Add speaker on/off control on Android for WebRTC

content/test/webrtc_audio_device_test.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/test/webrtc_audio_device_test.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/file_util.h"
@@ skipping 24 matching lines @@
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/include/voe_base.h"
 #include "third_party/webrtc/voice_engine/include/voe_file.h"
 #include "third_party/webrtc/voice_engine/include/voe_network.h"
 
 #if defined(OS_WIN)
 #include "base/win/scoped_com_initializer.h"
 #endif
 
+#if defined(OS_ANDROID)
+#include "base/android/jni_android.h"
+#include "media/audio/audio_manager_base.h"
+#endif
+
 using media::AudioParameters;
 using media::ChannelLayout;
 using testing::_;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace content {
 
 // This class is a mock of the child process singleton which is needed
@@ skipping 66 matching lines @@
 }
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
     : render_thread_(NULL), audio_hardware_config_(NULL),
       has_input_devices_(false), has_output_devices_(false) {
 }
 
 WebRTCAudioDeviceTest::~WebRTCAudioDeviceTest() {}
 
 void WebRTCAudioDeviceTest::SetUp() {
+#if defined(OS_ANDROID)
+    media::AudioManagerBase::RegisterAudioManager(
+        base::android::AttachCurrentThread());
+#endif
+
   // This part sets up a RenderThread environment to ensure that
   // RenderThread::current() (<=> TLS pointer) is valid.
   // Main parts are inspired by the RenderViewFakeResourcesTest.
   // Note that, the IPC part is not utilized in this test.
   saved_content_renderer_.reset(
       new ReplaceContentClientRenderer(&content_renderer_client_));
   mock_process_.reset(new WebRTCMockRenderProcess());
   ui_thread_.reset(new TestBrowserThread(BrowserThread::UI,
                                          MessageLoop::current()));
 
@@ skipping 216 matching lines @@
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }
 
 }  // namespace content