Adds AudioDevice factory for all audio clients in Chrome

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index cf3a45364f5ee6642c1da1e46a7342b4536e97a9..cd0e57341a48830bf9d4c083ba27c89466b91283 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(
-      new WebRtcAudioDeviceImpl());
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl(audio_device_factory()));
 
-  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(
-      new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl(audio_device_factory()));
+  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(
-      new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl(audio_device_factory()));
+  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(
-      new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl(audio_device_factory()));
+  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(
-      new WebRtcAudioDeviceImpl());
-  audio_device->SetSessionId(1);
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl(audio_device_factory()));
+  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());