaker on/off control on Andro..." (Closed) Base URL: svn://svn.chromium.org/chrome/branches/1410/src/
Patch Set: Created 7 years, 8 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch | Annotate | Revision Log
OLDNEW
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"
(...skipping 22 matching lines...) Expand all
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 {
(...skipping 64 matching lines...) Expand 10 before | Expand all | Expand 10 after
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
(...skipping 222 matching lines...) Expand 10 before | Expand all | Expand 10 after
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
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698