From d13842a16ccaa2c2cd03c200249779e6549398d6 Mon Sep 17 00:00:00 2001
From: "henrika@chromium.org"
 <henrika@chromium.org@0039d316-1c4b-4281-b951-d872f2087c98>
Date: Fri, 29 Jun 2012 08:59:04 +0000
Subject: [PATCH] This CL adds a new factory method called AudioDeviceFactory.
 It is a template based factory which allows the user to create a
 media::AudioRenderSink implementation (AudioDevice) using a factory which is
 created once and then distributed to all clients.

Advantages:

+ We can mock (or try alternative implementations of) AudioDevice for all clients at one centralized place.
+ The factory now owns the audio message filter, hence the restriction that each client must be created on the main renderer thread should now be removed.

BUG=none
TEST=content_unittests, misc. WebRTC demos, WebAudio demos and <audio> tag demos.

Review URL: https://chromiumcodereview.appspot.com/10537121

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@144888 0039d316-1c4b-4281-b951-d872f2087c98
---
 content/content_renderer.gypi                      |  2 +
 content/renderer/media/audio_device.cc             | 17 ++------
 content/renderer/media/audio_device.h              |  5 +--
 content/renderer/media/audio_device_factory.cc     | 32 +++++++++++++++
 content/renderer/media/audio_device_factory.h      | 47 ++++++++++++++++++++++
 content/renderer/media/audio_message_filter.cc     | 12 ++++++
 content/renderer/media/audio_message_filter.h      |  6 +++
 .../renderer/media/render_audiosourceprovider.cc   | 13 +++---
 .../renderer/media/render_audiosourceprovider.h    |  2 +-
 .../renderer/media/renderer_webaudiodevice_impl.cc |  7 +++-
 .../renderer/media/renderer_webaudiodevice_impl.h  | 10 ++---
 content/renderer/media/webrtc_audio_device_impl.cc | 24 ++++++-----
 content/renderer/media/webrtc_audio_device_impl.h  |  6 +--
 .../renderer/media/webrtc_audio_device_unittest.cc | 38 ++++++++---------
 content/renderer/render_view_impl.cc               |  1 -
 content/test/webrtc_audio_device_test.cc           |  2 +
 16 files changed, 160 insertions(+), 64 deletions(-)
 create mode 100644 content/renderer/media/audio_device_factory.cc
 create mode 100644 content/renderer/media/audio_device_factory.h

diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi
index d41b2a5e54c4..a0081a6a2a68 100644
--- a/content/content_renderer.gypi
+++ b/content/content_renderer.gypi
@@ -92,6 +92,8 @@
     'renderer/load_progress_tracker.h',
     'renderer/media/audio_device.cc',
     'renderer/media/audio_device.h',
+    'renderer/media/audio_device_factory.cc',
+    'renderer/media/audio_device_factory.h',
     'renderer/media/audio_device_thread.cc',
     'renderer/media/audio_device_thread.h',
     'renderer/media/audio_hardware.cc',
diff --git a/content/renderer/media/audio_device.cc b/content/renderer/media/audio_device.cc
index b87f30041662..7365bde690a8 100644
--- a/content/renderer/media/audio_device.cc
+++ b/content/renderer/media/audio_device.cc
@@ -11,7 +11,6 @@
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
 #include "content/common/view_messages.h"
-#include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_output_controller.h"
 #include "media/audio/audio_util.h"
 
@@ -46,19 +45,9 @@ AudioDevice::AudioDevice()
       stream_id_(0),
       play_on_start_(true),
       is_started_(false) {
-  filter_ = RenderThreadImpl::current()->audio_message_filter();
-}
-
-AudioDevice::AudioDevice(const media::AudioParameters& params,
-                         RenderCallback* callback)
-    : ScopedLoopObserver(ChildProcess::current()->io_message_loop()),
-      audio_parameters_(params),
-      callback_(callback),
-      volume_(1.0),
-      stream_id_(0),
-      play_on_start_(true),
-      is_started_(false) {
-  filter_ = RenderThreadImpl::current()->audio_message_filter();
+  // Use the filter instance already created on the main render thread.
+  CHECK(AudioMessageFilter::Get()) << "Invalid audio message filter.";
+  filter_ = AudioMessageFilter::Get();
 }
 
 void AudioDevice::Initialize(const media::AudioParameters& params,
diff --git a/content/renderer/media/audio_device.h b/content/renderer/media/audio_device.h
index 74b28d33d855..adce14333cdd 100644
--- a/content/renderer/media/audio_device.h
+++ b/content/renderer/media/audio_device.h
@@ -87,11 +87,10 @@ class CONTENT_EXPORT AudioDevice
  public:
   // Methods called on main render thread -------------------------------------
 
-  // Minimal constructor where Initialize() must be called later.
+  // Creates an uninitialized AudioDevice. Clients must call Initialize()
+  // before using.
   AudioDevice();
 
-  AudioDevice(const media::AudioParameters& params, RenderCallback* callback);
-
   // AudioRendererSink implementation.
   virtual void Initialize(const media::AudioParameters& params,
                           RenderCallback* callback) OVERRIDE;
diff --git a/content/renderer/media/audio_device_factory.cc b/content/renderer/media/audio_device_factory.cc
new file mode 100644
index 000000000000..555ebd631ae6
--- /dev/null
+++ b/content/renderer/media/audio_device_factory.cc
@@ -0,0 +1,32 @@
+// 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/renderer/media/audio_device_factory.h"
+
+#include "base/logging.h"
+#include "content/renderer/media/audio_device.h"
+
+namespace content {
+
+// static
+AudioDeviceFactory* AudioDeviceFactory::factory_ = NULL;
+
+// static
+media::AudioRendererSink* AudioDeviceFactory::Create() {
+  if (factory_) {
+    return factory_->CreateAudioDevice();
+  }
+  return new AudioDevice();
+}
+
+AudioDeviceFactory::AudioDeviceFactory() {
+  DCHECK(!factory_) << "Can't register two factories at once.";
+  factory_ = this;
+}
+
+AudioDeviceFactory::~AudioDeviceFactory() {
+  factory_ = NULL;
+}
+
+}  // namespace content
diff --git a/content/renderer/media/audio_device_factory.h b/content/renderer/media/audio_device_factory.h
new file mode 100644
index 000000000000..089b5496a9d9
--- /dev/null
+++ b/content/renderer/media/audio_device_factory.h
@@ -0,0 +1,47 @@
+// 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.
+
+#ifndef CONTENT_RENDERER_MEDIA_AUDIO_DEVICE_FACTORY_H_
+#define CONTENT_RENDERER_MEDIA_AUDIO_DEVICE_FACTORY_H_
+#pragma once
+
+#include "base/basictypes.h"
+#include "content/common/content_export.h"
+
+namespace media {
+class AudioRendererSink;
+}
+
+namespace content {
+
+// A factory for creating AudioRendererSinks. There is a global factory
+// function that can be installed for the purposes of testing to provide
+// a specialized AudioRendererSink class.
+// This class uses the same pattern as content::RenderViewHostFactory.
+class CONTENT_EXPORT AudioDeviceFactory {
+ public:
+  // Creates an AudioRendererSink using the currently registered factory,
+  // or the default one if no factory is registered. Ownership of the returned
+  // pointer will be passed to the caller.
+ static media::AudioRendererSink* Create();
+
+ protected:
+  AudioDeviceFactory();
+  virtual ~AudioDeviceFactory();
+
+  // You can derive from this class and specify an implementation for this
+  // function to create a different kind of AudioRendererSink for testing.
+  virtual media::AudioRendererSink* CreateAudioDevice() = 0;
+
+ private:
+  // The current globally registered factory. This is NULL when we should
+  // create the default AudioRendererSinks.
+  static AudioDeviceFactory* factory_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioDeviceFactory);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_AUDIO_DEVICE_FACTORY_H_
diff --git a/content/renderer/media/audio_message_filter.cc b/content/renderer/media/audio_message_filter.cc
index 65c41f34f3d2..81689925d5e4 100644
--- a/content/renderer/media/audio_message_filter.cc
+++ b/content/renderer/media/audio_message_filter.cc
@@ -9,11 +9,21 @@
 #include "base/time.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
+#include "content/renderer/render_thread_impl.h"
 #include "ipc/ipc_logging.h"
 
+AudioMessageFilter* AudioMessageFilter::filter_ = NULL;
+
+// static
+AudioMessageFilter* AudioMessageFilter::Get() {
+  return filter_;
+}
+
 AudioMessageFilter::AudioMessageFilter()
     : channel_(NULL) {
   VLOG(1) << "AudioMessageFilter::AudioMessageFilter()";
+  DCHECK(!filter_);
+  filter_ = this;
 }
 
 int32 AudioMessageFilter::AddDelegate(Delegate* delegate) {
@@ -69,6 +79,8 @@ void AudioMessageFilter::OnChannelClosing() {
 
 AudioMessageFilter::~AudioMessageFilter() {
   VLOG(1) << "AudioMessageFilter::~AudioMessageFilter()";
+  DCHECK(filter_);
+  filter_ = NULL;
 }
 
 void AudioMessageFilter::OnStreamCreated(
diff --git a/content/renderer/media/audio_message_filter.h b/content/renderer/media/audio_message_filter.h
index 40faeabec271..598600a92df3 100644
--- a/content/renderer/media/audio_message_filter.h
+++ b/content/renderer/media/audio_message_filter.h
@@ -39,6 +39,9 @@ class CONTENT_EXPORT AudioMessageFilter
 
   AudioMessageFilter();
 
+  // Getter for the one AudioMessageFilter object.
+  static AudioMessageFilter* Get();
+
   // Add a delegate to the map and return id of the entry.
   int32 AddDelegate(Delegate* delegate);
 
@@ -75,6 +78,9 @@ class CONTENT_EXPORT AudioMessageFilter
   // changed.
   void OnStreamStateChanged(int stream_id, AudioStreamState state);
 
+  // The singleton instance for this filter.
+  static AudioMessageFilter* filter_;
+
   // A map of stream ids to delegates.
   IDMap<Delegate> delegates_;
 
diff --git a/content/renderer/media/render_audiosourceprovider.cc b/content/renderer/media/render_audiosourceprovider.cc
index 10803f8a57e8..039f75fb8f8c 100644
--- a/content/renderer/media/render_audiosourceprovider.cc
+++ b/content/renderer/media/render_audiosourceprovider.cc
@@ -6,8 +6,10 @@
 
 #include "base/basictypes.h"
 #include "base/logging.h"
+#include "content/renderer/media/audio_device_factory.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProviderClient.h"
 
+using content::AudioDeviceFactory;
 using std::vector;
 using WebKit::WebVector;
 
@@ -19,12 +21,11 @@ RenderAudioSourceProvider::RenderAudioSourceProvider()
       volume_(1.0),
       renderer_(NULL),
       client_(NULL) {
-  // We create the AudioDevice here because it must be created in the
-  // main thread.  But we don't yet know the audio format (sample-rate, etc.)
-  // at this point.  Later, when Initialize() is called, we have
-  // the audio format information and call the AudioDevice::Initialize()
-  // method to fully initialize it.
-  default_sink_ = new AudioDevice();
+  // We create the AudioDevice here using the factory. But we don't yet know
+  // the audio format (sample-rate, etc.) at this point.  Later, when
+  // Initialize() is called, we have the audio format information and call
+  // the AudioDevice::Initialize() method to fully initialize it.
+  default_sink_ = AudioDeviceFactory::Create();
 }
 
 void RenderAudioSourceProvider::setClient(
diff --git a/content/renderer/media/render_audiosourceprovider.h b/content/renderer/media/render_audiosourceprovider.h
index ab4510f93e8e..a615708e12db 100644
--- a/content/renderer/media/render_audiosourceprovider.h
+++ b/content/renderer/media/render_audiosourceprovider.h
@@ -24,7 +24,7 @@
 
 #include <vector>
 
-#include "content/renderer/media/audio_device.h"
+#include "base/synchronization/lock.h"
 #include "media/base/audio_renderer_sink.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebVector.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProvider.h"
diff --git a/content/renderer/media/renderer_webaudiodevice_impl.cc b/content/renderer/media/renderer_webaudiodevice_impl.cc
index 07ea9c2c6e4e..f67c28212d61 100644
--- a/content/renderer/media/renderer_webaudiodevice_impl.cc
+++ b/content/renderer/media/renderer_webaudiodevice_impl.cc
@@ -4,6 +4,10 @@
 
 #include "content/renderer/media/renderer_webaudiodevice_impl.h"
 
+#include "base/logging.h"
+#include "content/renderer/media/audio_device_factory.h"
+
+using content::AudioDeviceFactory;
 using WebKit::WebAudioDevice;
 using WebKit::WebVector;
 
@@ -12,7 +16,8 @@ RendererWebAudioDeviceImpl::RendererWebAudioDeviceImpl(
     WebAudioDevice::RenderCallback* callback)
     : is_running_(false),
       client_callback_(callback) {
-  audio_device_ = new AudioDevice(params, this);
+  audio_device_ = AudioDeviceFactory::Create();
+  audio_device_->Initialize(params, this);
 }
 
 RendererWebAudioDeviceImpl::~RendererWebAudioDeviceImpl() {
diff --git a/content/renderer/media/renderer_webaudiodevice_impl.h b/content/renderer/media/renderer_webaudiodevice_impl.h
index 5b06242e4a6a..f984f1dc62ea 100644
--- a/content/renderer/media/renderer_webaudiodevice_impl.h
+++ b/content/renderer/media/renderer_webaudiodevice_impl.h
@@ -8,17 +8,17 @@
 #include <vector>
 
 #include "base/memory/ref_counted.h"
-#include "content/renderer/media/audio_device.h"
+#include "media/base/audio_renderer_sink.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebAudioDevice.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebVector.h"
 
 namespace media {
-class AudioParameters;
+class AudioRendererSink;
 }
 
 class RendererWebAudioDeviceImpl
     : public WebKit::WebAudioDevice,
-      public AudioDevice::RenderCallback {
+      public media::AudioRendererSink::RenderCallback {
  public:
   RendererWebAudioDeviceImpl(const media::AudioParameters& params,
                              WebKit::WebAudioDevice::RenderCallback* callback);
@@ -29,14 +29,14 @@ class RendererWebAudioDeviceImpl
   virtual void stop();
   virtual double sampleRate();
 
-  // AudioDevice::RenderCallback implementation.
+  // AudioRendererSink::RenderCallback implementation.
   virtual int Render(const std::vector<float*>& audio_data,
                      int number_of_frames,
                      int audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
  private:
-  scoped_refptr<AudioDevice> audio_device_;
+  scoped_refptr<media::AudioRendererSink> audio_device_;
   bool is_running_;
 
   // Weak reference to the callback into WebKit code.
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 37d2f3d6aeab..d1bbb1cfd635 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -8,12 +8,14 @@
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
+#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/audio_parameters.h"
 #include "media/audio/sample_rates.h"
 
+using content::AudioDeviceFactory;
 using media::AudioParameters;
 
 static const int64 kMillisecondsBetweenProcessCalls = 5000;
@@ -140,9 +142,13 @@ WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()
       playing_(false),
       recording_(false),
       agc_is_enabled_(false) {
-    DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()";
-    DCHECK(RenderThreadImpl::current()) <<
-        "WebRtcAudioDeviceImpl must be constructed on the render thread";
+  DVLOG(1) << "WebRtcAudioDeviceImpl::WebRtcAudioDeviceImpl()";
+  // TODO(henrika): remove this restriction when factory is used for the
+  // input side as well.
+  DCHECK(RenderThreadImpl::current()) <<
+      "WebRtcAudioDeviceImpl must be constructed on the render thread";
+  audio_output_device_ = AudioDeviceFactory::Create();
+  DCHECK(audio_output_device_);
 }
 
 WebRtcAudioDeviceImpl::~WebRtcAudioDeviceImpl() {
@@ -410,7 +416,6 @@ int32_t WebRtcAudioDeviceImpl::Init() {
     return 0;
 
   DCHECK(!audio_input_device_);
-  DCHECK(!audio_output_device_);
   DCHECK(!input_buffer_.get());
   DCHECK(!output_buffer_.get());
 
@@ -583,11 +588,10 @@ int32_t WebRtcAudioDeviceImpl::Init() {
   AddHistogramFramesPerBuffer(kAudioOutput, out_buffer_size);
   AddHistogramFramesPerBuffer(kAudioInput, in_buffer_size);
 
-  // Create and configure the audio rendering client.
-  audio_output_device_ = new AudioDevice(output_audio_parameters_, this);
+  // Configure the audio rendering client.
+  audio_output_device_->Initialize(output_audio_parameters_, this);
 
   DCHECK(audio_input_device_);
-  DCHECK(audio_output_device_);
 
   // Allocate local audio buffers based on the parameters above.
   // It is assumed that each audio sample contains 16 bits and each
@@ -627,13 +631,11 @@ int32_t WebRtcAudioDeviceImpl::Terminate() {
     return 0;
 
   DCHECK(audio_input_device_);
-  DCHECK(audio_output_device_);
   DCHECK(input_buffer_.get());
   DCHECK(output_buffer_.get());
 
   // Release all resources allocated in Init().
   audio_input_device_ = NULL;
-  audio_output_device_ = NULL;
   input_buffer_.reset();
   output_buffer_.reset();
 
@@ -697,7 +699,7 @@ int32_t WebRtcAudioDeviceImpl::SetRecordingDevice(WindowsDeviceType device) {
 
 int32_t WebRtcAudioDeviceImpl::PlayoutIsAvailable(bool* available) {
   DVLOG(1) << "PlayoutIsAvailable()";
-  *available = (audio_output_device_ != NULL);
+  *available = initialized();
   return 0;
 }
 
@@ -709,7 +711,7 @@ int32_t WebRtcAudioDeviceImpl::InitPlayout() {
 
 bool WebRtcAudioDeviceImpl::PlayoutIsInitialized() const {
   DVLOG(1) << "PlayoutIsInitialized()";
-  return (audio_output_device_ != NULL);
+  return initialized();
 }
 
 int32_t WebRtcAudioDeviceImpl::RecordingIsAvailable(bool* available) {
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index 6d4442ade71d..8c53248f3da4 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -16,8 +16,8 @@
 #include "base/message_loop_proxy.h"
 #include "base/time.h"
 #include "content/common/content_export.h"
-#include "content/renderer/media/audio_device.h"
 #include "content/renderer/media/audio_input_device.h"
+#include "media/base/audio_renderer_sink.h"
 #include "third_party/webrtc/modules/audio_device/main/interface/audio_device.h"
 
 // A WebRtcAudioDeviceImpl instance implements the abstract interface
@@ -220,7 +220,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   // We need this one to support runnable method tasks.
   static bool ImplementsThreadSafeReferenceCounting() { return true; }
 
-  // AudioDevice::RenderCallback implementation.
+  // media::AudioRendererSink::RenderCallback implementation.
   virtual int Render(const std::vector<float*>& audio_data,
                      int number_of_frames,
                      int audio_delay_milliseconds) OVERRIDE;
@@ -407,7 +407,7 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   scoped_refptr<AudioInputDevice> audio_input_device_;
 
   // Provides access to the native audio output layer in the browser process.
-  scoped_refptr<AudioDevice> audio_output_device_;
+  scoped_refptr<media::AudioRendererSink> audio_output_device_;
 
   // Weak reference to the audio callback.
   // The webrtc client defines |audio_transport_callback_| by calling
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index cf3a45364f5e..944741d02eb6 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -232,16 +232,16 @@ TEST_F(WebRTCAudioDeviceTest, TestValidOutputRates) {
 TEST_F(WebRTCAudioDeviceTest, Construct) {
   AudioUtilNoHardware audio_util(48000, 48000, CHANNEL_LAYOUT_MONO);
   SetAudioUtilCallback(&audio_util);
-  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
 
-  audio_device->SetSessionId(1);
+  webrtc_audio_device->SetSessionId(1);
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
-  int err = base->Init(audio_device);
+  int err = base->Init(webrtc_audio_device);
   EXPECT_EQ(0, err);
   EXPECT_EQ(0, base->Terminate());
 }
@@ -273,15 +273,15 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
-  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  webrtc_audio_device->SetSessionId(1);
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
-  int err = base->Init(audio_device);
+  int err = base->Init(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
@@ -302,8 +302,8 @@ TEST_F(WebRTCAudioDeviceTest, StartPlayout) {
       base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
   WaitForIOThreadCompletion();
 
-  EXPECT_TRUE(audio_device->playing());
-  EXPECT_FALSE(audio_device->recording());
+  EXPECT_TRUE(webrtc_audio_device->playing());
+  EXPECT_FALSE(webrtc_audio_device->recording());
   EXPECT_EQ(ch, media_process->channel_id());
   EXPECT_EQ(webrtc::kPlaybackPerChannel, media_process->type());
   EXPECT_EQ(80, media_process->packet_size());
@@ -342,15 +342,15 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {
   // for new interfaces, like OnSetAudioStreamRecording(). When done, add
   // EXPECT_CALL() macros here.
 
-  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  webrtc_audio_device->SetSessionId(1);
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
-  int err = base->Init(audio_device);
+  int err = base->Init(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
@@ -378,8 +378,8 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {
       base::TimeDelta::FromMilliseconds(TestTimeouts::action_timeout_ms())));
   WaitForIOThreadCompletion();
 
-  EXPECT_FALSE(audio_device->playing());
-  EXPECT_TRUE(audio_device->recording());
+  EXPECT_FALSE(webrtc_audio_device->playing());
+  EXPECT_TRUE(webrtc_audio_device->recording());
   EXPECT_EQ(ch, media_process->channel_id());
   EXPECT_EQ(webrtc::kRecordingPerChannel, media_process->type());
   EXPECT_EQ(80, media_process->packet_size());
@@ -419,16 +419,16 @@ TEST_F(WebRTCAudioDeviceTest, PlayLocalFile) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
-  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  webrtc_audio_device->SetSessionId(1);
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
-  int err = base->Init(audio_device);
+  int err = base->Init(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
@@ -487,15 +487,15 @@ TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
-  scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  webrtc_audio_device->SetSessionId(1);
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
-  int err = base->Init(audio_device);
+  int err = base->Init(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
diff --git a/content/renderer/render_view_impl.cc b/content/renderer/render_view_impl.cc
index ef3ad62ef4c7..d012810b227d 100644
--- a/content/renderer/render_view_impl.cc
+++ b/content/renderer/render_view_impl.cc
@@ -70,7 +70,6 @@
 #include "content/renderer/input_tag_speech_dispatcher.h"
 #include "content/renderer/java/java_bridge_dispatcher.h"
 #include "content/renderer/load_progress_tracker.h"
-#include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
 #include "content/renderer/media/media_stream_impl.h"
diff --git a/content/test/webrtc_audio_device_test.cc b/content/test/webrtc_audio_device_test.cc
index 770035b66084..44bc1cbe8d69 100644
--- a/content/test/webrtc_audio_device_test.cc
+++ b/content/test/webrtc_audio_device_test.cc
@@ -21,7 +21,9 @@
 #include "content/public/common/content_paths.h"
 #include "content/public/test/mock_resource_context.h"
 #include "content/public/test/test_browser_thread.h"
+#include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/audio_hardware.h"
+#include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
 #include "content/renderer/render_thread_impl.h"
-- 
2.11.4.GIT