RefCounted types should not have public destructors, content/browser part 2

content/renderer/media/render_audiosourceprovider.cc

 // Copyright (c) 2012 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 "content/renderer/media/render_audiosourceprovider.h"
 
 #include "base/basictypes.h"
 #include "base/logging.h"
 #include "third_party/WebKit/Source/WebKit/chromium/public/WebAudioSourceProviderClient.h"
 
 using std::vector;
 using WebKit::WebVector;
 
 RenderAudioSourceProvider::RenderAudioSourceProvider()
     : is_initialized_(false),
       channels_(0),
       sample_rate_(0),
       is_running_(false),
       volume_(1.0),
       renderer_(NULL),
       client_(NULL) {
   // We create the AudioDevice here because it must be created in the
   // main thread.  But we don't yet know the audio format (sample-rate, etc.)
   // at this point.  Later, when Initialize() is called, we have
   // the audio format information and call the AudioDevice::Initialize()
   // method to fully initialize it.
   default_sink_ = new AudioDevice();
 }
 
-RenderAudioSourceProvider::~RenderAudioSourceProvider() {}
+void RenderAudioSourceProvider::setClient(
+    WebKit::WebAudioSourceProviderClient* client) {
+  // Synchronize with other uses of client_ and default_sink_.
+  base::AutoLock auto_lock(sink_lock_);
+
+  if (client && client != client_) {
+    // Detach the audio renderer from normal playback.
+    default_sink_->Stop();
+
+    // The client will now take control by calling provideInput() periodically.
+    client_ = client;
+
+    if (is_initialized_) {
+      // The client needs to be notified of the audio format, if available.
+      // If the format is not yet available, we'll be notified later
+      // when Initialize() is called.
+
+      // Inform WebKit about the audio stream format.
+      client->setFormat(channels_, sample_rate_);
+    }
+  } else if (!client && client_) {
+    // Restore normal playback.
+    client_ = NULL;
+    // TODO(crogers): We should call default_sink_->Play() if we're
+    // in the playing state.
+  }
+}
+
+void RenderAudioSourceProvider::provideInput(
+    const WebVector<float*>& audio_data, size_t number_of_frames) {
+  DCHECK(client_);
+
+  if (renderer_ && is_initialized_ && is_running_) {
+    // Wrap WebVector as std::vector.
+    vector<float*> v(audio_data.size());
+    for (size_t i = 0; i < audio_data.size(); ++i)
+      v[i] = audio_data[i];
+
+    // TODO(crogers): figure out if we should volume scale here or in common
+    // WebAudio code.  In any case we need to take care of volume.
+    renderer_->Render(v, number_of_frames, 0);
+  } else {
+    // Provide silence if the source is not running.
+    for (size_t i = 0; i < audio_data.size(); ++i)
+      memset(audio_data[i], 0, sizeof(float) * number_of_frames);
+  }
+}
 
 void RenderAudioSourceProvider::Start() {
   base::AutoLock auto_lock(sink_lock_);
   if (!client_)
     default_sink_->Start();
   is_running_ = true;
 }
 
 void RenderAudioSourceProvider::Stop() {
   base::AutoLock auto_lock(sink_lock_);
@@ skipping 31 matching lines @@
 }
 
 void RenderAudioSourceProvider::GetVolume(double* volume) {
   if (!client_)
     default_sink_->GetVolume(volume);
   else if (volume)
     *volume = volume_;
 }
 
 void RenderAudioSourceProvider::Initialize(
-    const media::AudioParameters& params, RenderCallback* renderer) {
+    const media::AudioParameters& params,
+    RenderCallback* renderer) {
   base::AutoLock auto_lock(sink_lock_);
   CHECK(!is_initialized_);
   renderer_ = renderer;
 
   default_sink_->Initialize(params, renderer);
 
   // Keep track of the format in case the client hasn't yet been set.
   channels_ = params.channels();
   sample_rate_ = params.sample_rate();
 
   if (client_) {
     // Inform WebKit about the audio stream format.
     client_->setFormat(channels_, sample_rate_);
   }
 
   is_initialized_ = true;
 }
 
-void RenderAudioSourceProvider::setClient(
-    WebKit::WebAudioSourceProviderClient* client) {
-  // Synchronize with other uses of client_ and default_sink_.
-  base::AutoLock auto_lock(sink_lock_);
-
-  if (client && client != client_) {
-    // Detach the audio renderer from normal playback.
-    default_sink_->Stop();
-
-    // The client will now take control by calling provideInput() periodically.
-    client_ = client;
-
-    if (is_initialized_) {
-      // The client needs to be notified of the audio format, if available.
-      // If the format is not yet available, we'll be notified later
-      // when Initialize() is called.
-
-      // Inform WebKit about the audio stream format.
-      client->setFormat(channels_, sample_rate_);
-    }
-  } else if (!client && client_) {
-    // Restore normal playback.
-    client_ = NULL;
-    // TODO(crogers): We should call default_sink_->Play() if we're
-    // in the playing state.
-  }
-}
-
-void RenderAudioSourceProvider::provideInput(
-    const WebVector<float*>& audio_data, size_t number_of_frames) {
-  DCHECK(client_);
-
-  if (renderer_ && is_initialized_ && is_running_) {
-    // Wrap WebVector as std::vector.
-    vector<float*> v(audio_data.size());
-    for (size_t i = 0; i < audio_data.size(); ++i)
-      v[i] = audio_data[i];
-
-    // TODO(crogers): figure out if we should volume scale here or in common
-    // WebAudio code.  In any case we need to take care of volume.
-    renderer_->Render(v, number_of_frames, 0);
-  } else {
-    // Provide silence if the source is not running.
-    for (size_t i = 0; i < audio_data.size(); ++i)
-      memset(audio_data[i], 0, sizeof(float) * number_of_frames);
-  }
-}
+RenderAudioSourceProvider::~RenderAudioSourceProvider() {}