move the APM to chrome.

content/renderer/media/webrtc_local_audio_track_unittest.cc

Index: content/renderer/media/webrtc_local_audio_track_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_track_unittest.cc b/content/renderer/media/webrtc_local_audio_track_unittest.cc
index 7b349ba13d1bced76e2a6f057702f6be48c480d7..8d9cad24f3298142e6d6491c901b4fb94be8c05c 100644
--- a/content/renderer/media/webrtc_local_audio_track_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_track_unittest.cc
@@ -137,10 +137,6 @@ class WebRtcLocalAudioTrackTest : public ::testing::Test {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
     capturer_ = WebRtcAudioCapturer::CreateCapturer();
-    WebRtcLocalAudioSourceProvider* source_provider =
-        static_cast<WebRtcLocalAudioSourceProvider*>(
-            capturer_->audio_source_provider());
-    source_provider->SetSinkParamsForTesting(params_);
     capturer_source_ = new MockCapturerSource();
     EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), 0))
         .WillOnce(Return());
@@ -177,6 +173,8 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectAndDisconnectOneSink) {
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
   EXPECT_TRUE(track->enabled());
 
@@ -223,6 +221,8 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_DisableEnableAudioTrack) {
   scoped_refptr<WebRtcLocalAudioTrack> track =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
   static_cast<webrtc::AudioTrackInterface*>(track.get())->
       GetRenderer()->AddChannel(0);
@@ -274,6 +274,8 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
@@ -299,6 +301,8 @@ TEST_F(WebRtcLocalAudioTrackTest, DISABLED_MultipleAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
     WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
         &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
@@ -354,6 +358,8 @@ TEST_F(WebRtcLocalAudioTrackTest, StartOneAudioTrack) {
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
 
   // When the track goes away, it will automatically stop the
@@ -375,6 +381,8 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
           &constraints);
   static_cast<webrtc::AudioTrackInterface*>(track_1.get())->
       GetRenderer()->AddChannel(0);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
@@ -393,6 +401,8 @@ TEST_F(WebRtcLocalAudioTrackTest, StartAndStopAudioTracks) {
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
   static_cast<webrtc::AudioTrackInterface*>(track_2.get())->
       GetRenderer()->AddChannel(1);
@@ -426,6 +436,8 @@ TEST_F(WebRtcLocalAudioTrackTest, SetNewSourceForCapturerAfterStartTrack) {
   scoped_refptr<WebRtcLocalAudioTrack> track =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track->audio_source_provider())->SetSinkParamsForTesting(params_);
   track->Start();
 
   // Setting new source to the capturer and the track should still get packets.
@@ -453,6 +465,8 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   scoped_refptr<WebRtcLocalAudioTrack> track_1 =
       WebRtcLocalAudioTrack::Create(std::string(), capturer_, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_1->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_1->Start();
 
   // Connect a number of network channels to the |track_1|.
@@ -475,10 +489,6 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   // Create a new capturer with new source with different audio format.
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
       WebRtcAudioCapturer::CreateCapturer());
-  WebRtcLocalAudioSourceProvider* source_provider =
-      static_cast<WebRtcLocalAudioSourceProvider*>(
-          new_capturer->audio_source_provider());
-  source_provider->SetSinkParamsForTesting(params_);
   scoped_refptr<MockCapturerSource> new_source(new MockCapturerSource());
   EXPECT_CALL(*new_source.get(), Initialize(_, new_capturer.get(), 0))
       .WillOnce(Return());
@@ -497,6 +507,8 @@ TEST_F(WebRtcLocalAudioTrackTest, ConnectTracksToDifferentCapturers) {
   scoped_refptr<WebRtcLocalAudioTrack> track_2 =
       WebRtcLocalAudioTrack::Create(std::string(), new_capturer, NULL, NULL,
           &constraints);
+  static_cast<WebRtcLocalAudioSourceProvider*>(
+      track_2->audio_source_provider())->SetSinkParamsForTesting(params_);
   track_2->Start();
 
   // Connect a number of network channels to the |track_2|.