Revert 249790 "Remove the unified IO code on the browser."

http://build.chromium.org/p/chromium.chromiumos/builders/ChromiumOS%20%28amd64%29/builds/14117
chromeos-chrome-34.0.1829.0_alpha-r1: ../../../../../../../home/chrome-bot/chrome_root/src/media/audio/linux/audio_manager_linux.cc: In function 'media::AudioManager* media::CreateAudioManager(media::AudioLogFactory*)':
chromeos-chrome-34.0.1829.0_alpha-r1: ../../../../../../../home/chrome-bot/chrome_root/src/media/audio/linux/audio_manager_linux.cc:33:50: error: cannot allocate an object of abstract type 'media::AudioManagerCras'
chromeos-chrome-34.0.1829.0_alpha-r1:      return new AudioManagerCras(audio_log_factory);
chromeos-chrome-34.0.1829.0_alpha-r1:                                                   ^

> Remove the unified IO code on the browser.
> 
> Unified IO is not used any more and it should be removed.
> 
> 
> BUG=337096
> TEST=bots, and nothing breaks.
> 
> Review URL: https://codereview.chromium.org/153623004

TBR=xians@chromium.org

Review URL: https://codereview.chromium.org/136233005

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@249811 0039d316-1c4b-4281-b951-d872f2087c98

9 files changed, 2198 insertions, 8 deletions

diff --git a/media/audio/mac/aggregate_device_manager.cc b/media/audio/mac/aggregate_device_manager.cc
new file mode 100644
index 0000000..c7f3233
--- /dev/null
+++ b/media/audio/mac/aggregate_device_manager.cc
@@ -0,0 +1,371 @@
+// 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 "media/audio/mac/aggregate_device_manager.h"
+
+#include <CoreAudio/AudioHardware.h>
+#include <string>
+
+#include "base/mac/mac_logging.h"
+#include "base/mac/scoped_cftyperef.h"
+#include "media/audio/audio_parameters.h"
+#include "media/audio/mac/audio_manager_mac.h"
+
+using base::ScopedCFTypeRef;
+
+namespace media {
+
+AggregateDeviceManager::AggregateDeviceManager()
+    : plugin_id_(kAudioObjectUnknown),
+      input_device_(kAudioDeviceUnknown),
+      output_device_(kAudioDeviceUnknown),
+      aggregate_device_(kAudioObjectUnknown) {
+}
+
+AggregateDeviceManager::~AggregateDeviceManager() {
+  DestroyAggregateDevice();
+}
+
+AudioDeviceID AggregateDeviceManager::GetDefaultAggregateDevice() {
+  AudioDeviceID current_input_device;
+  AudioDeviceID current_output_device;
+  AudioManagerMac::GetDefaultInputDevice(&current_input_device);
+  AudioManagerMac::GetDefaultOutputDevice(&current_output_device);
+
+  if (AudioManagerMac::HardwareSampleRateForDevice(current_input_device) !=
+      AudioManagerMac::HardwareSampleRateForDevice(current_output_device)) {
+    // TODO(crogers): with some extra work we can make aggregate devices work
+    // if the clock domain is the same but the sample-rate differ.
+    // For now we fallback to the synchronized path.
+    return kAudioDeviceUnknown;
+  }
+
+  // Use a lazily created aggregate device if it's already available
+  // and still appropriate.
+  if (aggregate_device_ != kAudioObjectUnknown) {
+    // TODO(crogers): handle default device changes for synchronized I/O.
+    // For now, we check to make sure the default devices haven't changed
+    // since we lazily created the aggregate device.
+    if (current_input_device == input_device_ &&
+        current_output_device == output_device_)
+      return aggregate_device_;
+
+    // For now, once lazily created don't attempt to create another
+    // aggregate device.
+    return kAudioDeviceUnknown;
+  }
+
+  input_device_ = current_input_device;
+  output_device_ = current_output_device;
+
+  // Only create an aggregrate device if the clock domains match.
+  UInt32 input_clockdomain = GetClockDomain(input_device_);
+  UInt32 output_clockdomain = GetClockDomain(output_device_);
+  DVLOG(1) << "input_clockdomain: " << input_clockdomain;
+  DVLOG(1) << "output_clockdomain: " << output_clockdomain;
+
+  if (input_clockdomain == 0 || input_clockdomain != output_clockdomain)
+    return kAudioDeviceUnknown;
+
+  OSStatus result = CreateAggregateDevice(
+      input_device_,
+      output_device_,
+      &aggregate_device_);
+  if (result != noErr)
+    DestroyAggregateDevice();
+
+  return aggregate_device_;
+}
+
+CFStringRef AggregateDeviceManager::GetDeviceUID(AudioDeviceID id) {
+  static const AudioObjectPropertyAddress kDeviceUIDAddress = {
+    kAudioDevicePropertyDeviceUID,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  // As stated in the CoreAudio header (AudioHardwareBase.h),
+  // the caller is responsible for releasing the device_UID.
+  CFStringRef device_UID;
+  UInt32 size = sizeof(device_UID);
+  OSStatus result = AudioObjectGetPropertyData(
+      id,
+      &kDeviceUIDAddress,
+      0,
+      0,
+      &size,
+      &device_UID);
+
+  return (result == noErr) ? device_UID : NULL;
+}
+
+void AggregateDeviceManager::GetDeviceName(
+    AudioDeviceID id, char* name, UInt32 size) {
+  static const AudioObjectPropertyAddress kDeviceNameAddress = {
+    kAudioDevicePropertyDeviceName,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  OSStatus result = AudioObjectGetPropertyData(
+      id,
+      &kDeviceNameAddress,
+      0,
+      0,
+      &size,
+      name);
+
+  if (result != noErr && size > 0)
+    name[0] = 0;
+}
+
+UInt32 AggregateDeviceManager::GetClockDomain(AudioDeviceID device_id) {
+  static const AudioObjectPropertyAddress kClockDomainAddress = {
+    kAudioDevicePropertyClockDomain,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  UInt32 clockdomain = 0;
+  UInt32 size = sizeof(UInt32);
+  OSStatus result = AudioObjectGetPropertyData(
+      device_id,
+      &kClockDomainAddress,
+      0,
+      0,
+      &size,
+      &clockdomain);
+
+  return (result == noErr) ? clockdomain : 0;
+}
+
+OSStatus AggregateDeviceManager::GetPluginID(AudioObjectID* id) {
+  DCHECK(id);
+
+  // Get the audio hardware plugin.
+  CFStringRef bundle_name = CFSTR("com.apple.audio.CoreAudio");
+
+  AudioValueTranslation plugin_translation;
+  plugin_translation.mInputData = &bundle_name;
+  plugin_translation.mInputDataSize = sizeof(bundle_name);
+  plugin_translation.mOutputData = id;
+  plugin_translation.mOutputDataSize = sizeof(*id);
+
+  static const AudioObjectPropertyAddress kPlugInAddress = {
+    kAudioHardwarePropertyPlugInForBundleID,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  UInt32 size = sizeof(plugin_translation);
+  OSStatus result = AudioObjectGetPropertyData(
+      kAudioObjectSystemObject,
+      &kPlugInAddress,
+      0,
+      0,
+      &size,
+      &plugin_translation);
+
+  DVLOG(1) << "CoreAudio plugin ID: " << *id;
+
+  return result;
+}
+
+CFMutableDictionaryRef
+AggregateDeviceManager::CreateAggregateDeviceDictionary(
+    AudioDeviceID input_id,
+    AudioDeviceID output_id) {
+  CFMutableDictionaryRef aggregate_device_dict = CFDictionaryCreateMutable(
+      NULL,
+      0,
+      &kCFTypeDictionaryKeyCallBacks,
+      &kCFTypeDictionaryValueCallBacks);
+  if (!aggregate_device_dict)
+    return NULL;
+
+  const CFStringRef kAggregateDeviceName =
+      CFSTR("ChromeAggregateAudioDevice");
+  const CFStringRef kAggregateDeviceUID =
+      CFSTR("com.google.chrome.AggregateAudioDevice");
+
+  // Add name and UID of the device to the dictionary.
+  CFDictionaryAddValue(
+      aggregate_device_dict,
+      CFSTR(kAudioAggregateDeviceNameKey),
+      kAggregateDeviceName);
+  CFDictionaryAddValue(
+      aggregate_device_dict,
+      CFSTR(kAudioAggregateDeviceUIDKey),
+      kAggregateDeviceUID);
+
+  // Add a "private aggregate key" to the dictionary.
+  // The 1 value means that the created aggregate device will
+  // only be accessible from the process that created it, and
+  // won't be visible to outside processes.
+  int value = 1;
+  ScopedCFTypeRef<CFNumberRef> aggregate_device_number(CFNumberCreate(
+      NULL,
+      kCFNumberIntType,
+      &value));
+  CFDictionaryAddValue(
+      aggregate_device_dict,
+      CFSTR(kAudioAggregateDeviceIsPrivateKey),
+      aggregate_device_number);
+
+  return aggregate_device_dict;
+}
+
+CFMutableArrayRef
+AggregateDeviceManager::CreateSubDeviceArray(
+    CFStringRef input_device_UID, CFStringRef output_device_UID) {
+  CFMutableArrayRef sub_devices_array = CFArrayCreateMutable(
+      NULL,
+      0,
+      &kCFTypeArrayCallBacks);
+
+  CFArrayAppendValue(sub_devices_array, input_device_UID);
+  CFArrayAppendValue(sub_devices_array, output_device_UID);
+
+  return sub_devices_array;
+}
+
+OSStatus AggregateDeviceManager::CreateAggregateDevice(
+    AudioDeviceID input_id,
+    AudioDeviceID output_id,
+    AudioDeviceID* aggregate_device) {
+  DCHECK(aggregate_device);
+
+  const size_t kMaxDeviceNameLength = 256;
+
+  scoped_ptr<char[]> input_device_name(new char[kMaxDeviceNameLength]);
+  GetDeviceName(
+      input_id,
+      input_device_name.get(),
+      sizeof(input_device_name));
+  DVLOG(1) << "Input device: \n" << input_device_name;
+
+  scoped_ptr<char[]> output_device_name(new char[kMaxDeviceNameLength]);
+  GetDeviceName(
+      output_id,
+      output_device_name.get(),
+      sizeof(output_device_name));
+  DVLOG(1) << "Output device: \n" << output_device_name;
+
+  OSStatus result = GetPluginID(&plugin_id_);
+  if (result != noErr)
+    return result;
+
+  // Create a dictionary for the aggregate device.
+  ScopedCFTypeRef<CFMutableDictionaryRef> aggregate_device_dict(
+      CreateAggregateDeviceDictionary(input_id, output_id));
+  if (!aggregate_device_dict)
+    return -1;
+
+  // Create the aggregate device.
+  static const AudioObjectPropertyAddress kCreateAggregateDeviceAddress = {
+    kAudioPlugInCreateAggregateDevice,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  UInt32 size = sizeof(*aggregate_device);
+  result = AudioObjectGetPropertyData(
+      plugin_id_,
+      &kCreateAggregateDeviceAddress,
+      sizeof(aggregate_device_dict),
+      &aggregate_device_dict,
+      &size,
+      aggregate_device);
+  if (result != noErr) {
+    DLOG(ERROR) << "Error creating aggregate audio device!";
+    return result;
+  }
+
+  // Set the sub-devices for the aggregate device.
+  // In this case we use two: the input and output devices.
+
+  ScopedCFTypeRef<CFStringRef> input_device_UID(GetDeviceUID(input_id));
+  ScopedCFTypeRef<CFStringRef> output_device_UID(GetDeviceUID(output_id));
+  if (!input_device_UID || !output_device_UID) {
+    DLOG(ERROR) << "Error getting audio device UID strings.";
+    return -1;
+  }
+
+  ScopedCFTypeRef<CFMutableArrayRef> sub_devices_array(
+      CreateSubDeviceArray(input_device_UID, output_device_UID));
+  if (sub_devices_array == NULL) {
+    DLOG(ERROR) << "Error creating sub-devices array.";
+    return -1;
+  }
+
+  static const AudioObjectPropertyAddress kSetSubDevicesAddress = {
+    kAudioAggregateDevicePropertyFullSubDeviceList,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  size = sizeof(CFMutableArrayRef);
+  result = AudioObjectSetPropertyData(
+      *aggregate_device,
+      &kSetSubDevicesAddress,
+      0,
+      NULL,
+      size,
+      &sub_devices_array);
+  if (result != noErr) {
+    DLOG(ERROR) << "Error setting aggregate audio device sub-devices!";
+    return result;
+  }
+
+  // Use the input device as the master device.
+  static const AudioObjectPropertyAddress kSetMasterDeviceAddress = {
+    kAudioAggregateDevicePropertyMasterSubDevice,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  size = sizeof(CFStringRef);
+  result = AudioObjectSetPropertyData(
+      *aggregate_device,
+      &kSetMasterDeviceAddress,
+      0,
+      NULL,
+      size,
+      &input_device_UID);
+  if (result != noErr) {
+    DLOG(ERROR) << "Error setting aggregate audio device master device!";
+    return result;
+  }
+
+  DVLOG(1) << "New aggregate device: " << *aggregate_device;
+  return noErr;
+}
+
+void AggregateDeviceManager::DestroyAggregateDevice() {
+  if (aggregate_device_ == kAudioObjectUnknown)
+    return;
+
+  static const AudioObjectPropertyAddress kDestroyAddress = {
+    kAudioPlugInDestroyAggregateDevice,
+    kAudioObjectPropertyScopeGlobal,
+    kAudioObjectPropertyElementMaster
+  };
+
+  UInt32 size = sizeof(aggregate_device_);
+  OSStatus result = AudioObjectGetPropertyData(
+      plugin_id_,
+      &kDestroyAddress,
+      0,
+      NULL,
+      &size,
+      &aggregate_device_);
+  if (result != noErr) {
+    DLOG(ERROR) << "Error destroying aggregate audio device!";
+    return;
+  }
+
+  aggregate_device_ = kAudioObjectUnknown;
+}
+
+}  // namespace media
diff --git a/media/audio/mac/aggregate_device_manager.h b/media/audio/mac/aggregate_device_manager.h
new file mode 100644
index 0000000..7b8b71f
--- /dev/null
+++ b/media/audio/mac/aggregate_device_manager.h
@@ -0,0 +1,58 @@
+// 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.
+
+#ifndef MEDIA_AUDIO_MAC_AGGREGATE_DEVICE_MANAGER_H_
+#define MEDIA_AUDIO_MAC_AGGREGATE_DEVICE_MANAGER_H_
+
+#include <CoreAudio/CoreAudio.h>
+
+#include "base/basictypes.h"
+#include "base/compiler_specific.h"
+#include "media/base/media_export.h"
+
+namespace media {
+
+class MEDIA_EXPORT AggregateDeviceManager {
+ public:
+  AggregateDeviceManager();
+  ~AggregateDeviceManager();
+
+  // Lazily creates an aggregate device based on the default
+  // input and output devices.
+  // It will either return a valid device or kAudioDeviceUnknown
+  // if the default devices are not suitable for aggregate devices.
+  AudioDeviceID GetDefaultAggregateDevice();
+
+ private:
+  // The caller is responsible for releasing the CFStringRef.
+  static CFStringRef GetDeviceUID(AudioDeviceID id);
+
+  static void GetDeviceName(AudioDeviceID id, char* name, UInt32 size);
+  static UInt32 GetClockDomain(AudioDeviceID device_id);
+  static OSStatus GetPluginID(AudioObjectID* id);
+
+  CFMutableDictionaryRef CreateAggregateDeviceDictionary(
+      AudioDeviceID input_id,
+      AudioDeviceID output_id);
+
+  CFMutableArrayRef CreateSubDeviceArray(CFStringRef input_device_UID,
+                                         CFStringRef output_device_UID);
+
+  OSStatus CreateAggregateDevice(AudioDeviceID input_id,
+                                 AudioDeviceID output_id,
+                                 AudioDeviceID* aggregate_device);
+  void DestroyAggregateDevice();
+
+  AudioObjectID plugin_id_;
+  AudioDeviceID input_device_;
+  AudioDeviceID output_device_;
+
+  AudioDeviceID aggregate_device_;
+
+  DISALLOW_COPY_AND_ASSIGN(AggregateDeviceManager);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_MAC_AGGREGATE_DEVICE_MANAGER_H_
diff --git a/media/audio/mac/audio_auhal_mac_unittest.cc b/media/audio/mac/audio_auhal_mac_unittest.cc
index fd0ffff..9babd80 100644
--- a/media/audio/mac/audio_auhal_mac_unittest.cc
+++ b/media/audio/mac/audio_auhal_mac_unittest.cc
@@ -45,7 +45,7 @@ class AUHALStreamTest : public testing::Test {
 
   AudioOutputStream* Create() {
     return manager_->MakeAudioOutputStream(
-        manager_->GetDefaultOutputStreamParameters(), "");
+        manager_->GetDefaultOutputStreamParameters(), "", "");
   }
 
   bool CanRunAudioTests() {
diff --git a/media/audio/mac/audio_manager_mac.cc b/media/audio/mac/audio_manager_mac.cc
index e86e7b48..c08efff 100644
--- a/media/audio/mac/audio_manager_mac.cc
+++ b/media/audio/mac/audio_manager_mac.cc
@@ -20,6 +20,8 @@
 #include "media/audio/mac/audio_input_mac.h"
 #include "media/audio/mac/audio_low_latency_input_mac.h"
 #include "media/audio/mac/audio_low_latency_output_mac.h"
+#include "media/audio/mac/audio_synchronized_mac.h"
+#include "media/audio/mac/audio_unified_mac.h"
 #include "media/base/bind_to_current_loop.h"
 #include "media/base/channel_layout.h"
 #include "media/base/limits.h"
@@ -54,6 +56,16 @@ static bool HasAudioHardware(AudioObjectPropertySelector selector) {
       output_device_id != kAudioObjectUnknown;
 }
 
+// Returns true if the default input device is the same as
+// the default output device.
+bool AudioManagerMac::HasUnifiedDefaultIO() {
+  AudioDeviceID input_id, output_id;
+  if (!GetDefaultInputDevice(&input_id) || !GetDefaultOutputDevice(&output_id))
+    return false;
+
+  return input_id == output_id;
+}
+
 // Retrieves information on audio devices, and prepends the default
 // device to the list if the list is non-empty.
 static void GetAudioDeviceInfo(bool is_input,
@@ -554,18 +566,72 @@ std::string AudioManagerMac::GetAssociatedOutputDeviceID(
 
 AudioOutputStream* AudioManagerMac::MakeLinearOutputStream(
     const AudioParameters& params) {
-  return MakeLowLatencyOutputStream(params, std::string());
+  return MakeLowLatencyOutputStream(params, std::string(), std::string());
 }
 
 AudioOutputStream* AudioManagerMac::MakeLowLatencyOutputStream(
     const AudioParameters& params,
-    const std::string& device_id) {
-  AudioDeviceID device = GetAudioDeviceIdByUId(false, device_id);
-  if (device == kAudioObjectUnknown) {
-    DLOG(ERROR) << "Failed to open output device: " << device_id;
+    const std::string& device_id,
+    const std::string& input_device_id) {
+  // Handle basic output with no input channels.
+  if (params.input_channels() == 0) {
+    AudioDeviceID device = GetAudioDeviceIdByUId(false, device_id);
+    if (device == kAudioObjectUnknown) {
+      DLOG(ERROR) << "Failed to open output device: " << device_id;
+      return NULL;
+    }
+    return new AUHALStream(this, params, device);
+  }
+
+  DLOG_IF(ERROR, !device_id.empty()) << "Not implemented!";
+
+  // TODO(xians): support more than stereo input.
+  if (params.input_channels() != 2) {
+    // WebAudio is currently hard-coded to 2 channels so we should not
+    // see this case.
+    NOTREACHED() << "Only stereo input is currently supported!";
     return NULL;
   }
-  return new AUHALStream(this, params, device);
+
+  AudioDeviceID device = kAudioObjectUnknown;
+  if (HasUnifiedDefaultIO()) {
+    // For I/O, the simplest case is when the default input and output
+    // devices are the same.
+    GetDefaultOutputDevice(&device);
+    VLOG(0) << "UNIFIED: default input and output devices are identical";
+  } else {
+    // Some audio hardware is presented as separate input and output devices
+    // even though they are really the same physical hardware and
+    // share the same "clock domain" at the lowest levels of the driver.
+    // A common of example of this is the "built-in" audio hardware:
+    //     "Built-in Line Input"
+    //     "Built-in Output"
+    // We would like to use an "aggregate" device for these situations, since
+    // CoreAudio will make the most efficient use of the shared "clock domain"
+    // so we get the lowest latency and use fewer threads.
+    device = aggregate_device_manager_.GetDefaultAggregateDevice();
+    if (device != kAudioObjectUnknown)
+      VLOG(0) << "Using AGGREGATE audio device";
+  }
+
+  if (device != kAudioObjectUnknown &&
+      input_device_id == AudioManagerBase::kDefaultDeviceId)
+    return new AUHALStream(this, params, device);
+
+  // Fallback to AudioSynchronizedStream which will handle completely
+  // different and arbitrary combinations of input and output devices
+  // even running at different sample-rates.
+  // kAudioDeviceUnknown translates to "use default" here.
+  // TODO(xians): consider tracking UMA stats on AUHALStream
+  // versus AudioSynchronizedStream.
+  AudioDeviceID audio_device_id = GetAudioDeviceIdByUId(true, input_device_id);
+  if (audio_device_id == kAudioObjectUnknown)
+    return NULL;
+
+  return new AudioSynchronizedStream(this,
+                                     params,
+                                     audio_device_id,
+                                     kAudioDeviceUnknown);
 }
 
 std::string AudioManagerMac::GetDefaultOutputDeviceID() {
diff --git a/media/audio/mac/audio_manager_mac.h b/media/audio/mac/audio_manager_mac.h
index 0d4e05f..641f9d3 100644
--- a/media/audio/mac/audio_manager_mac.h
+++ b/media/audio/mac/audio_manager_mac.h
@@ -11,6 +11,7 @@
 #include "base/basictypes.h"
 #include "base/compiler_specific.h"
 #include "media/audio/audio_manager_base.h"
+#include "media/audio/mac/aggregate_device_manager.h"
 #include "media/audio/mac/audio_device_listener_mac.h"
 
 namespace media {
@@ -39,7 +40,8 @@ class MEDIA_EXPORT AudioManagerMac : public AudioManagerBase {
       const AudioParameters& params) OVERRIDE;
   virtual AudioOutputStream* MakeLowLatencyOutputStream(
       const AudioParameters& params,
-      const std::string& device_id) OVERRIDE;
+      const std::string& device_id,
+      const std::string& input_device_id) OVERRIDE;
   virtual AudioInputStream* MakeLinearInputStream(
       const AudioParameters& params, const std::string& device_id) OVERRIDE;
   virtual AudioInputStream* MakeLowLatencyInputStream(
@@ -78,6 +80,8 @@ class MEDIA_EXPORT AudioManagerMac : public AudioManagerBase {
       const AudioParameters& input_params) OVERRIDE;
 
  private:
+  bool HasUnifiedDefaultIO();
+
   // Helper methods for constructing AudioDeviceListenerMac on the audio thread.
   void CreateDeviceListener();
   void DestroyDeviceListener();
@@ -95,6 +99,8 @@ class MEDIA_EXPORT AudioManagerMac : public AudioManagerBase {
   int current_sample_rate_;
   AudioDeviceID current_output_device_;
 
+  AggregateDeviceManager aggregate_device_manager_;
+
   // Helper class which monitors power events to determine if output streams
   // should defer Start() calls.  Required to workaround an OSX bug.  See
   // http://crbug.com/160920 for more details.
diff --git a/media/audio/mac/audio_synchronized_mac.cc b/media/audio/mac/audio_synchronized_mac.cc
new file mode 100644
index 0000000..a9bc88e
--- /dev/null
+++ b/media/audio/mac/audio_synchronized_mac.cc
@@ -0,0 +1,976 @@
+// 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 "media/audio/mac/audio_synchronized_mac.h"
+
+#include <CoreServices/CoreServices.h>
+#include <algorithm>
+
+#include "base/basictypes.h"
+#include "base/debug/trace_event.h"
+#include "base/logging.h"
+#include "base/mac/mac_logging.h"
+#include "media/audio/mac/audio_manager_mac.h"
+#include "media/base/channel_mixer.h"
+
+namespace media {
+
+static const int kHardwareBufferSize = 128;
+static const int kFifoSize = 16384;
+
+// TODO(crogers): handle the non-stereo case.
+static const int kChannels = 2;
+
+// This value was determined empirically for minimum latency while still
+// guarding against FIFO under-runs.
+static const int kBaseTargetFifoFrames = 256 + 64;
+
+// If the input and output sample-rate don't match, then we need to maintain
+// an additional safety margin due to the callback timing jitter and the
+// varispeed buffering.  This value was empirically tuned.
+static const int kAdditionalTargetFifoFrames = 128;
+
+static void ZeroBufferList(AudioBufferList* buffer_list) {
+  for (size_t i = 0; i < buffer_list->mNumberBuffers; ++i)
+    memset(buffer_list->mBuffers[i].mData,
+           0,
+           buffer_list->mBuffers[i].mDataByteSize);
+}
+
+static void WrapBufferList(AudioBufferList* buffer_list,
+                           AudioBus* bus,
+                           int frames) {
+  DCHECK(buffer_list);
+  DCHECK(bus);
+  int channels = bus->channels();
+  int buffer_list_channels = buffer_list->mNumberBuffers;
+
+  // Copy pointers from AudioBufferList.
+  int source_idx = 0;
+  for (int i = 0; i < channels; ++i) {
+    bus->SetChannelData(
+        i, static_cast<float*>(buffer_list->mBuffers[source_idx].mData));
+
+    // It's ok to pass in a |buffer_list| with fewer channels, in which
+    // case we just duplicate the last channel.
+    if (source_idx < buffer_list_channels - 1)
+      ++source_idx;
+  }
+
+  // Finally set the actual length.
+  bus->set_frames(frames);
+}
+
+AudioSynchronizedStream::AudioSynchronizedStream(
+    AudioManagerMac* manager,
+    const AudioParameters& params,
+    AudioDeviceID input_id,
+    AudioDeviceID output_id)
+    : manager_(manager),
+      params_(params),
+      input_sample_rate_(0),
+      output_sample_rate_(0),
+      input_id_(input_id),
+      output_id_(output_id),
+      input_buffer_list_(NULL),
+      fifo_(kChannels, kFifoSize),
+      target_fifo_frames_(kBaseTargetFifoFrames),
+      average_delta_(0.0),
+      fifo_rate_compensation_(1.0),
+      input_unit_(0),
+      varispeed_unit_(0),
+      output_unit_(0),
+      first_input_time_(-1),
+      is_running_(false),
+      hardware_buffer_size_(kHardwareBufferSize),
+      channels_(kChannels) {
+  VLOG(1) << "AudioSynchronizedStream::AudioSynchronizedStream()";
+}
+
+AudioSynchronizedStream::~AudioSynchronizedStream() {
+  DCHECK(!input_unit_);
+  DCHECK(!output_unit_);
+  DCHECK(!varispeed_unit_);
+}
+
+bool AudioSynchronizedStream::Open() {
+  if (params_.channels() != kChannels) {
+    LOG(ERROR) << "Only stereo output is currently supported.";
+    return false;
+  }
+
+  // Create the input, output, and varispeed AudioUnits.
+  OSStatus result = CreateAudioUnits();
+  if (result != noErr) {
+    LOG(ERROR) << "Cannot create AudioUnits.";
+    return false;
+  }
+
+  result = SetupInput(input_id_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error configuring input AudioUnit.";
+    return false;
+  }
+
+  result = SetupOutput(output_id_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error configuring output AudioUnit.";
+    return false;
+  }
+
+  result = SetupCallbacks();
+  if (result != noErr) {
+    LOG(ERROR) << "Error setting up callbacks on AudioUnits.";
+    return false;
+  }
+
+  result = SetupStreamFormats();
+  if (result != noErr) {
+    LOG(ERROR) << "Error configuring stream formats on AudioUnits.";
+    return false;
+  }
+
+  AllocateInputData();
+
+  // Final initialization of the AudioUnits.
+  result = AudioUnitInitialize(input_unit_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error initializing input AudioUnit.";
+    return false;
+  }
+
+  result = AudioUnitInitialize(output_unit_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error initializing output AudioUnit.";
+    return false;
+  }
+
+  result = AudioUnitInitialize(varispeed_unit_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error initializing varispeed AudioUnit.";
+    return false;
+  }
+
+  if (input_sample_rate_ != output_sample_rate_) {
+    // Add extra safety margin.
+    target_fifo_frames_ += kAdditionalTargetFifoFrames;
+  }
+
+  // Buffer initial silence corresponding to target I/O buffering.
+  fifo_.Clear();
+  scoped_ptr<AudioBus> silence =
+      AudioBus::Create(channels_, target_fifo_frames_);
+  silence->Zero();
+  fifo_.Push(silence.get());
+
+  return true;
+}
+
+void AudioSynchronizedStream::Close() {
+  DCHECK(!is_running_);
+
+  if (input_buffer_list_) {
+    free(input_buffer_list_);
+    input_buffer_list_ = 0;
+    input_bus_.reset(NULL);
+    wrapper_bus_.reset(NULL);
+  }
+
+  if (input_unit_) {
+    AudioUnitUninitialize(input_unit_);
+    CloseComponent(input_unit_);
+  }
+
+  if (output_unit_) {
+    AudioUnitUninitialize(output_unit_);
+    CloseComponent(output_unit_);
+  }
+
+  if (varispeed_unit_) {
+    AudioUnitUninitialize(varispeed_unit_);
+    CloseComponent(varispeed_unit_);
+  }
+
+  input_unit_ = NULL;
+  output_unit_ = NULL;
+  varispeed_unit_ = NULL;
+
+  // Inform the audio manager that we have been closed. This can cause our
+  // destruction.
+  manager_->ReleaseOutputStream(this);
+}
+
+void AudioSynchronizedStream::Start(AudioSourceCallback* callback) {
+  DCHECK(callback);
+  DCHECK(input_unit_);
+  DCHECK(output_unit_);
+  DCHECK(varispeed_unit_);
+
+  if (is_running_ || !input_unit_ || !output_unit_ || !varispeed_unit_)
+    return;
+
+  source_ = callback;
+
+  // Reset state variables each time we Start().
+  fifo_rate_compensation_ = 1.0;
+  average_delta_ = 0.0;
+
+  OSStatus result = noErr;
+
+  if (!is_running_) {
+    first_input_time_ = -1;
+
+    result = AudioOutputUnitStart(input_unit_);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      result = AudioOutputUnitStart(output_unit_);
+      OSSTATUS_DCHECK(result == noErr, result);
+    }
+  }
+
+  is_running_ = true;
+}
+
+void AudioSynchronizedStream::Stop() {
+  OSStatus result = noErr;
+  if (is_running_) {
+    result = AudioOutputUnitStop(input_unit_);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      result = AudioOutputUnitStop(output_unit_);
+      OSSTATUS_DCHECK(result == noErr, result);
+    }
+  }
+
+  if (result == noErr)
+    is_running_ = false;
+}
+
+bool AudioSynchronizedStream::IsRunning() {
+  return is_running_;
+}
+
+// TODO(crogers): implement - or remove from AudioOutputStream.
+void AudioSynchronizedStream::SetVolume(double volume) {}
+void AudioSynchronizedStream::GetVolume(double* volume) {}
+
+OSStatus AudioSynchronizedStream::SetOutputDeviceAsCurrent(
+    AudioDeviceID output_id) {
+  OSStatus result = noErr;
+
+  // Get the default output device if device is unknown.
+  if (output_id == kAudioDeviceUnknown) {
+    AudioObjectPropertyAddress pa;
+    pa.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+    pa.mScope = kAudioObjectPropertyScopeGlobal;
+    pa.mElement = kAudioObjectPropertyElementMaster;
+    UInt32 size = sizeof(output_id);
+
+    result = AudioObjectGetPropertyData(
+        kAudioObjectSystemObject,
+        &pa,
+        0,
+        0,
+        &size,
+        &output_id);
+
+    OSSTATUS_DCHECK(result == noErr, result);
+    if (result != noErr)
+      return result;
+  }
+
+  // Set the render frame size.
+  UInt32 frame_size = hardware_buffer_size_;
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectSetPropertyData(
+      output_id,
+      &pa,
+      0,
+      0,
+      sizeof(frame_size),
+      &frame_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  output_info_.Initialize(output_id, false);
+
+  // Set the Current Device to the Default Output Unit.
+  result = AudioUnitSetProperty(
+      output_unit_,
+      kAudioOutputUnitProperty_CurrentDevice,
+      kAudioUnitScope_Global,
+      0,
+      &output_info_.id_,
+      sizeof(output_info_.id_));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::SetInputDeviceAsCurrent(
+    AudioDeviceID input_id) {
+  OSStatus result = noErr;
+
+  // Get the default input device if device is unknown.
+  if (input_id == kAudioDeviceUnknown) {
+    AudioObjectPropertyAddress pa;
+    pa.mSelector = kAudioHardwarePropertyDefaultInputDevice;
+    pa.mScope = kAudioObjectPropertyScopeGlobal;
+    pa.mElement = kAudioObjectPropertyElementMaster;
+    UInt32 size = sizeof(input_id);
+
+    result = AudioObjectGetPropertyData(
+        kAudioObjectSystemObject,
+        &pa,
+        0,
+        0,
+        &size,
+        &input_id);
+
+    OSSTATUS_DCHECK(result == noErr, result);
+    if (result != noErr)
+      return result;
+  }
+
+  // Set the render frame size.
+  UInt32 frame_size = hardware_buffer_size_;
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectSetPropertyData(
+      input_id,
+      &pa,
+      0,
+      0,
+      sizeof(frame_size),
+      &frame_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  input_info_.Initialize(input_id, true);
+
+  // Set the Current Device to the AUHAL.
+  // This should be done only after I/O has been enabled on the AUHAL.
+  result = AudioUnitSetProperty(
+      input_unit_,
+      kAudioOutputUnitProperty_CurrentDevice,
+      kAudioUnitScope_Global,
+      0,
+      &input_info_.id_,
+      sizeof(input_info_.id_));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::CreateAudioUnits() {
+  // Q: Why do we need a varispeed unit?
+  // A: If the input device and the output device are running at
+  // different sample rates and/or on different clocks, we will need
+  // to compensate to avoid a pitch change and
+  // to avoid buffer under and over runs.
+  ComponentDescription varispeed_desc;
+  varispeed_desc.componentType = kAudioUnitType_FormatConverter;
+  varispeed_desc.componentSubType = kAudioUnitSubType_Varispeed;
+  varispeed_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+  varispeed_desc.componentFlags = 0;
+  varispeed_desc.componentFlagsMask = 0;
+
+  Component varispeed_comp = FindNextComponent(NULL, &varispeed_desc);
+  if (varispeed_comp == NULL)
+    return -1;
+
+  OSStatus result = OpenAComponent(varispeed_comp, &varispeed_unit_);
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Open input AudioUnit.
+  ComponentDescription input_desc;
+  input_desc.componentType = kAudioUnitType_Output;
+  input_desc.componentSubType = kAudioUnitSubType_HALOutput;
+  input_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+  input_desc.componentFlags = 0;
+  input_desc.componentFlagsMask = 0;
+
+  Component input_comp = FindNextComponent(NULL, &input_desc);
+  if (input_comp == NULL)
+    return -1;
+
+  result = OpenAComponent(input_comp, &input_unit_);
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Open output AudioUnit.
+  ComponentDescription output_desc;
+  output_desc.componentType = kAudioUnitType_Output;
+  output_desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+  output_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+  output_desc.componentFlags = 0;
+  output_desc.componentFlagsMask = 0;
+
+  Component output_comp = FindNextComponent(NULL, &output_desc);
+  if (output_comp == NULL)
+    return -1;
+
+  result = OpenAComponent(output_comp, &output_unit_);
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  return noErr;
+}
+
+OSStatus AudioSynchronizedStream::SetupInput(AudioDeviceID input_id) {
+  // The AUHAL used for input needs to be initialized
+  // before anything is done to it.
+  OSStatus result = AudioUnitInitialize(input_unit_);
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // We must enable the Audio Unit (AUHAL) for input and disable output
+  // BEFORE setting the AUHAL's current device.
+  result = EnableIO();
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  result = SetInputDeviceAsCurrent(input_id);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::EnableIO() {
+  // Enable input on the AUHAL.
+  UInt32 enable_io = 1;
+  OSStatus result = AudioUnitSetProperty(
+      input_unit_,
+      kAudioOutputUnitProperty_EnableIO,
+      kAudioUnitScope_Input,
+      1,  // input element
+      &enable_io,
+      sizeof(enable_io));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Disable Output on the AUHAL.
+  enable_io = 0;
+  result = AudioUnitSetProperty(
+      input_unit_,
+      kAudioOutputUnitProperty_EnableIO,
+      kAudioUnitScope_Output,
+      0,  // output element
+      &enable_io,
+      sizeof(enable_io));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::SetupOutput(AudioDeviceID output_id) {
+  OSStatus result = noErr;
+
+  result = SetOutputDeviceAsCurrent(output_id);
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Tell the output unit not to reset timestamps.
+  // Otherwise sample rate changes will cause sync loss.
+  UInt32 start_at_zero = 0;
+  result = AudioUnitSetProperty(
+      output_unit_,
+      kAudioOutputUnitProperty_StartTimestampsAtZero,
+      kAudioUnitScope_Global,
+      0,
+      &start_at_zero,
+      sizeof(start_at_zero));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::SetupCallbacks() {
+  // Set the input callback.
+  AURenderCallbackStruct callback;
+  callback.inputProc = InputProc;
+  callback.inputProcRefCon = this;
+  OSStatus result = AudioUnitSetProperty(
+      input_unit_,
+      kAudioOutputUnitProperty_SetInputCallback,
+      kAudioUnitScope_Global,
+      0,
+      &callback,
+      sizeof(callback));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Set the output callback.
+  callback.inputProc = OutputProc;
+  callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      output_unit_,
+      kAudioUnitProperty_SetRenderCallback,
+      kAudioUnitScope_Input,
+      0,
+      &callback,
+      sizeof(callback));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Set the varispeed callback.
+  callback.inputProc = VarispeedProc;
+  callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      varispeed_unit_,
+      kAudioUnitProperty_SetRenderCallback,
+      kAudioUnitScope_Input,
+      0,
+      &callback,
+      sizeof(callback));
+
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::SetupStreamFormats() {
+  AudioStreamBasicDescription asbd, asbd_dev1_in, asbd_dev2_out;
+
+  // Get the Stream Format (Output client side).
+  UInt32 property_size = sizeof(asbd_dev1_in);
+  OSStatus result = AudioUnitGetProperty(
+      input_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Input,
+      1,
+      &asbd_dev1_in,
+      &property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Get the Stream Format (client side).
+  property_size = sizeof(asbd);
+  result = AudioUnitGetProperty(
+      input_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Output,
+      1,
+      &asbd,
+      &property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Get the Stream Format (Output client side).
+  property_size = sizeof(asbd_dev2_out);
+  result = AudioUnitGetProperty(
+      output_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Output,
+      0,
+      &asbd_dev2_out,
+      &property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Set the format of all the AUs to the input/output devices channel count.
+  // For a simple case, you want to set this to
+  // the lower of count of the channels in the input device vs output device.
+  asbd.mChannelsPerFrame = std::min(asbd_dev1_in.mChannelsPerFrame,
+                                    asbd_dev2_out.mChannelsPerFrame);
+
+  // We must get the sample rate of the input device and set it to the
+  // stream format of AUHAL.
+  Float64 rate = 0;
+  property_size = sizeof(rate);
+
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
+  pa.mScope = kAudioObjectPropertyScopeWildcard;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectGetPropertyData(
+      input_info_.id_,
+      &pa,
+      0,
+      0,
+      &property_size,
+      &rate);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  input_sample_rate_ = rate;
+
+  asbd.mSampleRate = rate;
+  property_size = sizeof(asbd);
+
+  // Set the new formats to the AUs...
+  result = AudioUnitSetProperty(
+      input_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Output,
+      1,
+      &asbd,
+      property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  result = AudioUnitSetProperty(
+      varispeed_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Input,
+      0,
+      &asbd,
+      property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Set the correct sample rate for the output device,
+  // but keep the channel count the same.
+  property_size = sizeof(rate);
+
+  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
+  pa.mScope = kAudioObjectPropertyScopeWildcard;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectGetPropertyData(
+      output_info_.id_,
+      &pa,
+      0,
+      0,
+      &property_size,
+      &rate);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  output_sample_rate_ = rate;
+
+  // The requested sample-rate must match the hardware sample-rate.
+  if (output_sample_rate_ != params_.sample_rate()) {
+    LOG(ERROR) << "Requested sample-rate: " << params_.sample_rate()
+        <<  " must match the hardware sample-rate: " << output_sample_rate_;
+    return kAudioDeviceUnsupportedFormatError;
+  }
+
+  asbd.mSampleRate = rate;
+  property_size = sizeof(asbd);
+
+  // Set the new audio stream formats for the rest of the AUs...
+  result = AudioUnitSetProperty(
+      varispeed_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Output,
+      0,
+      &asbd,
+      property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  result = AudioUnitSetProperty(
+      output_unit_,
+      kAudioUnitProperty_StreamFormat,
+      kAudioUnitScope_Input,
+      0,
+      &asbd,
+      property_size);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  return result;
+}
+
+void AudioSynchronizedStream::AllocateInputData() {
+  // Get the native number of input channels that the hardware supports.
+  int hardware_channels = 0;
+  bool got_hardware_channels = AudioManagerMac::GetDeviceChannels(
+      input_id_, kAudioDevicePropertyScopeInput, &hardware_channels);
+  if (!got_hardware_channels || hardware_channels > 2) {
+    // Only mono and stereo are supported on the input side. When it fails to
+    // get the native channel number or the native channel number is bigger
+    // than 2, we open the device in stereo mode.
+    hardware_channels = 2;
+  }
+
+  // Allocate storage for the AudioBufferList used for the
+  // input data from the input AudioUnit.
+  // We allocate enough space for with one AudioBuffer per channel.
+  size_t malloc_size = offsetof(AudioBufferList, mBuffers[0]) +
+      (sizeof(AudioBuffer) * hardware_channels);
+
+  input_buffer_list_ = static_cast<AudioBufferList*>(malloc(malloc_size));
+  input_buffer_list_->mNumberBuffers = hardware_channels;
+
+  input_bus_ = AudioBus::Create(hardware_channels, hardware_buffer_size_);
+  wrapper_bus_ = AudioBus::CreateWrapper(channels_);
+  if (hardware_channels != params_.input_channels()) {
+    ChannelLayout hardware_channel_layout =
+        GuessChannelLayout(hardware_channels);
+    ChannelLayout requested_channel_layout =
+        GuessChannelLayout(params_.input_channels());
+    channel_mixer_.reset(new ChannelMixer(hardware_channel_layout,
+                                          requested_channel_layout));
+    mixer_bus_ = AudioBus::Create(params_.input_channels(),
+                                  hardware_buffer_size_);
+  }
+
+  // Allocate buffers for AudioBufferList.
+  UInt32 buffer_size_bytes = input_bus_->frames() * sizeof(Float32);
+  for (size_t i = 0; i < input_buffer_list_->mNumberBuffers; ++i) {
+    input_buffer_list_->mBuffers[i].mNumberChannels = 1;
+    input_buffer_list_->mBuffers[i].mDataByteSize = buffer_size_bytes;
+    input_buffer_list_->mBuffers[i].mData = input_bus_->channel(i);
+  }
+}
+
+OSStatus AudioSynchronizedStream::HandleInputCallback(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  TRACE_EVENT0("audio", "AudioSynchronizedStream::HandleInputCallback");
+
+  if (first_input_time_ < 0.0)
+    first_input_time_ = time_stamp->mSampleTime;
+
+  // Get the new audio input data.
+  OSStatus result = AudioUnitRender(
+      input_unit_,
+      io_action_flags,
+      time_stamp,
+      bus_number,
+      number_of_frames,
+      input_buffer_list_);
+
+  // TODO(xians): Add back the DCHECK after synchronize IO supports all
+  // combination of input and output params. See http://issue/246521.
+  if (result != noErr)
+    return result;
+
+  // Buffer input into FIFO.
+  int available_frames = fifo_.max_frames() - fifo_.frames();
+  if (input_bus_->frames() <= available_frames) {
+    if (channel_mixer_) {
+      channel_mixer_->Transform(input_bus_.get(), mixer_bus_.get());
+      fifo_.Push(mixer_bus_.get());
+    } else {
+      fifo_.Push(input_bus_.get());
+    }
+  }
+
+  return result;
+}
+
+OSStatus AudioSynchronizedStream::HandleVarispeedCallback(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  // Create a wrapper bus on the AudioBufferList.
+  WrapBufferList(io_data, wrapper_bus_.get(), number_of_frames);
+
+  if (fifo_.frames() < static_cast<int>(number_of_frames)) {
+    // We don't DCHECK here, since this is a possible run-time condition
+    // if the machine is bogged down.
+    wrapper_bus_->Zero();
+    return noErr;
+  }
+
+  // Read from the FIFO to feed the varispeed.
+  fifo_.Consume(wrapper_bus_.get(), 0, number_of_frames);
+
+  return noErr;
+}
+
+OSStatus AudioSynchronizedStream::HandleOutputCallback(
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  // Input callback hasn't run yet or we've suddenly changed sample-rates
+  // -> silence.
+  if (first_input_time_ < 0.0 ||
+      static_cast<int>(number_of_frames) != params_.frames_per_buffer()) {
+    ZeroBufferList(io_data);
+    return noErr;
+  }
+
+  // Use the varispeed playback rate to offset small discrepancies
+  // in hardware clocks, and also any differences in sample-rate
+  // between input and output devices.
+
+  // Calculate a varispeed rate scalar factor to compensate for drift between
+  // input and output.  We use the actual number of frames still in the FIFO
+  // compared with the ideal value of |target_fifo_frames_|.
+  int delta = fifo_.frames() - target_fifo_frames_;
+
+  // Average |delta| because it can jitter back/forth quite frequently
+  // by +/- the hardware buffer-size *if* the input and output callbacks are
+  // happening at almost exactly the same time.  Also, if the input and output
+  // sample-rates are different then |delta| will jitter quite a bit due to
+  // the rate conversion happening in the varispeed, plus the jittering of
+  // the callbacks.  The average value is what's important here.
+  average_delta_ += (delta - average_delta_) * 0.1;
+
+  // Compute a rate compensation which always attracts us back to the
+  // |target_fifo_frames_| over a period of kCorrectionTimeSeconds.
+  const double kCorrectionTimeSeconds = 0.1;
+  double correction_time_frames = kCorrectionTimeSeconds * output_sample_rate_;
+  fifo_rate_compensation_ =
+      (correction_time_frames + average_delta_) / correction_time_frames;
+
+  // Adjust for FIFO drift.
+  OSStatus result = AudioUnitSetParameter(
+      varispeed_unit_,
+      kVarispeedParam_PlaybackRate,
+      kAudioUnitScope_Global,
+      0,
+      fifo_rate_compensation_,
+      0);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Render to the output using the varispeed.
+  result = AudioUnitRender(
+      varispeed_unit_,
+      io_action_flags,
+      time_stamp,
+      0,
+      number_of_frames,
+      io_data);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+  if (result != noErr)
+    return result;
+
+  // Create a wrapper bus on the AudioBufferList.
+  WrapBufferList(io_data, wrapper_bus_.get(), number_of_frames);
+
+  // Process in-place!
+  source_->OnMoreIOData(wrapper_bus_.get(),
+                        wrapper_bus_.get(),
+                        AudioBuffersState(0, 0));
+
+  return noErr;
+}
+
+OSStatus AudioSynchronizedStream::InputProc(
+    void* user_data,
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  AudioSynchronizedStream* stream =
+      static_cast<AudioSynchronizedStream*>(user_data);
+  DCHECK(stream);
+
+  return stream->HandleInputCallback(
+      io_action_flags,
+      time_stamp,
+      bus_number,
+      number_of_frames,
+      io_data);
+}
+
+OSStatus AudioSynchronizedStream::VarispeedProc(
+    void* user_data,
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  AudioSynchronizedStream* stream =
+      static_cast<AudioSynchronizedStream*>(user_data);
+  DCHECK(stream);
+
+  return stream->HandleVarispeedCallback(
+      io_action_flags,
+      time_stamp,
+      bus_number,
+      number_of_frames,
+      io_data);
+}
+
+OSStatus AudioSynchronizedStream::OutputProc(
+    void* user_data,
+    AudioUnitRenderActionFlags* io_action_flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 number_of_frames,
+    AudioBufferList* io_data) {
+  AudioSynchronizedStream* stream =
+      static_cast<AudioSynchronizedStream*>(user_data);
+  DCHECK(stream);
+
+  return stream->HandleOutputCallback(
+      io_action_flags,
+      time_stamp,
+      bus_number,
+      number_of_frames,
+      io_data);
+}
+
+void AudioSynchronizedStream::AudioDeviceInfo::Initialize(
+    AudioDeviceID id, bool is_input) {
+  id_ = id;
+  is_input_ = is_input;
+  if (id_ == kAudioDeviceUnknown)
+    return;
+
+  UInt32 property_size = sizeof(buffer_size_frames_);
+
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
+  pa.mScope = kAudioObjectPropertyScopeWildcard;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  OSStatus result = AudioObjectGetPropertyData(
+      id_,
+      &pa,
+      0,
+      0,
+      &property_size,
+      &buffer_size_frames_);
+
+  OSSTATUS_DCHECK(result == noErr, result);
+}
+
+}  // namespace media
diff --git a/media/audio/mac/audio_synchronized_mac.h b/media/audio/mac/audio_synchronized_mac.h
new file mode 100644
index 0000000..a6db48e
--- /dev/null
+++ b/media/audio/mac/audio_synchronized_mac.h
@@ -0,0 +1,216 @@
+// 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 MEDIA_AUDIO_MAC_AUDIO_SYNCHRONIZED_MAC_H_
+#define MEDIA_AUDIO_MAC_AUDIO_SYNCHRONIZED_MAC_H_
+
+#include <AudioToolbox/AudioToolbox.h>
+#include <AudioUnit/AudioUnit.h>
+#include <CoreAudio/CoreAudio.h>
+
+#include "base/compiler_specific.h"
+#include "base/synchronization/lock.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_bus.h"
+#include "media/base/audio_fifo.h"
+
+namespace media {
+
+class AudioManagerMac;
+class ChannelMixer;
+
+// AudioSynchronizedStream allows arbitrary combinations of input and output
+// devices running off different clocks and using different drivers, with
+// potentially differing sample-rates.  It implements AudioOutputStream
+// and shuttles its synchronized I/O data using AudioSourceCallback.
+//
+// It is required to first acquire the native sample rate of the selected
+// output device and then use the same rate when creating this object.
+//
+// ............................................................................
+// Theory of Operation:
+//                                       .
+//      INPUT THREAD                     .             OUTPUT THREAD
+//   +-----------------+     +------+    .
+//   | Input AudioUnit | --> |      |    .
+//   +-----------------+     |      |    .
+//                           | FIFO |    .
+//                           |      |        +-----------+
+//                           |      | -----> | Varispeed |
+//                           |      |        +-----------+
+//                           +------+    .         |
+//                                       .         |              +-----------+
+//                                       .    OnMoreIOData() -->  | Output AU |
+//                                       .                        +-----------+
+//
+// The input AudioUnit's InputProc is called on one thread which feeds the
+// FIFO.  The output AudioUnit's OutputProc is called on a second thread
+// which pulls on the varispeed to get the current input data.  The varispeed
+// handles mismatches between input and output sample-rate and also clock drift
+// between the input and output drivers.  The varispeed consumes its data from
+// the FIFO and adjusts its rate dynamically according to the amount
+// of data buffered in the FIFO.  If the FIFO starts getting too much data
+// buffered then the varispeed will speed up slightly to compensate
+// and similarly if the FIFO doesn't have enough data buffered then the
+// varispeed will slow down slightly.
+//
+// Finally, once the input data is available then OnMoreIOData() is called
+// which is given this input, and renders the output which is finally sent
+// to the Output AudioUnit.
+class AudioSynchronizedStream : public AudioOutputStream {
+ public:
+  // The ctor takes all the usual parameters, plus |manager| which is the
+  // the audio manager who is creating this object.
+  AudioSynchronizedStream(AudioManagerMac* manager,
+                          const AudioParameters& params,
+                          AudioDeviceID input_id,
+                          AudioDeviceID output_id);
+
+  virtual ~AudioSynchronizedStream();
+
+  // Implementation of AudioOutputStream.
+  virtual bool Open() OVERRIDE;
+  virtual void Close() OVERRIDE;
+  virtual void Start(AudioSourceCallback* callback) OVERRIDE;
+  virtual void Stop() OVERRIDE;
+
+  virtual void SetVolume(double volume) OVERRIDE;
+  virtual void GetVolume(double* volume) OVERRIDE;
+
+  OSStatus SetInputDeviceAsCurrent(AudioDeviceID input_id);
+  OSStatus SetOutputDeviceAsCurrent(AudioDeviceID output_id);
+  AudioDeviceID GetInputDeviceID() { return input_info_.id_;  }
+  AudioDeviceID GetOutputDeviceID() { return output_info_.id_; }
+
+  bool IsRunning();
+
+ private:
+  // Initialization.
+  OSStatus CreateAudioUnits();
+  OSStatus SetupInput(AudioDeviceID input_id);
+  OSStatus EnableIO();
+  OSStatus SetupOutput(AudioDeviceID output_id);
+  OSStatus SetupCallbacks();
+  OSStatus SetupStreamFormats();
+  void AllocateInputData();
+
+  // Handlers for the AudioUnit callbacks.
+  OSStatus HandleInputCallback(AudioUnitRenderActionFlags* io_action_flags,
+                               const AudioTimeStamp* time_stamp,
+                               UInt32 bus_number,
+                               UInt32 number_of_frames,
+                               AudioBufferList* io_data);
+
+  OSStatus HandleVarispeedCallback(AudioUnitRenderActionFlags* io_action_flags,
+                                   const AudioTimeStamp* time_stamp,
+                                   UInt32 bus_number,
+                                   UInt32 number_of_frames,
+                                   AudioBufferList* io_data);
+
+  OSStatus HandleOutputCallback(AudioUnitRenderActionFlags* io_action_flags,
+                                const AudioTimeStamp* time_stamp,
+                                UInt32 bus_number,
+                                UInt32 number_of_frames,
+                                AudioBufferList* io_data);
+
+  // AudioUnit callbacks.
+  static OSStatus InputProc(void* user_data,
+                            AudioUnitRenderActionFlags* io_action_flags,
+                            const AudioTimeStamp* time_stamp,
+                            UInt32 bus_number,
+                            UInt32 number_of_frames,
+                            AudioBufferList* io_data);
+
+  static OSStatus VarispeedProc(void* user_data,
+                                AudioUnitRenderActionFlags* io_action_flags,
+                                const AudioTimeStamp* time_stamp,
+                                UInt32 bus_number,
+                                UInt32 number_of_frames,
+                                AudioBufferList* io_data);
+
+  static OSStatus OutputProc(void* user_data,
+                             AudioUnitRenderActionFlags* io_action_flags,
+                             const AudioTimeStamp* time_stamp,
+                             UInt32 bus_number,
+                             UInt32 number_of_frames,
+                             AudioBufferList*  io_data);
+
+  // Our creator.
+  AudioManagerMac* manager_;
+
+  // Client parameters.
+  AudioParameters params_;
+
+  double input_sample_rate_;
+  double output_sample_rate_;
+
+  // Pointer to the object that will provide the audio samples.
+  AudioSourceCallback* source_;
+
+  // Values used in Open().
+  AudioDeviceID input_id_;
+  AudioDeviceID output_id_;
+
+  // The input AudioUnit renders its data here.
+  AudioBufferList* input_buffer_list_;
+
+  // Holds the actual data for |input_buffer_list_|.
+  scoped_ptr<AudioBus> input_bus_;
+
+  // Used to overlay AudioBufferLists.
+  scoped_ptr<AudioBus> wrapper_bus_;
+
+  class AudioDeviceInfo {
+   public:
+    AudioDeviceInfo()
+        : id_(kAudioDeviceUnknown),
+          is_input_(false),
+          buffer_size_frames_(0) {}
+    void Initialize(AudioDeviceID inID, bool isInput);
+    bool IsInitialized() const { return id_ != kAudioDeviceUnknown; }
+
+    AudioDeviceID id_;
+    bool is_input_;
+    UInt32 buffer_size_frames_;
+  };
+
+  AudioDeviceInfo input_info_;
+  AudioDeviceInfo output_info_;
+
+  // Used for input to output buffering.
+  AudioFifo fifo_;
+
+  // The optimal number of frames we'd like to keep in the FIFO at all times.
+  int target_fifo_frames_;
+
+  // A running average of the measured delta between actual number of frames
+  // in the FIFO versus |target_fifo_frames_|.
+  double average_delta_;
+
+  // A varispeed rate scalar which is calculated based on FIFO drift.
+  double fifo_rate_compensation_;
+
+  // AudioUnits.
+  AudioUnit input_unit_;
+  AudioUnit varispeed_unit_;
+  AudioUnit output_unit_;
+
+  double first_input_time_;
+
+  bool is_running_;
+  int hardware_buffer_size_;
+  int channels_;
+
+  // Channel mixer used to transform mono to stereo data. It is only created
+  // if the input_hardware_channels is mono.
+  scoped_ptr<ChannelMixer> channel_mixer_;
+  scoped_ptr<AudioBus> mixer_bus_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioSynchronizedStream);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_MAC_AUDIO_SYNCHRONIZED_MAC_H_
diff --git a/media/audio/mac/audio_unified_mac.cc b/media/audio/mac/audio_unified_mac.cc
new file mode 100644
index 0000000..d1dc007
--- /dev/null
+++ b/media/audio/mac/audio_unified_mac.cc
@@ -0,0 +1,397 @@
+// 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 "media/audio/mac/audio_unified_mac.h"
+
+#include <CoreServices/CoreServices.h>
+
+#include "base/basictypes.h"
+#include "base/logging.h"
+#include "base/mac/mac_logging.h"
+#include "media/audio/mac/audio_manager_mac.h"
+
+namespace media {
+
+// TODO(crogers): support more than hard-coded stereo input.
+// Ideally we would like to receive this value as a constructor argument.
+static const int kDefaultInputChannels = 2;
+
+AudioHardwareUnifiedStream::AudioHardwareUnifiedStream(
+    AudioManagerMac* manager, const AudioParameters& params)
+    : manager_(manager),
+      source_(NULL),
+      client_input_channels_(kDefaultInputChannels),
+      volume_(1.0f),
+      input_channels_(0),
+      output_channels_(0),
+      input_channels_per_frame_(0),
+      output_channels_per_frame_(0),
+      io_proc_id_(0),
+      device_(kAudioObjectUnknown),
+      is_playing_(false) {
+  DCHECK(manager_);
+
+  // A frame is one sample across all channels. In interleaved audio the per
+  // frame fields identify the set of n |channels|. In uncompressed audio, a
+  // packet is always one frame.
+  format_.mSampleRate = params.sample_rate();
+  format_.mFormatID = kAudioFormatLinearPCM;
+  format_.mFormatFlags = kLinearPCMFormatFlagIsPacked |
+                         kLinearPCMFormatFlagIsSignedInteger;
+  format_.mBitsPerChannel = params.bits_per_sample();
+  format_.mChannelsPerFrame = params.channels();
+  format_.mFramesPerPacket = 1;
+  format_.mBytesPerPacket = (format_.mBitsPerChannel * params.channels()) / 8;
+  format_.mBytesPerFrame = format_.mBytesPerPacket;
+  format_.mReserved = 0;
+
+  // Calculate the number of sample frames per callback.
+  number_of_frames_ = params.GetBytesPerBuffer() / format_.mBytesPerPacket;
+
+  input_bus_ = AudioBus::Create(client_input_channels_,
+                                params.frames_per_buffer());
+  output_bus_ = AudioBus::Create(params);
+}
+
+AudioHardwareUnifiedStream::~AudioHardwareUnifiedStream() {
+  DCHECK_EQ(device_, kAudioObjectUnknown);
+}
+
+bool AudioHardwareUnifiedStream::Open() {
+  // Obtain the current output device selected by the user.
+  AudioObjectPropertyAddress pa;
+  pa.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  pa.mScope = kAudioObjectPropertyScopeGlobal;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  UInt32 size = sizeof(device_);
+
+  OSStatus result = AudioObjectGetPropertyData(
+      kAudioObjectSystemObject,
+      &pa,
+      0,
+      0,
+      &size,
+      &device_);
+
+  if ((result != kAudioHardwareNoError) || (device_ == kAudioDeviceUnknown)) {
+    LOG(ERROR) << "Cannot open unified AudioDevice.";
+    return false;
+  }
+
+  // The requested sample-rate must match the hardware sample-rate.
+  Float64 sample_rate = 0.0;
+  size = sizeof(sample_rate);
+
+  pa.mSelector = kAudioDevicePropertyNominalSampleRate;
+  pa.mScope = kAudioObjectPropertyScopeWildcard;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  result = AudioObjectGetPropertyData(
+      device_,
+      &pa,
+      0,
+      0,
+      &size,
+      &sample_rate);
+
+  if (result != noErr || sample_rate != format_.mSampleRate) {
+    LOG(ERROR) << "Requested sample-rate: " << format_.mSampleRate
+        <<  " must match the hardware sample-rate: " << sample_rate;
+    return false;
+  }
+
+  // Configure buffer frame size.
+  UInt32 frame_size = number_of_frames_;
+
+  pa.mSelector = kAudioDevicePropertyBufferFrameSize;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+  result = AudioObjectSetPropertyData(
+      device_,
+      &pa,
+      0,
+      0,
+      sizeof(frame_size),
+      &frame_size);
+
+  if (result != noErr) {
+    LOG(ERROR) << "Unable to set input buffer frame size: "  << frame_size;
+    return false;
+  }
+
+  pa.mScope = kAudioDevicePropertyScopeOutput;
+  result = AudioObjectSetPropertyData(
+      device_,
+      &pa,
+      0,
+      0,
+      sizeof(frame_size),
+      &frame_size);
+
+  if (result != noErr) {
+    LOG(ERROR) << "Unable to set output buffer frame size: "  << frame_size;
+    return false;
+  }
+
+  DVLOG(1) << "Sample rate: " << sample_rate;
+  DVLOG(1) << "Frame size: " << frame_size;
+
+  // Determine the number of input and output channels.
+  // We handle both the interleaved and non-interleaved cases.
+
+  // Get input stream configuration.
+  pa.mSelector = kAudioDevicePropertyStreamConfiguration;
+  pa.mScope = kAudioDevicePropertyScopeInput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  result = AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr && size > 0) {
+    // Allocate storage.
+    scoped_ptr<uint8[]> input_list_storage(new uint8[size]);
+    AudioBufferList& input_list =
+        *reinterpret_cast<AudioBufferList*>(input_list_storage.get());
+
+    result = AudioObjectGetPropertyData(
+        device_,
+        &pa,
+        0,
+        0,
+        &size,
+        &input_list);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      // Determine number of input channels.
+      input_channels_per_frame_ = input_list.mNumberBuffers > 0 ?
+          input_list.mBuffers[0].mNumberChannels : 0;
+      if (input_channels_per_frame_ == 1 && input_list.mNumberBuffers > 1) {
+        // Non-interleaved.
+        input_channels_ = input_list.mNumberBuffers;
+      } else {
+        // Interleaved.
+        input_channels_ = input_channels_per_frame_;
+      }
+    }
+  }
+
+  DVLOG(1) << "Input channels: " << input_channels_;
+  DVLOG(1) << "Input channels per frame: " << input_channels_per_frame_;
+
+  // The hardware must have at least the requested input channels.
+  if (result != noErr || client_input_channels_ > input_channels_) {
+    LOG(ERROR) << "AudioDevice does not support requested input channels.";
+    return false;
+  }
+
+  // Get output stream configuration.
+  pa.mSelector = kAudioDevicePropertyStreamConfiguration;
+  pa.mScope = kAudioDevicePropertyScopeOutput;
+  pa.mElement = kAudioObjectPropertyElementMaster;
+
+  result = AudioObjectGetPropertyDataSize(device_, &pa, 0, 0, &size);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr && size > 0) {
+    // Allocate storage.
+    scoped_ptr<uint8[]> output_list_storage(new uint8[size]);
+    AudioBufferList& output_list =
+        *reinterpret_cast<AudioBufferList*>(output_list_storage.get());
+
+    result = AudioObjectGetPropertyData(
+        device_,
+        &pa,
+        0,
+        0,
+        &size,
+        &output_list);
+    OSSTATUS_DCHECK(result == noErr, result);
+
+    if (result == noErr) {
+      // Determine number of output channels.
+      output_channels_per_frame_ = output_list.mBuffers[0].mNumberChannels;
+      if (output_channels_per_frame_ == 1 && output_list.mNumberBuffers > 1) {
+        // Non-interleaved.
+        output_channels_ = output_list.mNumberBuffers;
+      } else {
+        // Interleaved.
+        output_channels_ = output_channels_per_frame_;
+      }
+    }
+  }
+
+  DVLOG(1) << "Output channels: " << output_channels_;
+  DVLOG(1) << "Output channels per frame: " << output_channels_per_frame_;
+
+  // The hardware must have at least the requested output channels.
+  if (result != noErr ||
+      output_channels_ < static_cast<int>(format_.mChannelsPerFrame)) {
+    LOG(ERROR) << "AudioDevice does not support requested output channels.";
+    return false;
+  }
+
+  // Setup the I/O proc.
+  result = AudioDeviceCreateIOProcID(device_, RenderProc, this, &io_proc_id_);
+  if (result != noErr) {
+    LOG(ERROR) << "Error creating IOProc.";
+    return false;
+  }
+
+  return true;
+}
+
+void AudioHardwareUnifiedStream::Close() {
+  DCHECK(!is_playing_);
+
+  OSStatus result = AudioDeviceDestroyIOProcID(device_, io_proc_id_);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  io_proc_id_ = 0;
+  device_ = kAudioObjectUnknown;
+
+  // Inform the audio manager that we have been closed. This can cause our
+  // destruction.
+  manager_->ReleaseOutputStream(this);
+}
+
+void AudioHardwareUnifiedStream::Start(AudioSourceCallback* callback) {
+  DCHECK(callback);
+  DCHECK_NE(device_, kAudioObjectUnknown);
+  DCHECK(!is_playing_);
+  if (device_ == kAudioObjectUnknown || is_playing_)
+    return;
+
+  source_ = callback;
+
+  OSStatus result = AudioDeviceStart(device_, io_proc_id_);
+  OSSTATUS_DCHECK(result == noErr, result);
+
+  if (result == noErr)
+    is_playing_ = true;
+}
+
+void AudioHardwareUnifiedStream::Stop() {
+  if (!is_playing_)
+    return;
+
+  if (device_ != kAudioObjectUnknown) {
+    OSStatus result = AudioDeviceStop(device_, io_proc_id_);
+    OSSTATUS_DCHECK(result == noErr, result);
+  }
+
+  is_playing_ = false;
+  source_ = NULL;
+}
+
+void AudioHardwareUnifiedStream::SetVolume(double volume) {
+  volume_ = static_cast<float>(volume);
+  // TODO(crogers): set volume property
+}
+
+void AudioHardwareUnifiedStream::GetVolume(double* volume) {
+  *volume = volume_;
+}
+
+// Pulls on our provider with optional input, asking it to render output.
+// Note to future hackers of this function: Do not add locks here because this
+// is running on a real-time thread (for low-latency).
+OSStatus AudioHardwareUnifiedStream::Render(
+    AudioDeviceID device,
+    const AudioTimeStamp* now,
+    const AudioBufferList* input_data,
+    const AudioTimeStamp* input_time,
+    AudioBufferList* output_data,
+    const AudioTimeStamp* output_time) {
+  // Convert the input data accounting for possible interleaving.
+  // TODO(crogers): it's better to simply memcpy() if source is already planar.
+  if (input_channels_ >= client_input_channels_) {
+    for (int channel_index = 0; channel_index < client_input_channels_;
+         ++channel_index) {
+      float* source;
+
+      int source_channel_index = channel_index;
+
+      if (input_channels_per_frame_ > 1) {
+        // Interleaved.
+        source = static_cast<float*>(input_data->mBuffers[0].mData) +
+            source_channel_index;
+      } else {
+        // Non-interleaved.
+        source = static_cast<float*>(
+            input_data->mBuffers[source_channel_index].mData);
+      }
+
+      float* p = input_bus_->channel(channel_index);
+      for (int i = 0; i < number_of_frames_; ++i) {
+        p[i] = *source;
+        source += input_channels_per_frame_;
+      }
+    }
+  } else if (input_channels_) {
+    input_bus_->Zero();
+  }
+
+  // Give the client optional input data and have it render the output data.
+  source_->OnMoreIOData(input_bus_.get(),
+                        output_bus_.get(),
+                        AudioBuffersState(0, 0));
+
+  // TODO(crogers): handle final Core Audio 5.1 layout for 5.1 audio.
+
+  // Handle interleaving as necessary.
+  // TODO(crogers): it's better to simply memcpy() if dest is already planar.
+
+  for (int channel_index = 0;
+       channel_index < static_cast<int>(format_.mChannelsPerFrame);
+       ++channel_index) {
+    float* dest;
+
+    int dest_channel_index = channel_index;
+
+    if (output_channels_per_frame_ > 1) {
+      // Interleaved.
+      dest = static_cast<float*>(output_data->mBuffers[0].mData) +
+          dest_channel_index;
+    } else {
+      // Non-interleaved.
+      dest = static_cast<float*>(
+          output_data->mBuffers[dest_channel_index].mData);
+    }
+
+    float* p = output_bus_->channel(channel_index);
+    for (int i = 0; i < number_of_frames_; ++i) {
+      *dest = p[i];
+      dest += output_channels_per_frame_;
+    }
+  }
+
+  return noErr;
+}
+
+OSStatus AudioHardwareUnifiedStream::RenderProc(
+    AudioDeviceID device,
+    const AudioTimeStamp* now,
+    const AudioBufferList* input_data,
+    const AudioTimeStamp* input_time,
+    AudioBufferList* output_data,
+    const AudioTimeStamp* output_time,
+    void* user_data) {
+  AudioHardwareUnifiedStream* audio_output =
+      static_cast<AudioHardwareUnifiedStream*>(user_data);
+  DCHECK(audio_output);
+  if (!audio_output)
+    return -1;
+
+  return audio_output->Render(
+      device,
+      now,
+      input_data,
+      input_time,
+      output_data,
+      output_time);
+}
+
+}  // namespace media
diff --git a/media/audio/mac/audio_unified_mac.h b/media/audio/mac/audio_unified_mac.h
new file mode 100644
index 0000000..ff090e3
--- /dev/null
+++ b/media/audio/mac/audio_unified_mac.h
@@ -0,0 +1,100 @@
+// 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 MEDIA_AUDIO_MAC_AUDIO_UNIFIED_MAC_H_
+#define MEDIA_AUDIO_MAC_AUDIO_UNIFIED_MAC_H_
+
+#include <CoreAudio/CoreAudio.h>
+
+#include "base/memory/scoped_ptr.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_parameters.h"
+
+namespace media {
+
+class AudioManagerMac;
+
+// Implementation of AudioOutputStream for Mac OS X using the
+// CoreAudio AudioHardware API suitable for low-latency unified audio I/O
+// when using devices which support *both* input and output
+// in the same driver.  This is the case with professional
+// USB and Firewire devices.
+//
+// Please note that it's required to first get the native sample-rate of the
+// default output device and use that sample-rate when creating this object.
+class AudioHardwareUnifiedStream : public AudioOutputStream {
+ public:
+  // The ctor takes all the usual parameters, plus |manager| which is the
+  // the audio manager who is creating this object.
+  AudioHardwareUnifiedStream(AudioManagerMac* manager,
+                             const AudioParameters& params);
+  // The dtor is typically called by the AudioManager only and it is usually
+  // triggered by calling AudioOutputStream::Close().
+  virtual ~AudioHardwareUnifiedStream();
+
+  // Implementation of AudioOutputStream.
+  virtual bool Open() OVERRIDE;
+  virtual void Close() OVERRIDE;
+  virtual void Start(AudioSourceCallback* callback) OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual void SetVolume(double volume) OVERRIDE;
+  virtual void GetVolume(double* volume) OVERRIDE;
+
+  int input_channels() const { return input_channels_; }
+  int output_channels() const { return output_channels_; }
+
+ private:
+  OSStatus Render(AudioDeviceID device,
+                  const AudioTimeStamp* now,
+                  const AudioBufferList* input_data,
+                  const AudioTimeStamp* input_time,
+                  AudioBufferList* output_data,
+                  const AudioTimeStamp* output_time);
+
+  static OSStatus RenderProc(AudioDeviceID device,
+                             const AudioTimeStamp* now,
+                             const AudioBufferList* input_data,
+                             const AudioTimeStamp* input_time,
+                             AudioBufferList* output_data,
+                             const AudioTimeStamp* output_time,
+                             void* user_data);
+
+  // Our creator, the audio manager needs to be notified when we close.
+  AudioManagerMac* manager_;
+
+  // Pointer to the object that will provide the audio samples.
+  AudioSourceCallback* source_;
+
+  // Structure that holds the stream format details such as bitrate.
+  AudioStreamBasicDescription format_;
+
+  // Hardware buffer size.
+  int number_of_frames_;
+
+  // Number of audio channels provided to the client via OnMoreIOData().
+  int client_input_channels_;
+
+  // Volume level from 0 to 1.
+  float volume_;
+
+  // Number of input and output channels queried from the hardware.
+  int input_channels_;
+  int output_channels_;
+  int input_channels_per_frame_;
+  int output_channels_per_frame_;
+
+  AudioDeviceIOProcID io_proc_id_;
+  AudioDeviceID device_;
+  bool is_playing_;
+
+  // Intermediate buffers used with call to OnMoreIOData().
+  scoped_ptr<AudioBus> input_bus_;
+  scoped_ptr<AudioBus> output_bus_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioHardwareUnifiedStream);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_MAC_AUDIO_UNIFIED_MAC_H_