Adds AudioDevice factory for all audio clients in Chrome

content/renderer/media/audio_device.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/audio_device.h"
 
 #include "base/debug/trace_event.h"
 #include "base/message_loop.h"
 #include "base/threading/thread_restrictions.h"
 #include "base/time.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
 #include "content/common/view_messages.h"
-#include "content/renderer/render_thread_impl.h"
+#include "content/renderer/media/audio_message_filter.h"
 #include "media/audio/audio_output_controller.h"
 #include "media/audio/audio_util.h"
 
 using media::AudioRendererSink;
 
 // Takes care of invoking the render callback on the audio thread.
 // An instance of this class is created for each capture stream in
 // OnStreamCreated().
 class AudioDevice::AudioThreadCallback
     : public AudioDeviceThread::Callback {
  public:
   AudioThreadCallback(const media::AudioParameters& audio_parameters,
                       base::SharedMemoryHandle memory,
                       int memory_length,
                       AudioRendererSink::RenderCallback* render_callback);
   virtual ~AudioThreadCallback();
 
   virtual void MapSharedMemory() OVERRIDE;
 
   // Called whenever we receive notifications about pending data.
   virtual void Process(int pending_data) OVERRIDE;
 
  private:
   AudioRendererSink::RenderCallback* render_callback_;
   DISALLOW_COPY_AND_ASSIGN(AudioThreadCallback);
 };
 
-AudioDevice::AudioDevice()
+AudioDevice::AudioDevice(AudioMessageFilter* filter)

[Chris Rogers, 2012/06/14 17:41:44]

This is leaking internal implementation details for AudioDevice into the public
interface which is not a good idea.  The whole purpose of the AudioDevice
interface is to shield the client from any particular low-level details such as
this.  With this change, any client of AudioDevice must now get entangled into
AudioMessageFilters.

     : ScopedLoopObserver(ChildProcess::current()->io_message_loop()),
       callback_(NULL),
       volume_(1.0),
+      filter_(filter),
       stream_id_(0),
       play_on_start_(true),
       is_started_(false) {
-  filter_ = RenderThreadImpl::current()->audio_message_filter();
-}
-
-AudioDevice::AudioDevice(const media::AudioParameters& params,
-                         RenderCallback* callback)
-    : ScopedLoopObserver(ChildProcess::current()->io_message_loop()),
-      audio_parameters_(params),
-      callback_(callback),
-      volume_(1.0),
-      stream_id_(0),
-      play_on_start_(true),
-      is_started_(false) {
-  filter_ = RenderThreadImpl::current()->audio_message_filter();
+  CHECK(filter_) << "Invalid audio message filter.";
 }
 
 void AudioDevice::Initialize(const media::AudioParameters& params,
                              RenderCallback* callback) {
   CHECK_EQ(0, stream_id_) <<
       "AudioDevice::Initialize() must be called before Start()";
 
   CHECK(!callback_);  // Calling Initialize() twice?
 
   audio_parameters_ = params;
@@ skipping 220 matching lines @@
   // TODO(crogers/vrk): Figure out a way to avoid the float -> int -> float
   // conversions that happen in the <audio> and WebRTC scenarios.
   media::InterleaveFloatToInt(audio_data_, shared_memory_.memory(),
       audio_parameters_.frames_per_buffer(),
       audio_parameters_.bits_per_sample() / 8);
 
   // Let the host know we are done.
   media::SetActualDataSizeInBytes(&shared_memory_, memory_length_,
       num_frames * audio_parameters_.GetBytesPerFrame());
 }