MediaStream audio object graph untangling and clean-ups.

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 "base/strings/utf_string_conversions.h"
-#include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.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_);
-    MockMediaConstraintFactory constraint_factory;
-    scoped_refptr<WebRtcAudioCapturer> capturer(
-        WebRtcAudioCapturer::CreateCapturer(
-            -1, StreamDeviceInfo(),
-            constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
     scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
+        new WebRtcLocalAudioTrack(adapter.get()));
     blink::WebMediaStreamSource audio_source;
-    audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
+    audio_source.initialize(blink::WebString::fromUTF8("dummy_source_id"),
                             blink::WebMediaStreamSource::TypeAudio,
-                            base::UTF8ToUTF16("dummy_source_name"),
+                            blink::WebString::fromUTF8("dummy_source_name"),
                             false /* remote */, true /* readonly */);
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
     blink_track_.setExtraData(native_track.release());
     source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
     source_provider_->SetSinkParamsForTesting(sink_params_);
     source_provider_->OnSetFormat(source_params_);
   }
 
   void TearDown() override {
@@ skipping 81 matching lines @@
   // Stop the audio track.
   WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
       MediaStreamTrack::GetTrack(blink_track_));
   native_track->Stop();
 
   // Delete the source provider.
   source_provider_.reset();
 }
 
 }  // namespace content