Introduce AudioHardwareConfig for renderer side audio device info.

As discussed, this creates a media::AudioHardwareConfig class lazily created,
owned, and filled by RenderThreadImpl via a single synchronous IPC and updated
via AudioMessageFilter.

This change does not plumb part 2 where AudioRendererMixer recreates itself based
on these device change updates.

BUG=157216
TEST=New unittest. Device changes on Windows/Mac continue to work.

Review URL: https://chromiumcodereview.appspot.com/11880009

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@180068 0039d316-1c4b-4281-b951-d872f2087c98

32 files changed, 400 insertions, 319 deletions

diff --git a/content/browser/renderer_host/media/audio_renderer_host.cc b/content/browser/renderer_host/media/audio_renderer_host.cc
index 3c87947..1901e2f 100644
--- a/content/browser/renderer_host/media/audio_renderer_host.cc
+++ b/content/browser/renderer_host/media/audio_renderer_host.cc
@@ -123,6 +123,17 @@ void AudioRendererHost::OnError(media::AudioOutputController* controller,
                  this, make_scoped_refptr(controller), error_code));
 }
 
+void AudioRendererHost::OnDeviceChange(media::AudioOutputController* controller,
+                                       int new_buffer_size,
+                                       int new_sample_rate) {
+  BrowserThread::PostTask(
+      BrowserThread::IO,
+      FROM_HERE,
+      base::Bind(&AudioRendererHost::DoSendDeviceChangeMessage,
+                 this, make_scoped_refptr(controller), new_buffer_size,
+                 new_sample_rate));
+}
+
 void AudioRendererHost::DoCompleteCreation(
     media::AudioOutputController* controller) {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
@@ -196,6 +207,19 @@ void AudioRendererHost::DoSendPausedMessage(
       entry->stream_id, media::AudioOutputIPCDelegate::kPaused));
 }
 
+void AudioRendererHost::DoSendDeviceChangeMessage(
+    media::AudioOutputController* controller, int new_buffer_size,
+    int new_sample_rate) {
+  DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
+
+  AudioEntry* entry = LookupByController(controller);
+  if (!entry)
+    return;
+
+  Send(new AudioMsg_NotifyDeviceChanged(
+      entry->stream_id, new_buffer_size, new_sample_rate));
+}
+
 void AudioRendererHost::DoHandleError(media::AudioOutputController* controller,
                                       int error_code) {
   DCHECK(BrowserThread::CurrentlyOn(BrowserThread::IO));
diff --git a/content/browser/renderer_host/media/audio_renderer_host.h b/content/browser/renderer_host/media/audio_renderer_host.h
index 15ebf89..c90d717 100644
--- a/content/browser/renderer_host/media/audio_renderer_host.h
+++ b/content/browser/renderer_host/media/audio_renderer_host.h
@@ -85,6 +85,9 @@ class CONTENT_EXPORT AudioRendererHost
   virtual void OnPaused(media::AudioOutputController* controller) OVERRIDE;
   virtual void OnError(media::AudioOutputController* controller,
                        int error_code) OVERRIDE;
+  virtual void OnDeviceChange(media::AudioOutputController* controller,
+                              int new_buffer_size,
+                              int new_sample_rate) OVERRIDE;
 
  private:
   friend class AudioRendererHostTest;
@@ -135,6 +138,8 @@ class CONTENT_EXPORT AudioRendererHost
   // Send a state change message to the renderer.
   void DoSendPlayingMessage(media::AudioOutputController* controller);
   void DoSendPausedMessage(media::AudioOutputController* controller);
+  void DoSendDeviceChangeMessage(media::AudioOutputController* controller,
+                                 int new_buffer_size, int new_sample_rate);
 
   // Handle error coming from audio stream.
   void DoHandleError(media::AudioOutputController* controller, int error_code);
diff --git a/content/browser/renderer_host/render_message_filter.cc b/content/browser/renderer_host/render_message_filter.cc
index 5393f5a..aacc6e9 100644
--- a/content/browser/renderer_host/render_message_filter.cc
+++ b/content/browser/renderer_host/render_message_filter.cc
@@ -417,14 +417,8 @@ bool RenderMessageFilter::OnMessageReceived(const IPC::Message& message,
     IPC_MESSAGE_HANDLER_DELAY_REPLY(ViewHostMsg_Keygen, OnKeygen)
     IPC_MESSAGE_HANDLER(ViewHostMsg_AsyncOpenFile, OnAsyncOpenFile)
     IPC_MESSAGE_HANDLER(ViewHostMsg_GetCPUUsage, OnGetCPUUsage)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareBufferSize,
-                        OnGetHardwareBufferSize)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
-                        OnGetHardwareInputSampleRate)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
-                        OnGetHardwareSampleRate)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputChannelLayout,
-                        OnGetHardwareInputChannelLayout)
+    IPC_MESSAGE_HANDLER(ViewHostMsg_GetAudioHardwareConfig,
+                        OnGetAudioHardwareConfig)
     IPC_MESSAGE_HANDLER(ViewHostMsg_GetMonitorColorProfile,
                         OnGetMonitorColorProfile)
     IPC_MESSAGE_HANDLER(ViewHostMsg_MediaLogEvent, OnMediaLogEvent)
@@ -778,24 +772,16 @@ void RenderMessageFilter::OnGetCPUUsage(int* cpu_usage) {
   *cpu_usage = cpu_usage_;
 }
 
-void RenderMessageFilter::OnGetHardwareBufferSize(uint32* buffer_size) {
-  *buffer_size = static_cast<uint32>(media::GetAudioHardwareBufferSize());
-}
+void RenderMessageFilter::OnGetAudioHardwareConfig(
+    int* output_buffer_size, int* output_sample_rate, int* input_sample_rate,
+    media::ChannelLayout* input_channel_layout) {
+  *output_buffer_size = media::GetAudioHardwareBufferSize();
+  *output_sample_rate = media::GetAudioHardwareSampleRate();
 
-void RenderMessageFilter::OnGetHardwareInputSampleRate(int* sample_rate) {
   // TODO(henrika): add support for all available input devices.
-  *sample_rate = media::GetAudioInputHardwareSampleRate(
+  *input_sample_rate = media::GetAudioInputHardwareSampleRate(
       media::AudioManagerBase::kDefaultDeviceId);
-}
-
-void RenderMessageFilter::OnGetHardwareSampleRate(int* sample_rate) {
-  *sample_rate = media::GetAudioHardwareSampleRate();
-}
-
-void RenderMessageFilter::OnGetHardwareInputChannelLayout(
-    media::ChannelLayout* layout) {
-  // TODO(henrika): add support for all available input devices.
-  *layout = media::GetAudioInputHardwareChannelLayout(
+  *input_channel_layout = media::GetAudioInputHardwareChannelLayout(
       media::AudioManagerBase::kDefaultDeviceId);
 }
 
diff --git a/content/browser/renderer_host/render_message_filter.h b/content/browser/renderer_host/render_message_filter.h
index 1d77fb3..af6036e 100644
--- a/content/browser/renderer_host/render_message_filter.h
+++ b/content/browser/renderer_host/render_message_filter.h
@@ -189,10 +189,9 @@ class RenderMessageFilter : public BrowserMessageFilter {
 
   void OnGetCPUUsage(int* cpu_usage);
 
-  void OnGetHardwareBufferSize(uint32* buffer_size);
-  void OnGetHardwareInputSampleRate(int* sample_rate);
-  void OnGetHardwareSampleRate(int* sample_rate);
-  void OnGetHardwareInputChannelLayout(media::ChannelLayout* layout);
+  void OnGetAudioHardwareConfig(int* output_buffer_size,
+                                int* output_sample_rate, int* input_sample_rate,
+                                media::ChannelLayout* input_channel_layout);
 
   // Used to look up the monitor color profile.
   void OnGetMonitorColorProfile(std::vector<char>* profile);
diff --git a/content/common/media/audio_messages.h b/content/common/media/audio_messages.h
index cac43d1..8977417 100644
--- a/content/common/media/audio_messages.h
+++ b/content/common/media/audio_messages.h
@@ -63,6 +63,13 @@ IPC_MESSAGE_CONTROL4(AudioInputMsg_NotifyStreamCreated,
                      uint32 /* length */)
 #endif
 
+// Notification message sent from AudioRendererHost to renderer after an output
+// device change has occurred.
+IPC_MESSAGE_CONTROL3(AudioMsg_NotifyDeviceChanged,
+                     int /* stream_id */,
+                     int /* new_buffer_size */,
+                     int /* new_sample_rate */)
+
 // Notification message sent from AudioRendererHost to renderer for state
 // update after the renderer has requested a Create/Start/Close.
 IPC_MESSAGE_CONTROL2(AudioMsg_NotifyStreamStateChanged,
diff --git a/content/common/view_messages.h b/content/common/view_messages.h
index 0b36fdc..4e064fa 100644
--- a/content/common/view_messages.h
+++ b/content/common/view_messages.h
@@ -720,20 +720,11 @@ IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GenerateRoutingID,
                             int /* routing_id */)
 
 // Asks the browser for the default audio hardware buffer-size.
-IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GetHardwareBufferSize,
-                            uint32 /* buffer_size */)
-
-// Asks the browser for the default audio input hardware sample-rate.
-IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GetHardwareInputSampleRate,
-                            int /* sample_rate */)
-
-// Asks the browser for the default audio hardware sample-rate.
-IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GetHardwareSampleRate,
-                            int /* sample_rate */)
-
-// Asks the browser for the default channel layout.
-IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GetHardwareInputChannelLayout,
-                            media::ChannelLayout /* channel layout */)
+IPC_SYNC_MESSAGE_CONTROL0_4(ViewHostMsg_GetAudioHardwareConfig,
+                            int /* output_buffer_size */,
+                            int /* output_sample_rate */,
+                            int /* input_sample_rate */,
+                            media::ChannelLayout /* input_channel_layout */)
 
 // Asks the browser for CPU usage of the renderer process in percents.
 IPC_SYNC_MESSAGE_CONTROL0_1(ViewHostMsg_GetCPUUsage,
diff --git a/content/content_renderer.gypi b/content/content_renderer.gypi
index 633717c..46b29b1 100644
--- a/content/content_renderer.gypi
+++ b/content/content_renderer.gypi
@@ -114,8 +114,6 @@
     'renderer/load_progress_tracker.h',
     'renderer/media/audio_device_factory.cc',
     'renderer/media/audio_device_factory.h',
-    'renderer/media/audio_hardware.cc',
-    'renderer/media/audio_hardware.h',
     'renderer/media/audio_input_message_filter.cc',
     'renderer/media/audio_input_message_filter.h',
     'renderer/media/audio_message_filter.cc',
diff --git a/content/renderer/media/audio_hardware.cc b/content/renderer/media/audio_hardware.cc
deleted file mode 100644
index 9f22a2d..0000000
--- a/content/renderer/media/audio_hardware.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-// 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_hardware.h"
-
-#include "base/logging.h"
-#include "content/common/view_messages.h"
-#include "content/renderer/render_thread_impl.h"
-
-using media::ChannelLayout;
-using media::CHANNEL_LAYOUT_NONE;
-
-static int output_sample_rate = 0;
-static int input_sample_rate = 0;
-static size_t output_buffer_size = 0;
-static ChannelLayout input_channel_layout = CHANNEL_LAYOUT_NONE;
-
-namespace content {
-
-int GetAudioOutputSampleRate() {
-  DCHECK(RenderThreadImpl::current() != NULL);
-
-  if (!output_sample_rate) {
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareSampleRate(&output_sample_rate));
-  }
-  return output_sample_rate;
-}
-
-int GetAudioInputSampleRate() {
-  DCHECK(RenderThreadImpl::current() != NULL);
-
-  if (!input_sample_rate) {
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareInputSampleRate(&input_sample_rate));
-  }
-  return input_sample_rate;
-}
-
-size_t GetAudioOutputBufferSize() {
-  DCHECK(RenderThreadImpl::current() != NULL);
-
-  if (!output_buffer_size) {
-    uint32 buffer_size = 0;
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareBufferSize(&buffer_size));
-    output_buffer_size = buffer_size;
-  }
-
-  return output_buffer_size;
-}
-
-ChannelLayout GetAudioInputChannelLayout() {
-  DCHECK(RenderThreadImpl::current() != NULL);
-
-  if (input_channel_layout == CHANNEL_LAYOUT_NONE) {
-    ChannelLayout layout = CHANNEL_LAYOUT_NONE;
-    RenderThreadImpl::current()->Send(
-        new ViewHostMsg_GetHardwareInputChannelLayout(&layout));
-    input_channel_layout = layout;
-  }
-
-  return input_channel_layout;
-}
-
-void ResetAudioCache() {
-  DCHECK(RenderThreadImpl::current() != NULL);
-
-  output_sample_rate = 0.0;
-  input_sample_rate = 0.0;
-  output_buffer_size = 0;
-  input_channel_layout = CHANNEL_LAYOUT_NONE;
-}
-
-}  // namespace content
diff --git a/content/renderer/media/audio_hardware.h b/content/renderer/media/audio_hardware.h
deleted file mode 100644
index 3171387..0000000
--- a/content/renderer/media/audio_hardware.h
+++ /dev/null
@@ -1,40 +0,0 @@
-// 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.
-//
-#ifndef CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
-#define CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
-
-#include "base/basictypes.h"
-#include "content/common/content_export.h"
-#include "media/base/channel_layout.h"
-
-// This file contains static methods to query audio hardware properties from
-// the browser process.  Values are cached to avoid unnecessary round trips,
-// but the cache can be cleared if needed (currently only used by tests).
-
-namespace content {
-
-// Fetch the sample rate of the default audio output end point device.
-// Must be called from RenderThreadImpl::current().
-CONTENT_EXPORT int GetAudioOutputSampleRate();
-
-// Fetch the sample rate of the default audio input end point device.
-// Must be called from RenderThreadImpl::current().
-CONTENT_EXPORT int GetAudioInputSampleRate();
-
-// Fetch the buffer size we use for the default output device.
-// Must be called from RenderThreadImpl::current().
-// Must be used in conjunction with AUDIO_PCM_LOW_LATENCY.
-CONTENT_EXPORT size_t GetAudioOutputBufferSize();
-
-// Fetch the audio channel layout for the default input device.
-// Must be called from RenderThreadImpl::current().
-CONTENT_EXPORT media::ChannelLayout GetAudioInputChannelLayout();
-
-// Forces the next call to any of the Get functions to query the hardware
-// and repopulate the cache.
-CONTENT_EXPORT void ResetAudioCache();
-}  // namespace content
-
-#endif  // CONTENT_RENDERER_MEDIA_AUDIO_HARDWARE_H_
diff --git a/content/renderer/media/audio_message_filter.cc b/content/renderer/media/audio_message_filter.cc
index 22915c5..4eef36c 100644
--- a/content/renderer/media/audio_message_filter.cc
+++ b/content/renderer/media/audio_message_filter.cc
@@ -23,21 +23,22 @@ AudioMessageFilter* AudioMessageFilter::Get() {
 
 AudioMessageFilter::AudioMessageFilter()
     : next_stream_id_(1),
-      channel_(NULL) {
+      channel_(NULL),
+      audio_hardware_config_(NULL) {
   DVLOG(1) << "AudioMessageFilter::AudioMessageFilter()";
   DCHECK(!filter_);
   filter_ = this;
 }
 
 int AudioMessageFilter::AddDelegate(media::AudioOutputIPCDelegate* delegate) {
-  base::AutoLock guard(delegates_lock_);
+  base::AutoLock auto_lock(lock_);
   const int id = next_stream_id_++;
   delegates_.insert(std::make_pair(id, delegate));
   return id;
 }
 
 void AudioMessageFilter::RemoveDelegate(int id) {
-  base::AutoLock guard(delegates_lock_);
+  base::AutoLock auto_lock(lock_);
   delegates_.erase(id);
 }
 
@@ -96,6 +97,7 @@ bool AudioMessageFilter::OnMessageReceived(const IPC::Message& message) {
   IPC_BEGIN_MESSAGE_MAP(AudioMessageFilter, message)
     IPC_MESSAGE_HANDLER(AudioMsg_NotifyStreamCreated, OnStreamCreated)
     IPC_MESSAGE_HANDLER(AudioMsg_NotifyStreamStateChanged, OnStreamStateChanged)
+    IPC_MESSAGE_HANDLER(AudioMsg_NotifyDeviceChanged, OnOutputDeviceChanged)
     IPC_MESSAGE_UNHANDLED(handled = false)
   IPC_END_MESSAGE_MAP()
   return handled;
@@ -115,7 +117,7 @@ void AudioMessageFilter::OnChannelClosing() {
 
   DelegateMap zombies;
   {
-    base::AutoLock guard(delegates_lock_);
+    base::AutoLock auto_lock(lock_);
     delegates_.swap(zombies);
   }
 
@@ -153,7 +155,7 @@ void AudioMessageFilter::OnStreamCreated(
 #endif
 
   {
-    base::AutoLock guard(delegates_lock_);
+    base::AutoLock auto_lock(lock_);
     DelegateMap::const_iterator it = delegates_.find(stream_id);
     if (it != delegates_.end()) {
       it->second->OnStreamCreated(handle, socket_handle, length);
@@ -169,7 +171,7 @@ void AudioMessageFilter::OnStreamCreated(
 
 void AudioMessageFilter::OnStreamStateChanged(
     int stream_id, media::AudioOutputIPCDelegate::State state) {
-  base::AutoLock guard(delegates_lock_);
+  base::AutoLock auto_lock(lock_);
   DelegateMap::const_iterator it = delegates_.find(stream_id);
   DLOG_IF(WARNING, it == delegates_.end())
       << "No delegate found for state change. " << state;
@@ -177,4 +179,24 @@ void AudioMessageFilter::OnStreamStateChanged(
     it->second->OnStateChanged(state);
 }
 
+void AudioMessageFilter::OnOutputDeviceChanged(int stream_id,
+                                               int new_buffer_size,
+                                               int new_sample_rate) {
+  base::AutoLock auto_lock(lock_);
+
+  // Ignore the message if an audio hardware config hasn't been created; this
+  // can occur if the renderer is using the high latency audio path.
+  // TODO(dalecurtis): After http://crbug.com/173435 is fixed, convert to CHECK.
+  if (!audio_hardware_config_)
+    return;
+
+  audio_hardware_config_->UpdateOutputConfig(new_buffer_size, new_sample_rate);
+}
+
+void AudioMessageFilter::SetAudioHardwareConfig(
+    media::AudioHardwareConfig* config) {
+  base::AutoLock auto_lock(lock_);
+  audio_hardware_config_ = config;
+}
+
 }  // namespace content
diff --git a/content/renderer/media/audio_message_filter.h b/content/renderer/media/audio_message_filter.h
index de746d5..1172607 100644
--- a/content/renderer/media/audio_message_filter.h
+++ b/content/renderer/media/audio_message_filter.h
@@ -19,6 +19,7 @@
 #include "ipc/ipc_channel_proxy.h"
 #include "media/audio/audio_buffers_state.h"
 #include "media/audio/audio_output_ipc.h"
+#include "media/base/audio_hardware_config.h"
 
 namespace content {
 
@@ -37,8 +38,8 @@ class CONTENT_EXPORT AudioMessageFilter
   // media::AudioOutputIPC implementation.
   virtual int AddDelegate(media::AudioOutputIPCDelegate* delegate) OVERRIDE;
   virtual void RemoveDelegate(int id) OVERRIDE;
-  virtual void CreateStream(int stream_id,
-      const media::AudioParameters& params, int input_channels) OVERRIDE;
+  virtual void CreateStream(int stream_id, const media::AudioParameters& params,
+                            int input_channels) OVERRIDE;
   virtual void PlayStream(int stream_id) OVERRIDE;
   virtual void PauseStream(int stream_id) OVERRIDE;
   virtual void FlushStream(int stream_id) OVERRIDE;
@@ -51,6 +52,11 @@ class CONTENT_EXPORT AudioMessageFilter
   virtual void OnFilterRemoved() OVERRIDE;
   virtual void OnChannelClosing() OVERRIDE;
 
+  // When set, AudioMessageFilter will update the AudioHardwareConfig with new
+  // configuration values as recieved by OnOutputDeviceChanged().  The provided
+  // |config| must outlive AudioMessageFilter.
+  void SetAudioHardwareConfig(media::AudioHardwareConfig* config);
+
  protected:
   virtual ~AudioMessageFilter();
 
@@ -75,21 +81,26 @@ class CONTENT_EXPORT AudioMessageFilter
   void OnStreamStateChanged(int stream_id,
                             media::AudioOutputIPCDelegate::State state);
 
+  // Received when the browser process detects an output device change.
+  void OnOutputDeviceChanged(int stream_id, int new_buffer_size,
+                             int new_sample_rate);
+
   // The singleton instance for this filter.
   static AudioMessageFilter* filter_;
 
-  // Guards delegates_ since AddDelegate() and RemoveDelegate() are called from
-  // threads other than the IO thread.
-  base::Lock delegates_lock_;
+  // Unique ID to use for next added delegate.
+  int next_stream_id_;
+  IPC::Channel* channel_;
+
+  // Guards all variables below which are accessed from multiple threads.
+  base::Lock lock_;
 
   // A map of stream ids to delegates.
   typedef base::hash_map<int, media::AudioOutputIPCDelegate*> DelegateMap;
   DelegateMap delegates_;
 
-  // Unique ID to use for next added delegate.
-  int next_stream_id_;
-
-  IPC::Channel* channel_;
+  // Audio hardware configuration to update when OnOutputDeviceChanged() fires.
+  media::AudioHardwareConfig* audio_hardware_config_;
 
   DISALLOW_COPY_AND_ASSIGN(AudioMessageFilter);
 };
diff --git a/content/renderer/media/audio_renderer_mixer_manager.cc b/content/renderer/media/audio_renderer_mixer_manager.cc
index 71fafcb..d6e88c8 100644
--- a/content/renderer/media/audio_renderer_mixer_manager.cc
+++ b/content/renderer/media/audio_renderer_mixer_manager.cc
@@ -8,15 +8,15 @@
 #include "base/bind_helpers.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/renderer_audio_output_device.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/audio_renderer_mixer.h"
 #include "media/base/audio_renderer_mixer_input.h"
 
 namespace content {
 
-AudioRendererMixerManager::AudioRendererMixerManager(int hardware_sample_rate,
-                                                     int hardware_buffer_size)
-    : hardware_sample_rate_(hardware_sample_rate),
-      hardware_buffer_size_(hardware_buffer_size),
+AudioRendererMixerManager::AudioRendererMixerManager(
+    media::AudioHardwareConfig* hardware_config)
+    : hardware_config_(hardware_config),
       sink_for_testing_(NULL) {
 }
 
@@ -57,7 +57,7 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
 #if defined(OS_LINUX)
   int sample_rate = params.sample_rate();
 #else
-  int sample_rate = hardware_sample_rate_;
+  int sample_rate = hardware_config_->GetOutputSampleRate();
 #endif
 
   // Create output parameters based on the audio hardware configuration for
@@ -65,7 +65,7 @@ media::AudioRendererMixer* AudioRendererMixerManager::GetMixer(
   // know that works well for WebAudio and WebRTC.
   media::AudioParameters output_params(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(),
-      sample_rate, 16, hardware_buffer_size_);
+      sample_rate, 16, hardware_config_->GetOutputBufferSize());
 
   // If we've created invalid output parameters, simply pass on the input params
   // and let the browser side handle automatic fallback.
diff --git a/content/renderer/media/audio_renderer_mixer_manager.h b/content/renderer/media/audio_renderer_mixer_manager.h
index 20afcb8..dd2575b 100644
--- a/content/renderer/media/audio_renderer_mixer_manager.h
+++ b/content/renderer/media/audio_renderer_mixer_manager.h
@@ -13,6 +13,7 @@
 #include "media/audio/audio_parameters.h"
 
 namespace media {
+class AudioHardwareConfig;
 class AudioRendererMixer;
 class AudioRendererMixerInput;
 class AudioRendererSink;
@@ -35,8 +36,11 @@ namespace content {
 // via the shared memory.  See http://crbug.com/114700.
 class CONTENT_EXPORT AudioRendererMixerManager {
  public:
-  // Construct an instance using the given audio hardware configuration.
-  AudioRendererMixerManager(int hardware_sample_rate, int hardware_buffer_size);
+  // Construct an instance using the given audio hardware configuration.  The
+  // provided |hardware_config| is not owned by AudioRendererMixerManager and
+  // must outlive it.
+  explicit AudioRendererMixerManager(
+      media::AudioHardwareConfig* hardware_config);
   ~AudioRendererMixerManager();
 
   // Creates an AudioRendererMixerInput with the proper callbacks necessary to
@@ -82,8 +86,7 @@ class CONTENT_EXPORT AudioRendererMixerManager {
 
   // Audio hardware configuration.  Used to construct output AudioParameters for
   // each AudioRendererMixer instance.
-  int hardware_sample_rate_;
-  int hardware_buffer_size_;
+  media::AudioHardwareConfig* const hardware_config_;
 
   media::AudioRendererSink* sink_for_testing_;
 
diff --git a/content/renderer/media/audio_renderer_mixer_manager_unittest.cc b/content/renderer/media/audio_renderer_mixer_manager_unittest.cc
index 46e78f3..0d00942 100644
--- a/content/renderer/media/audio_renderer_mixer_manager_unittest.cc
+++ b/content/renderer/media/audio_renderer_mixer_manager_unittest.cc
@@ -2,9 +2,11 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include "base/logging.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "content/renderer/media/audio_renderer_mixer_manager.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/audio_renderer_mixer.h"
 #include "media/base/audio_renderer_mixer_input.h"
 #include "media/base/fake_audio_render_callback.h"
@@ -24,8 +26,9 @@ static const int kAnotherRenderViewId = 456;
 
 class AudioRendererMixerManagerTest : public testing::Test {
  public:
-  AudioRendererMixerManagerTest() {
-    manager_.reset(new AudioRendererMixerManager(kSampleRate, kBufferSize));
+  AudioRendererMixerManagerTest()
+      : fake_config_(kBufferSize, kSampleRate, 0, media::CHANNEL_LAYOUT_NONE) {
+    manager_.reset(new AudioRendererMixerManager(&fake_config_));
 
     // We don't want to deal with instantiating a real AudioOutputDevice since
     // it's not important to our testing, so we inject a mock.
@@ -49,6 +52,7 @@ class AudioRendererMixerManagerTest : public testing::Test {
   }
 
  protected:
+  media::AudioHardwareConfig fake_config_;
   scoped_ptr<AudioRendererMixerManager> manager_;
   scoped_refptr<media::MockAudioRendererSink> mock_sink_;
 
diff --git a/content/renderer/media/webrtc_audio_capturer.cc b/content/renderer/media/webrtc_audio_capturer.cc
index 16836f2a..a3c6ebf 100644
--- a/content/renderer/media/webrtc_audio_capturer.cc
+++ b/content/renderer/media/webrtc_audio_capturer.cc
@@ -10,7 +10,6 @@
 #include "base/string_util.h"
 #include "content/common/child_process.h"
 #include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_renderer.h"
 #include "media/audio/audio_util.h"
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc
index 21fc788..32a5e49 100644
--- a/content/renderer/media/webrtc_audio_device_impl.cc
+++ b/content/renderer/media/webrtc_audio_device_impl.cc
@@ -8,7 +8,6 @@
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "base/win/windows_version.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
 #include "content/renderer/render_thread_impl.h"
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index 30157b0..e988aa2 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -4,13 +4,14 @@
 
 #include "base/environment.h"
 #include "base/test/test_timeouts.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
+#include "content/renderer/render_thread_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
-#include "media/audio/audio_manager.h"
+#include "media/audio/audio_manager_base.h"
 #include "media/audio/audio_util.h"
+#include "media/base/audio_hardware_config.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/include/voe_base.h"
@@ -30,52 +31,14 @@ namespace {
 
 const int kRenderViewId = 1;
 
-class AudioUtil : public AudioUtilInterface {
- public:
-  AudioUtil() {}
-
-  virtual int GetAudioHardwareSampleRate() OVERRIDE {
-    return media::GetAudioHardwareSampleRate();
-  }
-  virtual int GetAudioInputHardwareSampleRate(
-      const std::string& device_id) OVERRIDE {
-    return media::GetAudioInputHardwareSampleRate(device_id);
-  }
-  virtual media::ChannelLayout GetAudioInputHardwareChannelLayout(
-      const std::string& device_id) OVERRIDE {
-    return media::GetAudioInputHardwareChannelLayout(device_id);
-  }
- private:
-  DISALLOW_COPY_AND_ASSIGN(AudioUtil);
-};
-
-class AudioUtilNoHardware : public AudioUtilInterface {
- public:
-  AudioUtilNoHardware(int output_rate, int input_rate,
-                      media::ChannelLayout input_channel_layout)
-      : output_rate_(output_rate),
-        input_rate_(input_rate),
-        input_channel_layout_(input_channel_layout) {
-  }
-
-  virtual int GetAudioHardwareSampleRate() OVERRIDE {
-    return output_rate_;
-  }
-  virtual int GetAudioInputHardwareSampleRate(
-      const std::string& device_id) OVERRIDE {
-    return input_rate_;
-  }
-  virtual media::ChannelLayout GetAudioInputHardwareChannelLayout(
-      const std::string& device_id) OVERRIDE {
-    return input_channel_layout_;
-  }
-
- private:
-  int output_rate_;
-  int input_rate_;
-  media::ChannelLayout input_channel_layout_;
-  DISALLOW_COPY_AND_ASSIGN(AudioUtilNoHardware);
-};
+scoped_ptr<media::AudioHardwareConfig> CreateRealHardwareConfig() {
+  return make_scoped_ptr(new media::AudioHardwareConfig(
+      media::GetAudioHardwareBufferSize(), media::GetAudioHardwareSampleRate(),
+      media::GetAudioInputHardwareSampleRate(
+          media::AudioManagerBase::kDefaultDeviceId),
+      media::GetAudioInputHardwareChannelLayout(
+          media::AudioManagerBase::kDefaultDeviceId)));
+}
 
 // Return true if at least one element in the array matches |value|.
 bool FindElementInArray(int* array, int size, int value) {
@@ -95,8 +58,11 @@ bool HardwareSampleRatesAreValid() {
   int valid_input_rates[] = {16000, 32000, 44100, 48000, 96000};
   int valid_output_rates[] = {44100, 48000, 96000};
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+
   // Verify the input sample rate.
-  int input_sample_rate = GetAudioInputSampleRate();
+  int input_sample_rate = hardware_config->GetInputSampleRate();
 
   if (!FindElementInArray(valid_input_rates, arraysize(valid_input_rates),
                           input_sample_rate)) {
@@ -105,7 +71,7 @@ bool HardwareSampleRatesAreValid() {
   }
 
   // Given that the input rate was OK, verify the output rate as well.
-  int output_sample_rate = GetAudioOutputSampleRate();
+  int output_sample_rate = hardware_config->GetOutputSampleRate();
   if (!FindElementInArray(valid_output_rates, arraysize(valid_output_rates),
                           output_sample_rate)) {
     LOG(WARNING) << "Non-supported output sample rate detected.";
@@ -124,10 +90,14 @@ bool InitializeCapturer(WebRtcAudioDeviceImpl* webrtc_audio_device) {
   if (!capturer)
     return false;
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+
   // Use native capture sample rate and channel configuration to get some
   // action in this test.
-  int sample_rate = GetAudioInputSampleRate();
-  media::ChannelLayout channel_layout = GetAudioInputChannelLayout();
+  int sample_rate = hardware_config->GetInputSampleRate();
+  media::ChannelLayout channel_layout =
+      hardware_config->GetInputChannelLayout();
   if (!capturer->Initialize(channel_layout, sample_rate))
     return false;
 
@@ -246,8 +216,9 @@ TEST_F(WebRTCAudioDeviceTest, TestValidOutputRates) {
 // Basic test that instantiates and initializes an instance of
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
-  AudioUtilNoHardware audio_util(48000, 48000, media::CHANNEL_LAYOUT_MONO);
-  SetAudioUtilCallback(&audio_util);
+  media::AudioHardwareConfig audio_config(
+      480, 48000, 48000, media::CHANNEL_LAYOUT_MONO);
+  SetAudioHardwareConfig(&audio_config);
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
@@ -279,8 +250,8 @@ TEST_F(WebRTCAudioDeviceTest, DISABLED_StartPlayout) {
     return;
   }
 
-  AudioUtil audio_util;
-  SetAudioUtilCallback(&audio_util);
+  scoped_ptr<media::AudioHardwareConfig> config = CreateRealHardwareConfig();
+  SetAudioHardwareConfig(config.get());
 
   if (!HardwareSampleRatesAreValid())
     return;
@@ -357,8 +328,8 @@ TEST_F(WebRTCAudioDeviceTest, StartRecording) {
     return;
   }
 
-  AudioUtil audio_util;
-  SetAudioUtilCallback(&audio_util);
+  scoped_ptr<media::AudioHardwareConfig> config = CreateRealHardwareConfig();
+  SetAudioHardwareConfig(config.get());
 
   if (!HardwareSampleRatesAreValid())
     return;
@@ -430,8 +401,8 @@ TEST_F(WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
   std::string file_path(
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
 
-  AudioUtil audio_util;
-  SetAudioUtilCallback(&audio_util);
+  scoped_ptr<media::AudioHardwareConfig> config = CreateRealHardwareConfig();
+  SetAudioHardwareConfig(config.get());
 
   if (!HardwareSampleRatesAreValid())
     return;
@@ -501,8 +472,8 @@ TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {
     return;
   }
 
-  AudioUtil audio_util;
-  SetAudioUtilCallback(&audio_util);
+  scoped_ptr<media::AudioHardwareConfig> config = CreateRealHardwareConfig();
+  SetAudioHardwareConfig(config.get());
 
   if (!HardwareSampleRatesAreValid())
     return;
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index c21c662..0d33713 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -8,11 +8,13 @@
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
 #include "content/renderer/media/audio_device_factory.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/renderer_audio_output_device.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/render_thread_impl.h"
 #include "media/audio/audio_util.h"
 #include "media/audio/sample_rates.h"
+#include "media/base/audio_hardware_config.h"
+
 #if defined(OS_WIN)
 #include "base/win/windows_version.h"
 #include "media/audio/win/core_audio_util_win.h"
@@ -24,10 +26,10 @@ namespace {
 
 // Supported hardware sample rates for output sides.
 #if defined(OS_WIN) || defined(OS_MACOSX)
-// media::GetAudioOutputHardwareSampleRate() asks the audio layer
-// for its current sample rate (set by the user) on Windows and Mac OS X.
-// The listed rates below adds restrictions and Initialize()
-// will fail if the user selects any rate outside these ranges.
+// AudioHardwareConfig::GetOutputSampleRate() asks the audio layer for its
+// current sample rate (set by the user) on Windows and Mac OS X.  The listed
+// rates below adds restrictions and Initialize() will fail if the user selects
+// any rate outside these ranges.
 int kValidOutputRates[] = {96000, 48000, 44100};
 #elif defined(OS_LINUX) || defined(OS_OPENBSD)
 int kValidOutputRates[] = {48000, 44100};
@@ -104,9 +106,10 @@ bool WebRtcAudioRenderer::Initialize(WebRtcAudioRendererSource* source) {
   sink_ = AudioDeviceFactory::NewOutputDevice();
   DCHECK(sink_);
 
-  // Ask the browser for the default audio output hardware sample-rate.
-  // This request is based on a synchronous IPC message.
-  int sample_rate = GetAudioOutputSampleRate();
+  // Ask the renderer for the default audio output hardware sample-rate.
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
   DVLOG(1) << "Audio output hardware sample rate: " << sample_rate;
   UMA_HISTOGRAM_ENUMERATION("WebRTC.AudioOutputSampleRate",
                             sample_rate, media::kUnexpectedAudioSampleRate);
diff --git a/content/renderer/pepper/pepper_platform_audio_output_impl.cc b/content/renderer/pepper/pepper_platform_audio_output_impl.cc
index 6681d0a..2a6dbf6 100644
--- a/content/renderer/pepper/pepper_platform_audio_output_impl.cc
+++ b/content/renderer/pepper/pepper_platform_audio_output_impl.cc
@@ -11,9 +11,9 @@
 #include "build/build_config.h"
 #include "content/common/child_process.h"
 #include "content/common/media/audio_messages.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/render_thread_impl.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/media_switches.h"
 
 namespace content {
@@ -125,14 +125,17 @@ bool PepperPlatformAudioOutputImpl::Initialize(
   media::AudioParameters::Format format;
   const int kMaxFramesForLowLatency = 2047;
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+
   const CommandLine* cmd_line = CommandLine::ForCurrentProcess();
   if (!cmd_line->HasSwitch(switches::kDisableAudioOutputResampler)) {
     // Rely on AudioOutputResampler to handle any inconsistencies between the
     // hardware params required for low latency and the requested params.
     format = media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
-  } else if (sample_rate == GetAudioOutputSampleRate() &&
+  } else if (sample_rate == hardware_config->GetOutputSampleRate() &&
              frames_per_buffer <= kMaxFramesForLowLatency &&
-             frames_per_buffer % content::GetAudioOutputBufferSize() == 0) {
+             frames_per_buffer % hardware_config->GetOutputBufferSize() == 0) {
     // Use the low latency back end if the client request is compatible, and
     // the sample count is low enough to justify using AUDIO_PCM_LOW_LATENCY.
     format = media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
diff --git a/content/renderer/pepper/pepper_plugin_delegate_impl.cc b/content/renderer/pepper/pepper_plugin_delegate_impl.cc
index 8c6eb0d..5fdd139 100644
--- a/content/renderer/pepper/pepper_plugin_delegate_impl.cc
+++ b/content/renderer/pepper/pepper_plugin_delegate_impl.cc
@@ -35,7 +35,6 @@
 #include "content/public/renderer/content_renderer_client.h"
 #include "content/public/renderer/renderer_restrict_dispatch_group.h"
 #include "content/renderer/gamepad_shared_memory_reader.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
 #include "content/renderer/media/pepper_platform_video_decoder_impl.h"
 #include "content/renderer/p2p/socket_dispatcher.h"
@@ -57,6 +56,7 @@
 #include "content/renderer/webplugin_delegate_proxy.h"
 #include "googleurl/src/gurl.h"
 #include "ipc/ipc_channel_handle.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/video/capture/video_capture_proxy.h"
 #include "ppapi/c/dev/pp_video_dev.h"
 #include "ppapi/c/pp_errors.h"
@@ -864,11 +864,13 @@ void PepperPluginDelegateImpl::SelectedFindResultChanged(int identifier,
 }
 
 uint32_t PepperPluginDelegateImpl::GetAudioHardwareOutputSampleRate() {
-  return static_cast<uint32_t>(GetAudioOutputSampleRate());
+  RenderThreadImpl* thread = RenderThreadImpl::current();
+  return thread->GetAudioHardwareConfig()->GetOutputSampleRate();
 }
 
 uint32_t PepperPluginDelegateImpl::GetAudioHardwareOutputBufferSize() {
-  return static_cast<uint32_t>(GetAudioOutputBufferSize());
+  RenderThreadImpl* thread = RenderThreadImpl::current();
+  return thread->GetAudioHardwareConfig()->GetOutputBufferSize();
 }
 
 webkit::ppapi::PluginDelegate::PlatformAudioOutput*
diff --git a/content/renderer/render_thread_impl.cc b/content/renderer/render_thread_impl.cc
index 5cc42cd..7dd13c2 100644
--- a/content/renderer/render_thread_impl.cc
+++ b/content/renderer/render_thread_impl.cc
@@ -55,7 +55,6 @@
 #include "content/renderer/gpu/compositor_thread.h"
 #include "content/renderer/gpu/compositor_output_surface.h"
 #include "content/renderer/gpu/gpu_benchmarking_extension.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/audio_input_message_filter.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/audio_renderer_mixer_manager.h"
@@ -73,6 +72,7 @@
 #include "ipc/ipc_channel_handle.h"
 #include "ipc/ipc_forwarding_message_filter.h"
 #include "ipc/ipc_platform_file.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/media.h"
 #include "media/base/media_switches.h"
 #include "net/base/net_errors.h"
@@ -904,13 +904,32 @@ RenderThreadImpl::GetGpuVDAContext3D() {
 AudioRendererMixerManager* RenderThreadImpl::GetAudioRendererMixerManager() {
   if (!audio_renderer_mixer_manager_.get()) {
     audio_renderer_mixer_manager_.reset(new AudioRendererMixerManager(
-        GetAudioOutputSampleRate(),
-        GetAudioOutputBufferSize()));
+        GetAudioHardwareConfig()));
   }
 
   return audio_renderer_mixer_manager_.get();
 }
 
+media::AudioHardwareConfig* RenderThreadImpl::GetAudioHardwareConfig() {
+  if (!audio_hardware_config_) {
+    int output_buffer_size;
+    int output_sample_rate;
+    int input_sample_rate;
+    media::ChannelLayout input_channel_layout;
+
+    Send(new ViewHostMsg_GetAudioHardwareConfig(
+        &output_buffer_size, &output_sample_rate,
+        &input_sample_rate, &input_channel_layout));
+
+    audio_hardware_config_.reset(new media::AudioHardwareConfig(
+        output_buffer_size, output_sample_rate, input_sample_rate,
+        input_channel_layout));
+    audio_message_filter_->SetAudioHardwareConfig(audio_hardware_config_.get());
+  }
+
+  return audio_hardware_config_.get();
+}
+
 #if defined(OS_WIN)
 void RenderThreadImpl::PreCacheFontCharacters(const LOGFONT& log_font,
                                               const string16& str) {
diff --git a/content/renderer/render_thread_impl.h b/content/renderer/render_thread_impl.h
index a70be92..0ca7420 100644
--- a/content/renderer/render_thread_impl.h
+++ b/content/renderer/render_thread_impl.h
@@ -45,6 +45,10 @@ namespace IPC {
 class ForwardingMessageFilter;
 }
 
+namespace media {
+class AudioHardwareConfig;
+}
+
 namespace v8 {
 class Extension;
 }
@@ -246,6 +250,11 @@ class CONTENT_EXPORT RenderThreadImpl : public RenderThread,
   // first call.
   AudioRendererMixerManager* GetAudioRendererMixerManager();
 
+  // AudioHardwareConfig contains audio hardware configuration for
+  // renderer side clients.  Creation requires a synchronous IPC call so it is
+  // lazily created on the first call.
+  media::AudioHardwareConfig* GetAudioHardwareConfig();
+
 #if defined(OS_WIN)
   void PreCacheFontCharacters(const LOGFONT& log_font, const string16& str);
 #endif
@@ -381,6 +390,7 @@ class CONTENT_EXPORT RenderThreadImpl : public RenderThread,
   scoped_ptr<WebGraphicsContext3DCommandBufferImpl> gpu_vda_context3d_;
 
   scoped_ptr<AudioRendererMixerManager> audio_renderer_mixer_manager_;
+  scoped_ptr<media::AudioHardwareConfig> audio_hardware_config_;
 
   HistogramCustomizer histogram_customizer_;
 
diff --git a/content/renderer/renderer_webkitplatformsupport_impl.cc b/content/renderer/renderer_webkitplatformsupport_impl.cc
index 4b05c99..685d642 100644
--- a/content/renderer/renderer_webkitplatformsupport_impl.cc
+++ b/content/renderer/renderer_webkitplatformsupport_impl.cc
@@ -27,7 +27,6 @@
 #include "content/renderer/dom_storage/webstoragenamespace_impl.h"
 #include "content/renderer/gamepad_shared_memory_reader.h"
 #include "content/renderer/hyphenator/hyphenator.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/renderer_webaudiodevice_impl.h"
 #include "content/renderer/render_thread_impl.h"
@@ -36,6 +35,7 @@
 #include "googleurl/src/gurl.h"
 #include "ipc/ipc_sync_message_filter.h"
 #include "media/audio/audio_output_device.h"
+#include "media/base/audio_hardware_config.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebBlobRegistry.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebGamepads.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamCenter.h"
@@ -555,11 +555,13 @@ bool RendererWebKitPlatformSupportImpl::canAccelerate2dCanvas() {
 }
 
 double RendererWebKitPlatformSupportImpl::audioHardwareSampleRate() {
-  return GetAudioOutputSampleRate();
+  RenderThreadImpl* thread = RenderThreadImpl::current();
+  return thread->GetAudioHardwareConfig()->GetOutputSampleRate();
 }
 
 size_t RendererWebKitPlatformSupportImpl::audioHardwareBufferSize() {
-  return GetAudioOutputBufferSize();
+  RenderThreadImpl* thread = RenderThreadImpl::current();
+  return thread->GetAudioHardwareConfig()->GetOutputBufferSize();
 }
 
 // TODO(crogers): remove deprecated API as soon as WebKit calls new API.
diff --git a/content/test/webrtc_audio_device_test.cc b/content/test/webrtc_audio_device_test.cc
index 5191f9d..571c4a6 100644
--- a/content/test/webrtc_audio_device_test.cc
+++ b/content/test/webrtc_audio_device_test.cc
@@ -21,13 +21,13 @@
 #include "content/public/common/content_paths.h"
 #include "content/public/test/mock_resource_context.h"
 #include "content/public/test/test_browser_thread.h"
-#include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/audio_input_message_filter.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/renderer/renderer_webkitplatformsupport_impl.h"
+#include "media/base/audio_hardware_config.h"
 #include "net/url_request/url_request_test_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
@@ -115,7 +115,7 @@ ACTION_P(QuitMessageLoop, loop_or_proxy) {
 }
 
 WebRTCAudioDeviceTest::WebRTCAudioDeviceTest()
-    : render_thread_(NULL), audio_util_callback_(NULL),
+    : render_thread_(NULL), audio_hardware_config_(NULL),
       has_input_devices_(false), has_output_devices_(false) {
 }
 
@@ -147,7 +147,7 @@ void WebRTCAudioDeviceTest::SetUp() {
 }
 
 void WebRTCAudioDeviceTest::TearDown() {
-  SetAudioUtilCallback(NULL);
+  SetAudioHardwareConfig(NULL);
 
   // Run any pending cleanup tasks that may have been posted to the main thread.
   ChildProcess::current()->main_thread()->message_loop()->RunUntilIdle();
@@ -183,11 +183,9 @@ bool WebRTCAudioDeviceTest::Send(IPC::Message* message) {
   return channel_->Send(message);
 }
 
-void WebRTCAudioDeviceTest::SetAudioUtilCallback(AudioUtilInterface* callback) {
-  // Invalidate any potentially cached values since the new callback should
-  // be used for those queries.
-  ResetAudioCache();
-  audio_util_callback_ = callback;
+void WebRTCAudioDeviceTest::SetAudioHardwareConfig(
+    media::AudioHardwareConfig* hardware_config) {
+  audio_hardware_config_ = hardware_config;
 }
 
 void WebRTCAudioDeviceTest::InitializeIOThread(const char* thread_name) {
@@ -261,25 +259,17 @@ void WebRTCAudioDeviceTest::DestroyChannel() {
   audio_input_renderer_host_ = NULL;
 }
 
-void WebRTCAudioDeviceTest::OnGetHardwareSampleRate(int* sample_rate) {
-  EXPECT_TRUE(audio_util_callback_);
-  *sample_rate = audio_util_callback_ ?
-      audio_util_callback_->GetAudioHardwareSampleRate() : 0;
-}
+void WebRTCAudioDeviceTest::OnGetAudioHardwareConfig(
+    int* output_buffer_size, int* output_sample_rate, int* input_sample_rate,
+    media::ChannelLayout* input_channel_layout) {
+  ASSERT_TRUE(audio_hardware_config_);
 
-void WebRTCAudioDeviceTest::OnGetHardwareInputSampleRate(int* sample_rate) {
-  EXPECT_TRUE(audio_util_callback_);
-  *sample_rate = audio_util_callback_ ?
-      audio_util_callback_->GetAudioInputHardwareSampleRate(
-          media::AudioManagerBase::kDefaultDeviceId) : 0;
-}
+  *output_buffer_size = audio_hardware_config_->GetOutputBufferSize();
+  *output_sample_rate = audio_hardware_config_->GetOutputSampleRate();
 
-void WebRTCAudioDeviceTest::OnGetHardwareInputChannelLayout(
-    media::ChannelLayout* layout) {
-  EXPECT_TRUE(audio_util_callback_);
-  *layout = !audio_util_callback_ ? media::CHANNEL_LAYOUT_NONE :
-      audio_util_callback_->GetAudioInputHardwareChannelLayout(
-          media::AudioManagerBase::kDefaultDeviceId);
+  // TODO(henrika): add support for all available input devices.
+  *input_sample_rate = audio_hardware_config_->GetInputSampleRate();
+  *input_channel_layout = audio_hardware_config_->GetInputChannelLayout();
 }
 
 // IPC::Listener implementation.
@@ -310,12 +300,8 @@ bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
   bool handled ALLOW_UNUSED = true;
   bool message_is_ok = true;
   IPC_BEGIN_MESSAGE_MAP_EX(WebRTCAudioDeviceTest, message, message_is_ok)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareSampleRate,
-                        OnGetHardwareSampleRate)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputSampleRate,
-                        OnGetHardwareInputSampleRate)
-    IPC_MESSAGE_HANDLER(ViewHostMsg_GetHardwareInputChannelLayout,
-                        OnGetHardwareInputChannelLayout)
+    IPC_MESSAGE_HANDLER(ViewHostMsg_GetAudioHardwareConfig,
+                        OnGetAudioHardwareConfig)
     IPC_MESSAGE_UNHANDLED(handled = false)
   IPC_END_MESSAGE_MAP_EX()
 
diff --git a/content/test/webrtc_audio_device_test.h b/content/test/webrtc_audio_device_test.h
index b2974c4..2e52e94b 100644
--- a/content/test/webrtc_audio_device_test.h
+++ b/content/test/webrtc_audio_device_test.h
@@ -14,6 +14,7 @@
 #include "content/browser/renderer_host/media/mock_media_observer.h"
 #include "content/public/renderer/content_renderer_client.h"
 #include "ipc/ipc_listener.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/channel_layout.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/common_types.h"
@@ -109,18 +110,6 @@ class WebRTCAutoDelete {
   T* ptr_;
 };
 
-// Individual tests can provide an implementation (or mock) of this interface
-// when the audio code queries for hardware capabilities on the IO thread.
-class AudioUtilInterface {
- public:
-  virtual ~AudioUtilInterface() {}
-  virtual int GetAudioHardwareSampleRate() = 0;
-  virtual int GetAudioInputHardwareSampleRate(
-      const std::string& device_id) = 0;
-  virtual media::ChannelLayout GetAudioInputHardwareChannelLayout(
-      const std::string& device_id) = 0;
-};
-
 // Implemented and defined in the cc file.
 class ReplaceContentClientRenderer;
 
@@ -135,7 +124,7 @@ class WebRTCAudioDeviceTest : public ::testing::Test, public IPC::Listener {
   // Sends an IPC message to the IO thread channel.
   bool Send(IPC::Message* message);
 
-  void SetAudioUtilCallback(AudioUtilInterface* callback);
+  void SetAudioHardwareConfig(media::AudioHardwareConfig* hardware_config);
 
  protected:
   void InitializeIOThread(const char* thread_name);
@@ -143,9 +132,9 @@ class WebRTCAudioDeviceTest : public ::testing::Test, public IPC::Listener {
   void CreateChannel(const char* name);
   void DestroyChannel();
 
-  void OnGetHardwareSampleRate(int* sample_rate);
-  void OnGetHardwareInputSampleRate(int* sample_rate);
-  void OnGetHardwareInputChannelLayout(media::ChannelLayout* channels);
+  void OnGetAudioHardwareConfig(int* output_buffer_size,
+                                int* output_sample_rate, int* input_sample_rate,
+                                media::ChannelLayout* input_channel_layout);
 
   // IPC::Listener implementation.
   virtual bool OnMessageReceived(const IPC::Message& message) OVERRIDE;
@@ -177,7 +166,7 @@ class WebRTCAudioDeviceTest : public ::testing::Test, public IPC::Listener {
   scoped_refptr<AudioRendererHost> audio_render_host_;
   scoped_refptr<AudioInputRendererHost> audio_input_renderer_host_;
 
-  AudioUtilInterface* audio_util_callback_;  // Weak reference.
+  media::AudioHardwareConfig* audio_hardware_config_;  // Weak reference.
 
   // Initialized on the main test thread that we mark as the UI thread.
   scoped_ptr<TestBrowserThread> ui_thread_;
diff --git a/media/audio/audio_output_controller.cc b/media/audio/audio_output_controller.cc
index f48cf7f..9990389 100644
--- a/media/audio/audio_output_controller.cc
+++ b/media/audio/audio_output_controller.cc
@@ -10,6 +10,7 @@
 #include "base/threading/platform_thread.h"
 #include "base/time.h"
 #include "build/build_config.h"
+#include "media/audio/audio_util.h"
 #include "media/audio/shared_memory_util.h"
 
 using base::Time;
@@ -323,6 +324,10 @@ void AudioOutputController::DoStopCloseAndClearStream() {
 void AudioOutputController::OnDeviceChange() {
   DCHECK(message_loop_->BelongsToCurrentThread());
 
+  // TODO(dalecurtis): Notify the renderer side that a device change has
+  // occurred.  Currently querying the hardware information here will lead to
+  // crashes on OSX.  See http://crbug.com/158170.
+
   // Recreate the stream (DoCreate() will first shut down an existing stream).
   // Exit if we ran into an error.
   const State original_state = state_;
diff --git a/media/audio/audio_output_controller.h b/media/audio/audio_output_controller.h
index be6d68c..e8fae22 100644
--- a/media/audio/audio_output_controller.h
+++ b/media/audio/audio_output_controller.h
@@ -71,6 +71,8 @@ class MEDIA_EXPORT AudioOutputController
     virtual void OnPlaying(AudioOutputController* controller) = 0;
     virtual void OnPaused(AudioOutputController* controller) = 0;
     virtual void OnError(AudioOutputController* controller, int error_code) = 0;
+    virtual void OnDeviceChange(AudioOutputController* controller,
+                                int new_buffer_size, int new_sample_rate) = 0;
 
    protected:
     virtual ~EventHandler() {}
diff --git a/media/audio/audio_output_controller_unittest.cc b/media/audio/audio_output_controller_unittest.cc
index 746bc45..944d661 100644
--- a/media/audio/audio_output_controller_unittest.cc
+++ b/media/audio/audio_output_controller_unittest.cc
@@ -43,7 +43,8 @@ class MockAudioOutputControllerEventHandler
   MOCK_METHOD1(OnPaused, void(AudioOutputController* controller));
   MOCK_METHOD2(OnError, void(AudioOutputController* controller,
                              int error_code));
-
+  MOCK_METHOD3(OnDeviceChange, void(AudioOutputController* controller,
+                                    int new_buffer_size, int new_sample_rate));
  private:
   DISALLOW_COPY_AND_ASSIGN(MockAudioOutputControllerEventHandler);
 };
diff --git a/media/base/audio_hardware_config.cc b/media/base/audio_hardware_config.cc
new file mode 100644
index 0000000..eaacc69
--- /dev/null
+++ b/media/base/audio_hardware_config.cc
@@ -0,0 +1,53 @@
+// Copyright (c) 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 "media/base/audio_hardware_config.h"
+
+namespace media {
+
+AudioHardwareConfig::AudioHardwareConfig(
+    int output_buffer_size, int output_sample_rate,
+    int input_sample_rate, ChannelLayout input_channel_layout)
+    : output_buffer_size_(output_buffer_size),
+      output_sample_rate_(output_sample_rate),
+      input_sample_rate_(input_sample_rate),
+      input_channel_layout_(input_channel_layout) {
+}
+
+AudioHardwareConfig::~AudioHardwareConfig() {}
+
+int AudioHardwareConfig::GetOutputBufferSize() {
+  base::AutoLock auto_lock(config_lock_);
+  return output_buffer_size_;
+}
+
+int AudioHardwareConfig::GetOutputSampleRate() {
+  base::AutoLock auto_lock(config_lock_);
+  return output_sample_rate_;
+}
+
+int AudioHardwareConfig::GetInputSampleRate() {
+  base::AutoLock auto_lock(config_lock_);
+  return input_sample_rate_;
+}
+
+ChannelLayout AudioHardwareConfig::GetInputChannelLayout() {
+  base::AutoLock auto_lock(config_lock_);
+  return input_channel_layout_;
+}
+
+void AudioHardwareConfig::UpdateInputConfig(
+    int sample_rate, media::ChannelLayout channel_layout) {
+  base::AutoLock auto_lock(config_lock_);
+  input_sample_rate_ = sample_rate;
+  input_channel_layout_ = channel_layout;
+}
+
+void AudioHardwareConfig::UpdateOutputConfig(int buffer_size, int sample_rate) {
+  base::AutoLock auto_lock(config_lock_);
+  output_buffer_size_ = buffer_size;
+  output_sample_rate_ = sample_rate;
+}
+
+}  // namespace media
diff --git a/media/base/audio_hardware_config.h b/media/base/audio_hardware_config.h
new file mode 100644
index 0000000..e61d9ba
--- /dev/null
+++ b/media/base/audio_hardware_config.h
@@ -0,0 +1,50 @@
+// Copyright (c) 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.
+
+#ifndef MEDIA_BASE_AUDIO_HARDWARE_CONFIG_H_
+#define MEDIA_BASE_AUDIO_HARDWARE_CONFIG_H_
+
+#include "base/compiler_specific.h"
+#include "base/synchronization/lock.h"
+#include "media/base/channel_layout.h"
+#include "media/base/media_export.h"
+
+namespace media {
+
+// Provides thread safe access to the audio hardware configuration.
+class MEDIA_EXPORT AudioHardwareConfig {
+ public:
+  AudioHardwareConfig(int output_buffer_size, int output_sample_rate,
+                      int input_sample_rate,
+                      ChannelLayout input_channel_layout);
+  virtual ~AudioHardwareConfig();
+
+  // Accessors for the currently cached hardware configuration.  Safe to call
+  // from any thread.
+  int GetOutputBufferSize();
+  int GetOutputSampleRate();
+  int GetInputSampleRate();
+  ChannelLayout GetInputChannelLayout();
+
+  // Allows callers to update the cached values for either input or output.  The
+  // values are paired under the assumption that these values will only be set
+  // after an input or output device change respectively.  Safe to call from
+  // any thread.
+  void UpdateInputConfig(int sample_rate, media::ChannelLayout channel_layout);
+  void UpdateOutputConfig(int buffer_size, int sample_rate);
+
+ private:
+  // Cached values; access is protected by |config_lock_|.
+  base::Lock config_lock_;
+  int output_buffer_size_;
+  int output_sample_rate_;
+  int input_sample_rate_;
+  ChannelLayout input_channel_layout_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioHardwareConfig);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_BASE_AUDIO_HARDWARE_CONFIG_H_
diff --git a/media/base/audio_hardware_config_unittest.cc b/media/base/audio_hardware_config_unittest.cc
new file mode 100644
index 0000000..afa2e0d
--- /dev/null
+++ b/media/base/audio_hardware_config_unittest.cc
@@ -0,0 +1,50 @@
+// Copyright (c) 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 "media/base/audio_hardware_config.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+namespace media {
+
+static const int kOutputBufferSize = 2048;
+static const int kOutputSampleRate = 48000;
+static const int kInputSampleRate = 44100;
+static const ChannelLayout kInputChannelLayout = CHANNEL_LAYOUT_STEREO;
+
+TEST(AudioHardwareConfig, Getters) {
+  AudioHardwareConfig fake_config(
+      kOutputBufferSize, kOutputSampleRate, kInputSampleRate,
+      kInputChannelLayout);
+
+  EXPECT_EQ(kOutputBufferSize, fake_config.GetOutputBufferSize());
+  EXPECT_EQ(kOutputSampleRate, fake_config.GetOutputSampleRate());
+  EXPECT_EQ(kInputSampleRate, fake_config.GetInputSampleRate());
+  EXPECT_EQ(kInputChannelLayout, fake_config.GetInputChannelLayout());
+}
+
+TEST(AudioHardwareConfig, Setters) {
+  AudioHardwareConfig fake_config(
+      kOutputBufferSize, kOutputSampleRate, kInputSampleRate,
+      kInputChannelLayout);
+
+  // Verify output parameters.
+  const int kNewOutputBufferSize = kOutputBufferSize * 2;
+  const int kNewOutputSampleRate = kOutputSampleRate * 2;
+  EXPECT_NE(kNewOutputBufferSize, fake_config.GetOutputBufferSize());
+  EXPECT_NE(kNewOutputSampleRate, fake_config.GetOutputSampleRate());
+  fake_config.UpdateOutputConfig(kNewOutputBufferSize, kNewOutputSampleRate);
+  EXPECT_EQ(kNewOutputBufferSize, fake_config.GetOutputBufferSize());
+  EXPECT_EQ(kNewOutputSampleRate, fake_config.GetOutputSampleRate());
+
+  // Verify input parameters.
+  const int kNewInputSampleRate = kInputSampleRate * 2;
+  const ChannelLayout kNewInputChannelLayout = CHANNEL_LAYOUT_MONO;
+  EXPECT_NE(kNewInputSampleRate, fake_config.GetInputSampleRate());
+  EXPECT_NE(kNewInputChannelLayout, fake_config.GetInputChannelLayout());
+  fake_config.UpdateInputConfig(kNewInputSampleRate, kNewInputChannelLayout);
+  EXPECT_EQ(kNewInputSampleRate, fake_config.GetInputSampleRate());
+  EXPECT_EQ(kNewInputChannelLayout, fake_config.GetInputChannelLayout());
+}
+
+}  // namespace content
diff --git a/media/media.gyp b/media/media.gyp
index 82a297a..cc8be41 100644
--- a/media/media.gyp
+++ b/media/media.gyp
@@ -186,6 +186,8 @@
         'base/audio_decoder_config.h',
         'base/audio_fifo.cc',
         'base/audio_fifo.h',
+        'base/audio_hardware_config.cc',
+        'base/audio_hardware_config.h',
         'base/audio_pull_fifo.cc',
         'base/audio_pull_fifo.h',
         'base/audio_renderer.cc',
@@ -771,6 +773,7 @@
         'base/audio_bus_unittest.cc',
         'base/audio_converter_unittest.cc',
         'base/audio_fifo_unittest.cc',
+        'base/audio_hardware_config_unittest.cc',
         'base/audio_pull_fifo_unittest.cc',
         'base/audio_renderer_mixer_input_unittest.cc',
         'base/audio_renderer_mixer_unittest.cc',