Make AudioParameters a class instead of a struct

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_(0.0),
+      samples_per_second_(0.0),
       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();
@@ skipping 38 matching lines @@
 }
 
 void RenderAudioSourceProvider::GetVolume(double* volume) {
   if (!client_)
     default_sink_->GetVolume(volume);
   else if (volume)
     *volume = volume_;
 }
 
 void RenderAudioSourceProvider::Initialize(
-    size_t buffer_size,
-    int channels,
-    double sample_rate,
-    AudioParameters::Format latency_format,
-    RenderCallback* renderer) {
+    const AudioParameters& params, RenderCallback* renderer) {
   base::AutoLock auto_lock(sink_lock_);
   CHECK(!is_initialized_);
   renderer_ = renderer;
 
-  default_sink_->Initialize(buffer_size,
-                            channels,
-                            sample_rate,
-                            latency_format,
-                            renderer);
+  default_sink_->Initialize(params, renderer);
+
+  // Keep track of the format in case the client hasn't yet been set.
+  channels_ = params.channels();
+  samples_per_second_ = params.samples_per_second();
 
   if (client_) {
     // Inform WebKit about the audio stream format.
-    client_->setFormat(channels, sample_rate);
+    client_->setFormat(channels_, samples_per_second_);
   }
 
-  // Keep track of the format in case the client hasn't yet been set.
-  channels_ = channels;
-  sample_rate_ = 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_->Pause(true);
 
     // 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_);
+      client->setFormat(channels_, samples_per_second_);
     }
   } 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);
   }
 }