Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

Index: content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
index 6eda8f178ecb3a219b1c1f4c6bd8ca70170b4d35..874d0db035dc578e0447c6950978a9daef120894 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
@@ -5,8 +5,9 @@
 #include <stddef.h>
 
 #include "base/logging.h"
-#include "content/renderer/media/media_stream_audio_track.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_parameters.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -26,6 +27,10 @@
         media::CHANNEL_LAYOUT_STEREO, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     sink_bus_ = media::AudioBus::Create(sink_params_);
+    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
+        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
+    std::unique_ptr<WebRtcLocalAudioTrack> native_track(
+        new WebRtcLocalAudioTrack(adapter.get()));
     blink::WebMediaStreamSource audio_source;
     audio_source.initialize(blink::WebString::fromUTF8("dummy_source_id"),
                             blink::WebMediaStreamSource::TypeAudio,
@@ -33,7 +38,7 @@
                             false /* remote */);
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
-    blink_track_.setExtraData(new MediaStreamAudioTrack(true));
+    blink_track_.setExtraData(native_track.release());
     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
@@ -53,10 +58,6 @@
 };
 
 TEST_F(WebRtcLocalAudioSourceProviderTest, VerifyDataFlow) {
-  // TODO(miu): This test should be re-worked so that the audio data and format
-  // is feed into a MediaStreamAudioSource and, through the
-  // MediaStreamAudioTrack, ultimately delivered to the |source_provider_|.
-
   // Point the WebVector into memory owned by |sink_bus_|.
   blink::WebVector<float*> audio_data(
       static_cast<size_t>(sink_bus_->channels()));
@@ -118,13 +119,17 @@
   source_provider_.reset();
 
   // Stop the audio track.
-  MediaStreamAudioTrack::From(blink_track_)->Stop();
+  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
+      MediaStreamTrack::GetTrack(blink_track_));
+  native_track->Stop();
 }
 
 TEST_F(WebRtcLocalAudioSourceProviderTest,
        StopTrackBeforeDeletingSourceProvider) {
   // Stop the audio track.
-  MediaStreamAudioTrack::From(blink_track_)->Stop();
+  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
+      MediaStreamTrack::GetTrack(blink_track_));
+  native_track->Stop();
 
   // Delete the source provider.
   source_provider_.reset();