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

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

 // Copyright 2013 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 <stddef.h>
 
 #include "base/logging.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/media_stream_audio_track.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"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebString.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 
 namespace content {
 
 class WebRtcLocalAudioSourceProviderTest : public testing::Test {
  protected:
   void SetUp() override {
     source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                          media::CHANNEL_LAYOUT_MONO, 48000, 16, 480);
     sink_params_.Reset(
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
         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,
                             blink::WebString::fromUTF8("dummy_source_name"),
                             false /* remote */);
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
-    blink_track_.setExtraData(native_track.release());
+    blink_track_.setExtraData(new MediaStreamAudioTrack(true));
     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }
 
   void TearDown() override {
     source_provider_.reset();
     blink_track_.reset();
     blink::WebHeap::collectAllGarbageForTesting();
   }
 
   media::AudioParameters source_params_;
   media::AudioParameters sink_params_;
   std::unique_ptr<media::AudioBus> sink_bus_;
   blink::WebMediaStreamTrack blink_track_;
   std::unique_ptr<WebRtcLocalAudioSourceProvider> source_provider_;
 };
 
 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()));
   for (size_t i = 0; i < audio_data.size(); ++i)
     audio_data[i] = sink_bus_->channel(i);
 
   // Enable the |source_provider_| by asking for data. This will inject
   // source_params_.frames_per_buffer() of zero into the resampler since there
   // no available data in the FIFO.
   source_provider_->provideInput(audio_data, sink_params_.frames_per_buffer());
@@ skipping 41 matching lines @@
     EXPECT_NEAR(0.5f, sink_bus_->channel(1)[0], 0.001f);
     EXPECT_DOUBLE_EQ(sink_bus_->channel(0)[0], sink_bus_->channel(1)[0]);
   }
 }
 
 TEST_F(WebRtcLocalAudioSourceProviderTest,
        DeleteSourceProviderBeforeStoppingTrack) {
   source_provider_.reset();
 
   // Stop the audio track.
-  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
-      MediaStreamTrack::GetTrack(blink_track_));
-  native_track->Stop();
+  MediaStreamAudioTrack::From(blink_track_)->Stop();
 }
 
 TEST_F(WebRtcLocalAudioSourceProviderTest,
        StopTrackBeforeDeletingSourceProvider) {
   // Stop the audio track.
-  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
-      MediaStreamTrack::GetTrack(blink_track_));
-  native_track->Stop();
+  MediaStreamAudioTrack::From(blink_track_)->Stop();
 
   // Delete the source provider.
   source_provider_.reset();
 }
 
 }  // namespace content