clang-format audio_device/mac.

webrtc/modules/audio_device/mac/audio_device_mac.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
 #include "webrtc/modules/audio_device/mac/audio_device_mac.h"
 #include "webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/trace.h"
 
 #include <ApplicationServices/ApplicationServices.h>
-#include <libkern/OSAtomic.h>   // OSAtomicCompareAndSwap()
-#include <mach/mach.h>          // mach_task_self()
-#include <sys/sysctl.h>         // sysctlbyname()
-
-
-
-namespace webrtc
-{
-
-#define WEBRTC_CA_RETURN_ON_ERR(expr)                                   \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-            return -1;                                                  \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_ERR(expr)                                         \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,               \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
-
-#define WEBRTC_CA_LOG_WARN(expr)                                        \
-    do {                                                                \
-        err = expr;                                                     \
-        if (err != noErr) {                                             \
-            logCAMsg(kTraceWarning, kTraceAudioDevice, _id,             \
-                "Error in " #expr, (const char *)&err);                 \
-        }                                                               \
-    } while(0)
-
-enum
-{
-    MaxNumberDevices = 64
-};
-
-void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
-{
-    while (1)
-    {
-        int32_t oldValue = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
-            == true)
-        {
-            return;
-        }
-    }
-}
-
-int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
-{
-    while (1)
-    {
-        int32_t value = *theValue;
-        if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
-        {
-            return value;
-        }
-    }
+#include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
+#include <mach/mach.h>         // mach_task_self()
+#include <sys/sysctl.h>        // sysctlbyname()
+
+namespace webrtc {
+
+#define WEBRTC_CA_RETURN_ON_ERR(expr)                                  \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+      return -1;                                                       \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_ERR(expr)                                        \
+  do {                                                                 \
+    err = expr;                                                        \
+    if (err != noErr) {                                                \
+      logCAMsg(kTraceError, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                   \
+    }                                                                  \
+  } while (0)
+
+#define WEBRTC_CA_LOG_WARN(expr)                                         \
+  do {                                                                   \
+    err = expr;                                                          \
+    if (err != noErr) {                                                  \
+      logCAMsg(kTraceWarning, kTraceAudioDevice, _id, "Error in " #expr, \
+               (const char*) & err);                                     \
+    }                                                                    \
+  } while (0)
+
+enum { MaxNumberDevices = 64 };
+
+void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue) {
+  while (1) {
+    int32_t oldValue = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue) == true) {
+      return;
+    }
+  }
+}
+
+int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue) {
+  while (1) {
+    int32_t value = *theValue;
+    if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true) {
+      return value;
+    }
+  }
 }
 
 // CoreAudio errors are best interpreted as four character strings.
 void AudioDeviceMac::logCAMsg(const TraceLevel level,
                               const TraceModule module,
-                              const int32_t id, const char *msg,
-                              const char *err)
-{
+                              const int32_t id,
+                              const char* msg,
+                              const char* err) {
   RTC_DCHECK(msg != NULL);
   RTC_DCHECK(err != NULL);
 
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
+  WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
 #else
-    // We need to flip the characters in this case.
-    WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
-        + 2, err + 1, err);
+  // We need to flip the characters in this case.
+  WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err + 2,
+               err + 1, err);
 #endif
 }
 
-AudioDeviceMac::AudioDeviceMac(const int32_t id) :
-    _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _stopEventRec(*EventWrapper::Create()),
-    _stopEvent(*EventWrapper::Create()),
-    _id(id),
-    _mixerManager(id),
-    _inputDeviceIndex(0),
-    _outputDeviceIndex(0),
-    _inputDeviceID(kAudioObjectUnknown),
-    _outputDeviceID(kAudioObjectUnknown),
-    _inputDeviceIsSpecified(false),
-    _outputDeviceIsSpecified(false),
-    _recChannels(N_REC_CHANNELS),
-    _playChannels(N_PLAY_CHANNELS),
-    _captureBufData(NULL),
-    _renderBufData(NULL),
-    _playBufType(AudioDeviceModule::kFixedBufferSize),
-    _initialized(false),
-    _isShutDown(false),
-    _recording(false),
-    _playing(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
-    _AGC(false),
-    _renderDeviceIsAlive(1),
-    _captureDeviceIsAlive(1),
-    _twoDevices(true),
-    _doStop(false),
-    _doStopRec(false),
-    _macBookPro(false),
-    _macBookProPanRight(false),
-    _captureLatencyUs(0),
-    _renderLatencyUs(0),
-    _captureDelayUs(0),
-    _renderDelayUs(0),
-    _renderDelayOffsetSamples(0),
-    _playBufDelayFixed(20),
-    _playWarning(0),
-    _playError(0),
-    _recWarning(0),
-    _recError(0),
-    _paCaptureBuffer(NULL),
-    _paRenderBuffer(NULL),
-    _captureBufSizeSamples(0),
-    _renderBufSizeSamples(0),
-    prev_key_state_(),
-    get_mic_volume_counter_ms_(0)
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-
-    RTC_DCHECK(&_stopEvent != NULL);
-    RTC_DCHECK(&_stopEventRec != NULL);
-
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
-    memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
-    memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-}
-
-
-AudioDeviceMac::~AudioDeviceMac()
-{
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
-
-    if (!_isShutDown)
-    {
-        Terminate();
-    }
-
-    RTC_DCHECK(!capture_worker_thread_.get());
-    RTC_DCHECK(!render_worker_thread_.get());
-
-    if (_paRenderBuffer)
-    {
-        delete _paRenderBuffer;
-        _paRenderBuffer = NULL;
-    }
-
-    if (_paCaptureBuffer)
-    {
-        delete _paCaptureBuffer;
-        _paCaptureBuffer = NULL;
-    }
-
-    if (_renderBufData)
-    {
-        delete[] _renderBufData;
-        _renderBufData = NULL;
-    }
-
-    if (_captureBufData)
-    {
-        delete[] _captureBufData;
-        _captureBufData = NULL;
-    }
-
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
-
-    kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_destroy() error: %d", kernErr);
-    }
-
-    delete &_stopEvent;
-    delete &_stopEventRec;
-    delete &_critSect;
+AudioDeviceMac::AudioDeviceMac(const int32_t id)
+    : _ptrAudioBuffer(NULL),
+      _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _stopEventRec(*EventWrapper::Create()),
+      _stopEvent(*EventWrapper::Create()),
+      _id(id),
+      _mixerManager(id),
+      _inputDeviceIndex(0),
+      _outputDeviceIndex(0),
+      _inputDeviceID(kAudioObjectUnknown),
+      _outputDeviceID(kAudioObjectUnknown),
+      _inputDeviceIsSpecified(false),
+      _outputDeviceIsSpecified(false),
+      _recChannels(N_REC_CHANNELS),
+      _playChannels(N_PLAY_CHANNELS),
+      _captureBufData(NULL),
+      _renderBufData(NULL),
+      _playBufType(AudioDeviceModule::kFixedBufferSize),
+      _initialized(false),
+      _isShutDown(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _AGC(false),
+      _renderDeviceIsAlive(1),
+      _captureDeviceIsAlive(1),
+      _twoDevices(true),
+      _doStop(false),
+      _doStopRec(false),
+      _macBookPro(false),
+      _macBookProPanRight(false),
+      _captureLatencyUs(0),
+      _renderLatencyUs(0),
+      _captureDelayUs(0),
+      _renderDelayUs(0),
+      _renderDelayOffsetSamples(0),
+      _playBufDelayFixed(20),
+      _playWarning(0),
+      _playError(0),
+      _recWarning(0),
+      _recError(0),
+      _paCaptureBuffer(NULL),
+      _paRenderBuffer(NULL),
+      _captureBufSizeSamples(0),
+      _renderBufSizeSamples(0),
+      prev_key_state_(),
+      get_mic_volume_counter_ms_(0) {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
+
+  RTC_DCHECK(&_stopEvent != NULL);
+  RTC_DCHECK(&_stopEventRec != NULL);
+
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+  memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
+  memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
+}
+
+AudioDeviceMac::~AudioDeviceMac() {
+  WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
+               __FUNCTION__);
+
+  if (!_isShutDown) {
+    Terminate();
+  }
+
+  RTC_DCHECK(!capture_worker_thread_.get());
+  RTC_DCHECK(!render_worker_thread_.get());
+
+  if (_paRenderBuffer) {
+    delete _paRenderBuffer;
+    _paRenderBuffer = NULL;
+  }
+
+  if (_paCaptureBuffer) {
+    delete _paCaptureBuffer;
+    _paCaptureBuffer = NULL;
+  }
+
+  if (_renderBufData) {
+    delete[] _renderBufData;
+    _renderBufData = NULL;
+  }
+
+  if (_captureBufData) {
+    delete[] _captureBufData;
+    _captureBufData = NULL;
+  }
+
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
+
+  kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_destroy() error: %d", kernErr);
+  }
+
+  delete &_stopEvent;
+  delete &_stopEventRec;
+  delete &_critSect;
 }
 
 // ============================================================================
 //                                     API
 // ============================================================================
 
-void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    _ptrAudioBuffer = audioBuffer;
-
-    // inform the AudioBuffer about default settings for this implementation
+void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  CriticalSectionScoped lock(&_critSect);
+
+  _ptrAudioBuffer = audioBuffer;
+
+  // inform the AudioBuffer about default settings for this implementation
+  _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+  _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
+  _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+}
+
+int32_t AudioDeviceMac::ActiveAudioLayer(
+    AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
+}
+
+int32_t AudioDeviceMac::Init() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_initialized) {
+    return 0;
+  }
+
+  OSStatus err = noErr;
+
+  _isShutDown = false;
+
+  // PortAudio ring buffers require an elementCount which is a power of two.
+  if (_renderBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
+    }
+    _renderBufSizeSamples = powerOfTwo;
+    _renderBufData = new SInt16[_renderBufSizeSamples];
+  }
+
+  if (_paRenderBuffer == NULL) {
+    _paRenderBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize = PaUtil_InitializeRingBuffer(
+        _paRenderBuffer, sizeof(SInt16), _renderBufSizeSamples, _renderBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
+    }
+  }
+
+  if (_captureBufData == NULL) {
+    UInt32 powerOfTwo = 1;
+    while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES) {
+      powerOfTwo <<= 1;
+    }
+    _captureBufSizeSamples = powerOfTwo;
+    _captureBufData = new Float32[_captureBufSizeSamples];
+  }
+
+  if (_paCaptureBuffer == NULL) {
+    _paCaptureBuffer = new PaUtilRingBuffer;
+    PaRingBufferSize bufSize = -1;
+    bufSize =
+        PaUtil_InitializeRingBuffer(_paCaptureBuffer, sizeof(Float32),
+                                    _captureBufSizeSamples, _captureBufData);
+    if (bufSize == -1) {
+      WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                   " PaUtil_InitializeRingBuffer() error");
+      return -1;
+    }
+  }
+
+  kern_return_t kernErr = KERN_SUCCESS;
+  kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
+
+  kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
+                             SYNC_POLICY_FIFO, 0);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
+                 " semaphore_create() error: %d", kernErr);
+    return -1;
+  }
+
+  // Setting RunLoop to NULL here instructs HAL to manage its own thread for
+  // notifications. This was the default behaviour on OS X 10.5 and earlier,
+  // but now must be explicitly specified. HAL would otherwise try to use the
+  // main thread to issue notifications.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyRunLoop, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  CFRunLoopRef runLoop = NULL;
+  UInt32 size = sizeof(CFRunLoopRef);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, size, &runLoop));
+
+  // Listen for any device changes.
+  propertyAddress.mSelector = kAudioHardwarePropertyDevices;
+  WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  // Determine if this is a MacBook Pro
+  _macBookPro = false;
+  _macBookProPanRight = false;
+  char buf[128];
+  size_t length = sizeof(buf);
+  memset(buf, 0, length);
+
+  int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
+  if (intErr != 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Error in sysctlbyname(): %d", err);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Hardware model: %s",
+                 buf);
+    if (strncmp(buf, "MacBookPro", 10) == 0) {
+      _macBookPro = true;
+    }
+  }
+
+  _playWarning = 0;
+  _playError = 0;
+  _recWarning = 0;
+  _recError = 0;
+
+  get_mic_volume_counter_ms_ = 0;
+
+  _initialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::Terminate() {
+  if (!_initialized) {
+    return 0;
+  }
+
+  if (_recording) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording must be stopped");
+    return -1;
+  }
+
+  if (_playing) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Playback must be stopped");
+    return -1;
+  }
+
+  _critSect.Enter();
+
+  _mixerManager.Close();
+
+  OSStatus err = noErr;
+  int retVal = 0;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      kAudioObjectSystemObject, &propertyAddress, &objectListenerProc, this));
+
+  err = AudioHardwareUnload();
+  if (err != noErr) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Error in AudioHardwareUnload()", (const char*)&err);
+    retVal = -1;
+  }
+
+  _isShutDown = true;
+  _initialized = false;
+  _outputDeviceIsSpecified = false;
+  _inputDeviceIsSpecified = false;
+
+  _critSect.Leave();
+
+  return retVal;
+}
+
+bool AudioDeviceMac::Initialized() const {
+  return (_initialized);
+}
+
+int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a valid speaker
+  // exists.
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitSpeaker() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1) {
+    return -1;
+  }
+
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
+
+  if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1) {
+    return -1;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a valid microphone
+  // exists.
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitMicrophone() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1) {
+    return -1;
+  }
+
+  if (_inputDeviceID == _outputDeviceID) {
+    _twoDevices = false;
+  } else {
+    _twoDevices = true;
+  }
+
+  if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1) {
+    return -1;
+  }
+
+  return 0;
+}
+
+bool AudioDeviceMac::SpeakerIsInitialized() const {
+  return (_mixerManager.SpeakerIsInitialized());
+}
+
+bool AudioDeviceMac::MicrophoneIsInitialized() const {
+  return (_mixerManager.MicrophoneIsInitialized());
+}
+
+int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitSpeaker was successful, we know that a volume control exists
+  //
+  available = true;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume) {
+  return (_mixerManager.SetSpeakerVolume(volume));
+}
+
+int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.SpeakerVolume(level) == -1) {
+    return -1;
+  }
+
+  volume = level;
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetWaveOutVolume(uint16_t volumeLeft,
+                                         uint16_t volumeRight) {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
+}
+
+int32_t AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
+                                      uint16_t& /*volumeRight*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+  return -1;
+}
+
+int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxSpeakerVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinSpeakerVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
+
+  if (_mixerManager.SpeakerVolumeStepSize(delta) == -1) {
+    return -1;
+  }
+
+  stepSize = delta;
+  return 0;
+}
+
+int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  // Make an attempt to open up the
+  // output mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // If we end up here it means that the selected speaker has no volume
+    // control, hence it is safe to state that there is no mute control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected speaker has a mute control
+  //
+  _mixerManager.SpeakerMuteIsAvailable(isAvailable);
+
+  available = isAvailable;
+
+  // Close the initialized output mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetSpeakerMute(bool enable) {
+  return (_mixerManager.SetSpeakerMute(enable));
+}
+
+int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.SpeakerMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a mute control
+  //
+  _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetMicrophoneMute(bool enable) {
+  return (_mixerManager.SetMicrophoneMute(enable));
+}
+
+int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const {
+  bool muted(0);
+
+  if (_mixerManager.MicrophoneMute(muted) == -1) {
+    return -1;
+  }
+
+  enabled = muted;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Enumerate all avaliable microphone and make an attempt to open up the
+  // input mixer corresponding to the currently selected input device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control, hence it is safe to state that there is no boost control
+    // already at this stage.
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone has a boost control
+  //
+  _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable) {
+  return (_mixerManager.SetMicrophoneBoost(enable));
+}
+
+int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const {
+  bool onOff(0);
+
+  if (_mixerManager.MicrophoneBoost(onOff) == -1) {
+    return -1;
+  }
+
+  enabled = onOff;
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoRecordingIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetStereoRecording(bool enable) {
+  if (enable)
+    _recChannels = 2;
+  else
+    _recChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoRecording(bool& enabled) const {
+  if (_recChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available) {
+  bool isAvailable(false);
+  bool wasInitialized = _mixerManager.SpeakerIsInitialized();
+
+  if (!wasInitialized && InitSpeaker() == -1) {
+    // Cannot open the specified device
+    available = false;
+    return 0;
+  }
+
+  // Check if the selected microphone can record stereo
+  //
+  _mixerManager.StereoPlayoutIsAvailable(isAvailable);
+  available = isAvailable;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseSpeaker();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetStereoPlayout(bool enable) {
+  if (enable)
+    _playChannels = 2;
+  else
+    _playChannels = 1;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const {
+  if (_playChannels == 2)
+    enabled = true;
+  else
+    enabled = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetAGC(bool enable) {
+  _AGC = enable;
+
+  return 0;
+}
+
+bool AudioDeviceMac::AGC() const {
+  return _AGC;
+}
+
+int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available) {
+  bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
+
+  // Make an attempt to open up the
+  // input mixer corresponding to the currently selected output device.
+  //
+  if (!wasInitialized && InitMicrophone() == -1) {
+    // If we end up here it means that the selected microphone has no volume
+    // control.
+    available = false;
+    return 0;
+  }
+
+  // Given that InitMicrophone was successful, we know that a volume control
+  // exists
+  //
+  available = true;
+
+  // Close the initialized input mixer
+  //
+  if (!wasInitialized) {
+    _mixerManager.CloseMicrophone();
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume) {
+  return (_mixerManager.SetMicrophoneVolume(volume));
+}
+
+int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const {
+  uint32_t level(0);
+
+  if (_mixerManager.MicrophoneVolume(level) == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  failed to retrive current microphone level");
+    return -1;
+  }
+
+  volume = level;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  uint32_t maxVol(0);
+
+  if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1) {
+    return -1;
+  }
+
+  maxVolume = maxVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const {
+  uint32_t minVol(0);
+
+  if (_mixerManager.MinMicrophoneVolume(minVol) == -1) {
+    return -1;
+  }
+
+  minVolume = minVol;
+  return 0;
+}
+
+int32_t AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const {
+  uint16_t delta(0);
+
+  if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1) {
+    return -1;
+  }
+
+  stepSize = delta;
+  return 0;
+}
+
+int16_t AudioDeviceMac::PlayoutDevices() {
+  AudioDeviceID playDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
+                          MaxNumberDevices);
+}
+
+int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index) {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_playIsInitialized) {
+    return -1;
+  }
+
+  AudioDeviceID playDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
+                                       playDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio output devices is %u",
+               nDevices);
+
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
+
+  _outputDeviceIndex = index;
+  _outputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetPlayoutDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(PlayoutDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
+}
+
+int32_t AudioDeviceMac::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  const uint16_t nDevices(RecordingDevices());
+
+  if ((index > (nDevices - 1)) || (name == NULL)) {
+    return -1;
+  }
+
+  memset(name, 0, kAdmMaxDeviceNameSize);
+
+  if (guid != NULL) {
+    memset(guid, 0, kAdmMaxGuidSize);
+  }
+
+  return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
+}
+
+int16_t AudioDeviceMac::RecordingDevices() {
+  AudioDeviceID recDevices[MaxNumberDevices];
+  return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
+                          MaxNumberDevices);
+}
+
+int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index) {
+  if (_recIsInitialized) {
+    return -1;
+  }
+
+  AudioDeviceID recDevices[MaxNumberDevices];
+  uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
+                                       recDevices, MaxNumberDevices);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  number of availiable waveform-audio input devices is %u",
+               nDevices);
+
+  if (index > (nDevices - 1)) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  device index is out of range [0,%u]", (nDevices - 1));
+    return -1;
+  }
+
+  _inputDeviceIndex = index;
+  _inputDeviceIsSpecified = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::SetRecordingDevice(
+    AudioDeviceModule::WindowsDeviceType /*device*/) {
+  WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+               "WindowsDeviceType not supported");
+  return -1;
+}
+
+int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the playout side
+  if (InitPlayout() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
+  // We must actually start playout here in order to have the IOProc
+  // deleted by calling StopPlayout().
+  if (StartPlayout() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopPlayout() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::RecordingIsAvailable(bool& available) {
+  available = true;
+
+  // Try to initialize the recording side
+  if (InitRecording() == -1) {
+    available = false;
+  }
+
+  // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
+  // We must actually start recording here in order to have the IOProc
+  // deleted by calling StopRecording().
+  if (StartRecording() == -1) {
+    available = false;
+  }
+
+  // Cancel effect of initialization
+  if (StopRecording() == -1) {
+    available = false;
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_playing) {
+    return -1;
+  }
+
+  if (!_outputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_playIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the speaker (devices might have been added or removed)
+  if (InitSpeaker() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitSpeaker() failed");
+  }
+
+  if (!MicrophoneIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (MicrophoneIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  MicrophoneIsAvailable() failed");
+    }
+  }
+
+  PaUtil_FlushRingBuffer(_paRenderBuffer);
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  _renderDelayOffsetSamples = 0;
+  _renderDelayUs = 0;
+  _renderLatencyUs = 0;
+  _renderDeviceIsAlive = 1;
+  _doStop = false;
+
+  // The internal microphone of a MacBook Pro is located under the left speaker
+  // grille. When the internal speakers are in use, we want to fully stereo
+  // pan to the right.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  if (_macBookPro) {
+    _macBookProPanRight = false;
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      UInt32 dataSource = 0;
+      size = sizeof(dataSource);
+      WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(
+          _outputDeviceID, &propertyAddress, 0, NULL, &size, &dataSource));
+
+      if (dataSource == 'ispk') {
+        _macBookProPanRight = true;
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro using internal speakers; stereo"
+                     " panning right");
+      } else {
+        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                     "MacBook Pro not using internal speakers");
+      }
+
+      // Add a listener to determine if the status changes.
+      WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+    }
+  }
+
+  // Get current stream description
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
+  size = sizeof(_outStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &_outStreamFormat));
+
+  if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable output stream format -> mFormatID",
+             (const char*)&_outStreamFormat.mFormatID);
+    return -1;
+  }
+
+  if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on output device (mChannelsPerFrame = %d)",
+                 _outStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
+
+  if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Non-interleaved audio data is not supported.",
+                 "AudioHardware streams should not have this format.");
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Ouput stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               _outStreamFormat.mSampleRate,
+               _outStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _outStreamFormat.mBytesPerPacket,
+               _outStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _outStreamFormat.mBytesPerFrame,
+               _outStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               _outStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_outStreamFormat.mFormatID);
+
+  // Our preferred format to work with.
+  if (_outStreamFormat.mChannelsPerFrame < 2) {
+    // Disable stereo playout when we only have one channel on the device.
+    _playChannels = 1;
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 "Stereo playout unavailable on this device");
+  }
+  WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+
+  // Listen for format changes.
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  // Listen for processor overloads.
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_twoDevices || !_recIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _outputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
+
+  _playIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::InitRecording() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (_recording) {
+    return -1;
+  }
+
+  if (!_inputDeviceIsSpecified) {
+    return -1;
+  }
+
+  if (_recIsInitialized) {
+    return 0;
+  }
+
+  // Initialize the microphone (devices might have been added or removed)
+  if (InitMicrophone() == -1) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "  InitMicrophone() failed");
+  }
+
+  if (!SpeakerIsInitialized()) {
+    // Make this call to check if we are using
+    // one or two devices (_twoDevices)
+    bool available = false;
+    if (SpeakerIsAvailable(available) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "  SpeakerIsAvailable() failed");
+    }
+  }
+
+  OSStatus err = noErr;
+  UInt32 size = 0;
+
+  PaUtil_FlushRingBuffer(_paCaptureBuffer);
+
+  _captureDelayUs = 0;
+  _captureLatencyUs = 0;
+  _captureDeviceIsAlive = 1;
+  _doStopRec = false;
+
+  // Get current stream description
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
+  size = sizeof(_inStreamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &_inStreamFormat));
+
+  if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&_inStreamFormat.mFormatID);
+    return -1;
+  }
+
+  if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on input device (mChannelsPerFrame = %d)",
+                 _inStreamFormat.mChannelsPerFrame);
+    return -1;
+  }
+
+  const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
+                                    _inStreamFormat.mSampleRate / 100 *
+                                    N_BLOCKS_IO;
+  if (io_block_size_samples > _captureBufSizeSamples) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Input IO block size (%d) is larger than ring buffer (%u)",
+                 io_block_size_samples, _captureBufSizeSamples);
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mSampleRate = %f, mChannelsPerFrame = %u",
+               _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerPacket = %u, mFramesPerPacket = %u",
+               _inStreamFormat.mBytesPerPacket,
+               _inStreamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               " mBytesPerFrame = %u, mBitsPerChannel = %u",
+               _inStreamFormat.mBytesPerFrame, _inStreamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " mFormatFlags = %u",
+               _inStreamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&_inStreamFormat.mFormatID);
+
+  // Our preferred format to work with
+  if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+    _inDesiredFormat.mChannelsPerFrame = 2;
+  } else {
+    // Disable stereo recording when we only have one channel on the device.
+    _inDesiredFormat.mChannelsPerFrame = 1;
+    _recChannels = 1;
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                 "Stereo recording unavailable on this device");
+  }
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters
     _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-    _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
-    _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-int32_t AudioDeviceMac::ActiveAudioLayer(
-    AudioDeviceModule::AudioLayer& audioLayer) const
-{
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+    _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+  }
+
+  _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
+  _inDesiredFormat.mBytesPerPacket =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mFramesPerPacket = 1;
+  _inDesiredFormat.mBytesPerFrame =
+      _inDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+  _inDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+  _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+#endif
+  _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
+                                            &_captureConverter));
+
+  // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
+  // TODO(xians): investigate this block.
+  UInt32 bufByteCount =
+      (UInt32)((_inStreamFormat.mSampleRate / 1000.0) * 10.0 * N_BLOCKS_IO *
+               _inStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_inStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount =
+          ((UInt32)(bufByteCount / _inStreamFormat.mFramesPerPacket) + 1) *
+          _inStreamFormat.mFramesPerPacket;
+    }
+  }
+
+  // Ensure the buffer size is within the acceptable range provided by the
+  // device.
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
+
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get capture device latency
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs = (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Get capture stream latency
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _inputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _captureLatencyUs +=
+      (UInt32)((1.0e6 * latency) / _inStreamFormat.mSampleRate);
+
+  // Listen for format changes
+  // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
+  propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  // Listen for processor overloads
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, inDeviceIOProc, this, &_inDeviceIOProcID));
+  } else if (!_playIsInitialized) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(
+        _inputDeviceID, deviceIOProc, this, &_deviceIOProcID));
+  }
+
+  // Mark recording side as initialized
+  _recIsInitialized = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StartRecording() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_recIsInitialized) {
+    return -1;
+  }
+
+  if (_recording) {
     return 0;
-}
-
-int32_t AudioDeviceMac::Init()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_initialized)
-    {
-        return 0;
-    }
-
+  }
+
+  if (!_initialized) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Recording worker thread has not been started");
+    return -1;
+  }
+
+  RTC_DCHECK(!capture_worker_thread_.get());
+  capture_worker_thread_.reset(
+      new rtc::PlatformThread(RunCapture, this, "CaptureWorkerThread"));
+  RTC_DCHECK(capture_worker_thread_.get());
+  capture_worker_thread_->Start();
+  capture_worker_thread_->SetPriority(rtc::kRealtimePriority);
+
+  OSStatus err = noErr;
+  if (_twoDevices) {
+    WEBRTC_CA_RETURN_ON_ERR(
+        AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
+  } else if (!_playing) {
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
+  }
+
+  _recording = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StopRecording() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_recIsInitialized) {
+    return 0;
+  }
+
+  OSStatus err = noErr;
+
+  // Stop device
+  int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
+  if (_twoDevices) {
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStopRec = true;  // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEventRec.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the capture IOProc. "
+                     "We may have failed to detect a device removal.");
+
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStopRec = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, " Recording stopped");
+    }
+  } else {
+    // We signal a stop for a shared device even when rendering has
+    // not yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_recording|) before accessing
+    // resources we free below (e.g. the capture converter).
+    //
+    // In the case of a shared devcie, the IOProc will verify
+    // rendering has ended before stopping itself.
+    if (_recording && captureDeviceIsAlive == 1) {
+      _recording = false;
+      _doStop = true;     // Signal to io proc to stop audio device
+      _critSect.Leave();  // Cannot be under lock, risk of deadlock
+      if (kEventTimeout == _stopEvent.Wait(2000)) {
+        CriticalSectionScoped critScoped(&_critSect);
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     " Timed out stopping the shared IOProc. "
+                     "We may have failed to detect a device removal.");
+
+        // We assume rendering on a shared device has stopped as well if
+        // the IOProc times out.
+        WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+      }
+      _critSect.Enter();
+      _doStop = false;
+      WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                   " Recording stopped (shared)");
+    }
+  }
+
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_captureDeviceIsAlive, 0);
+
+  if (capture_worker_thread_.get()) {
+    _critSect.Leave();
+    capture_worker_thread_->Stop();
+    capture_worker_thread_.reset();
+    _critSect.Enter();
+  }
+
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeInput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _inputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  _recIsInitialized = false;
+  _recording = false;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RecordingIsInitialized() const {
+  return (_recIsInitialized);
+}
+
+bool AudioDeviceMac::Recording() const {
+  return (_recording);
+}
+
+bool AudioDeviceMac::PlayoutIsInitialized() const {
+  return (_playIsInitialized);
+}
+
+int32_t AudioDeviceMac::StartPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_playIsInitialized) {
+    return -1;
+  }
+
+  if (_playing) {
+    return 0;
+  }
+
+  RTC_DCHECK(!render_worker_thread_.get());
+  render_worker_thread_.reset(
+      new rtc::PlatformThread(RunRender, this, "RenderWorkerThread"));
+  render_worker_thread_->Start();
+  render_worker_thread_->SetPriority(rtc::kRealtimePriority);
+
+  if (_twoDevices || !_recording) {
     OSStatus err = noErr;
-
-    _isShutDown = false;
-
-    // PortAudio ring buffers require an elementCount which is a power of two.
-    if (_renderBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _renderBufSizeSamples = powerOfTwo;
-        _renderBufData = new SInt16[_renderBufSizeSamples];
-    }
-
-    if (_paRenderBuffer == NULL)
-    {
-        _paRenderBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
-                                              _renderBufSizeSamples,
-                                              _renderBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
-    }
-
-    if (_captureBufData == NULL)
-    {
-        UInt32 powerOfTwo = 1;
-        while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
-        {
-            powerOfTwo <<= 1;
-        }
-        _captureBufSizeSamples = powerOfTwo;
-        _captureBufData = new Float32[_captureBufSizeSamples];
-    }
-
-    if (_paCaptureBuffer == NULL)
-    {
-        _paCaptureBuffer = new PaUtilRingBuffer;
-        PaRingBufferSize bufSize = -1;
-        bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
-                                              sizeof(Float32),
-                                              _captureBufSizeSamples,
-                                              _captureBufData);
-        if (bufSize == -1)
-        {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
-                         _id, " PaUtil_InitializeRingBuffer() error");
-            return -1;
-        }
-    }
-
-    kern_return_t kernErr = KERN_SUCCESS;
-    kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
-
-    kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
-                               SYNC_POLICY_FIFO, 0);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                     " semaphore_create() error: %d", kernErr);
-        return -1;
-    }
-
-    // Setting RunLoop to NULL here instructs HAL to manage its own thread for
-    // notifications. This was the default behaviour on OS X 10.5 and earlier,
-    // but now must be explicitly specified. HAL would otherwise try to use the
-    // main thread to issue notifications.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyRunLoop,
-            kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    CFRunLoopRef runLoop = NULL;
-    UInt32 size = sizeof(CFRunLoopRef);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, size, &runLoop));
-
-    // Listen for any device changes.
-    propertyAddress.mSelector = kAudioHardwarePropertyDevices;
-    WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
-
-    // Determine if this is a MacBook Pro
-    _macBookPro = false;
-    _macBookProPanRight = false;
-    char buf[128];
-    size_t length = sizeof(buf);
-    memset(buf, 0, length);
-
-    int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
-    if (intErr != 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Error in sysctlbyname(): %d", err);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Hardware model: %s", buf);
-        if (strncmp(buf, "MacBookPro", 10) == 0)
-        {
-            _macBookPro = true;
-        }
-    }
-
-    _playWarning = 0;
-    _playError = 0;
-    _recWarning = 0;
-    _recError = 0;
-
-    get_mic_volume_counter_ms_ = 0;
-
-    _initialized = true;
-
+    WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
+  }
+  _playing = true;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::StopPlayout() {
+  CriticalSectionScoped lock(&_critSect);
+
+  if (!_playIsInitialized) {
     return 0;
-}
-
-int32_t AudioDeviceMac::Terminate()
-{
-
-    if (!_initialized)
-    {
-        return 0;
-    }
-
-    if (_recording)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording must be stopped");
-        return -1;
-    }
-
-    if (_playing)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Playback must be stopped");
-        return -1;
-    }
-
+  }
+
+  OSStatus err = noErr;
+
+  int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
+  if (_playing && renderDeviceIsAlive == 1) {
+    // We signal a stop for a shared device even when capturing has not
+    // yet ended. This is to ensure the IOProc will return early as
+    // intended (by checking |_playing|) before accessing resources we
+    // free below (e.g. the render converter).
+    //
+    // In the case of a shared device, the IOProc will verify capturing
+    // has ended before stopping itself.
+    _playing = false;
+    _doStop = true;     // Signal to io proc to stop audio device
+    _critSect.Leave();  // Cannot be under lock, risk of deadlock
+    if (kEventTimeout == _stopEvent.Wait(2000)) {
+      CriticalSectionScoped critScoped(&_critSect);
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " Timed out stopping the render IOProc. "
+                   "We may have failed to detect a device removal.");
+
+      // We assume capturing on a shared device has stopped as well if the
+      // IOProc times out.
+      WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+    }
     _critSect.Enter();
-
-    _mixerManager.Close();
-
-    OSStatus err = noErr;
-    int retVal = 0;
-
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
-            &propertyAddress, &objectListenerProc, this));
-
-    err = AudioHardwareUnload();
-    if (err != noErr)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Error in AudioHardwareUnload()", (const char*) &err);
-        retVal = -1;
-    }
-
-    _isShutDown = true;
-    _initialized = false;
-    _outputDeviceIsSpecified = false;
-    _inputDeviceIsSpecified = false;
-
+    _doStop = false;
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Playout stopped");
+  }
+
+  // Setting this signal will allow the worker thread to be stopped.
+  AtomicSet32(&_renderDeviceIsAlive, 0);
+  if (render_worker_thread_.get()) {
     _critSect.Leave();
-
-    return retVal;
-}
-
-bool AudioDeviceMac::Initialized() const
-{
-    return (_initialized);
-}
-
-int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        available = false;
-        return 0;
-    }
-
-    // Given that InitSpeaker was successful, we know that a valid speaker
-    // exists.
-    available = true;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::InitSpeaker()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playing)
-    {
-        return -1;
-    }
-
-    if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
-    {
-        return -1;
-    }
-
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
-
-    if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        available = false;
-        return 0;
-    }
-
-    // Given that InitMicrophone was successful, we know that a valid microphone
-    // exists.
-    available = true;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::InitMicrophone()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_recording)
-    {
-        return -1;
-    }
-
-    if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
-    {
-        return -1;
-    }
-
-    if (_inputDeviceID == _outputDeviceID)
-    {
-        _twoDevices = false;
-    } else
-    {
-        _twoDevices = true;
-    }
-
-    if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-bool AudioDeviceMac::SpeakerIsInitialized() const
-{
-    return (_mixerManager.SpeakerIsInitialized());
-}
-
-bool AudioDeviceMac::MicrophoneIsInitialized() const
-{
-    return (_mixerManager.MicrophoneIsInitialized());
-}
-
-int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control.
-        available = false;
-        return 0;
-    }
-
-    // Given that InitSpeaker was successful, we know that a volume control exists
-    //
-    available = true;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const
-{
-
-    uint32_t level(0);
-
-    if (_mixerManager.SpeakerVolume(level) == -1)
-    {
-        return -1;
-    }
-
-    volume = level;
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetWaveOutVolume(uint16_t volumeLeft,
-                                         uint16_t volumeRight)
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
-                              uint16_t& /*volumeRight*/) const
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const
-{
-
-    uint32_t maxVol(0);
-
-    if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
-    {
-        return -1;
-    }
-
-    maxVolume = maxVol;
-    return 0;
-}
-
-int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const
-{
-
-    uint32_t minVol(0);
-
-    if (_mixerManager.MinSpeakerVolume(minVol) == -1)
-    {
-        return -1;
-    }
-
-    minVolume = minVol;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
-{
-
-    uint16_t delta(0);
-
-    if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
-
-    stepSize = delta;
-    return 0;
-}
-
-int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    // Make an attempt to open up the
-    // output mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // If we end up here it means that the selected speaker has no volume
-        // control, hence it is safe to state that there is no mute control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected speaker has a mute control
-    //
-    _mixerManager.SpeakerMuteIsAvailable(isAvailable);
-
-    available = isAvailable;
-
-    // Close the initialized output mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetSpeakerMute(bool enable)
-{
-    return (_mixerManager.SetSpeakerMute(enable));
-}
-
-int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const
-{
-
-    bool muted(0);
-
-    if (_mixerManager.SpeakerMute(muted) == -1)
-    {
-        return -1;
-    }
-
-    enabled = muted;
-    return 0;
-}
-
-int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone has a mute control
-    //
-    _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetMicrophoneMute(bool enable)
-{
-    return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const
-{
-
-    bool muted(0);
-
-    if (_mixerManager.MicrophoneMute(muted) == -1)
-    {
-        return -1;
-    }
-
-    enabled = muted;
-    return 0;
-}
-
-int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Enumerate all avaliable microphone and make an attempt to open up the
-    // input mixer corresponding to the currently selected input device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control, hence it is safe to state that there is no boost control
-        // already at this stage.
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone has a boost control
-    //
-    _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable)
-{
-
-    return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const
-{
-
-    bool onOff(0);
-
-    if (_mixerManager.MicrophoneBoost(onOff) == -1)
-    {
-        return -1;
-    }
-
-    enabled = onOff;
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoRecordingIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetStereoRecording(bool enable)
-{
-
-    if (enable)
-        _recChannels = 2;
-    else
-        _recChannels = 1;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoRecording(bool& enabled) const
-{
-
-    if (_recChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
-{
-
-    bool isAvailable(false);
-    bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
-    if (!wasInitialized && InitSpeaker() == -1)
-    {
-        // Cannot open the specified device
-        available = false;
-        return 0;
-    }
-
-    // Check if the selected microphone can record stereo
-    //
-    _mixerManager.StereoPlayoutIsAvailable(isAvailable);
-    available = isAvailable;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseSpeaker();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetStereoPlayout(bool enable)
-{
-
-    if (enable)
-        _playChannels = 2;
-    else
-        _playChannels = 1;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const
-{
-
-    if (_playChannels == 2)
-        enabled = true;
-    else
-        enabled = false;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetAGC(bool enable)
-{
-
-    _AGC = enable;
-
-    return 0;
-}
-
-bool AudioDeviceMac::AGC() const
-{
-
-    return _AGC;
-}
-
-int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
-{
-
-    bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
-    // Make an attempt to open up the
-    // input mixer corresponding to the currently selected output device.
-    //
-    if (!wasInitialized && InitMicrophone() == -1)
-    {
-        // If we end up here it means that the selected microphone has no volume
-        // control.
-        available = false;
-        return 0;
-    }
-
-    // Given that InitMicrophone was successful, we know that a volume control
-    // exists
-    //
-    available = true;
-
-    // Close the initialized input mixer
-    //
-    if (!wasInitialized)
-    {
-        _mixerManager.CloseMicrophone();
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume)
-{
-
-    return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const
-{
-
-    uint32_t level(0);
-
-    if (_mixerManager.MicrophoneVolume(level) == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  failed to retrive current microphone level");
-        return -1;
-    }
-
-    volume = level;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
-{
-
-    uint32_t maxVol(0);
-
-    if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
-    {
-        return -1;
-    }
-
-    maxVolume = maxVol;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const
-{
-
-    uint32_t minVol(0);
-
-    if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
-    {
-        return -1;
-    }
-
-    minVolume = minVol;
-    return 0;
-}
-
-int32_t
-AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
-{
-
-    uint16_t delta(0);
-
-    if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
-    {
-        return -1;
-    }
-
-    stepSize = delta;
-    return 0;
-}
-
-int16_t AudioDeviceMac::PlayoutDevices()
-{
-
-    AudioDeviceID playDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
-                            MaxNumberDevices);
-}
-
-int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index)
-{
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playIsInitialized)
-    {
-        return -1;
-    }
-
-    AudioDeviceID playDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
-                                         playDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio output devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
-
-    _outputDeviceIndex = index;
-    _outputDeviceIsSpecified = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::SetPlayoutDevice(
-    AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
-}
-
-int32_t AudioDeviceMac::PlayoutDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
-
-    const uint16_t nDevices(PlayoutDevices());
-
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
-
-    memset(name, 0, kAdmMaxDeviceNameSize);
-
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
-}
-
-int32_t AudioDeviceMac::RecordingDeviceName(
-    uint16_t index,
-    char name[kAdmMaxDeviceNameSize],
-    char guid[kAdmMaxGuidSize])
-{
-
-    const uint16_t nDevices(RecordingDevices());
-
-    if ((index > (nDevices - 1)) || (name == NULL))
-    {
-        return -1;
-    }
-
-    memset(name, 0, kAdmMaxDeviceNameSize);
-
-    if (guid != NULL)
-    {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
-
-    return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
-}
-
-int16_t AudioDeviceMac::RecordingDevices()
-{
-
-    AudioDeviceID recDevices[MaxNumberDevices];
-    return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
-                            MaxNumberDevices);
-}
-
-int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index)
-{
-
-    if (_recIsInitialized)
-    {
-        return -1;
-    }
-
-    AudioDeviceID recDevices[MaxNumberDevices];
-    uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
-                                         recDevices, MaxNumberDevices);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  number of availiable waveform-audio input devices is %u",
-                 nDevices);
-
-    if (index > (nDevices - 1))
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  device index is out of range [0,%u]", (nDevices - 1));
-        return -1;
-    }
-
-    _inputDeviceIndex = index;
-    _inputDeviceIsSpecified = true;
-
-    return 0;
-}
-
-
-int32_t
-AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
-}
-
-int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available)
-{
-
-    available = true;
-
-    // Try to initialize the playout side
-    if (InitPlayout() == -1)
-    {
-        available = false;
-    }
-
-    // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
-    // We must actually start playout here in order to have the IOProc
-    // deleted by calling StopPlayout().
-    if (StartPlayout() == -1)
-    {
-        available = false;
-    }
-
-    // Cancel effect of initialization
-    if (StopPlayout() == -1)
-    {
-        available = false;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::RecordingIsAvailable(bool& available)
-{
-
-    available = true;
-
-    // Try to initialize the recording side
-    if (InitRecording() == -1)
-    {
-        available = false;
-    }
-
-    // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
-    // We must actually start recording here in order to have the IOProc
-    // deleted by calling StopRecording().
-    if (StartRecording() == -1)
-    {
-        available = false;
-    }
-
-    // Cancel effect of initialization
-    if (StopRecording() == -1)
-    {
-        available = false;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::InitPlayout()
-{
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_playing)
-    {
-        return -1;
-    }
-
-    if (!_outputDeviceIsSpecified)
-    {
-        return -1;
-    }
-
-    if (_playIsInitialized)
-    {
-        return 0;
-    }
-
-    // Initialize the speaker (devices might have been added or removed)
-    if (InitSpeaker() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitSpeaker() failed");
-    }
-
-    if (!MicrophoneIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (MicrophoneIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  MicrophoneIsAvailable() failed");
-        }
-    }
-
-    PaUtil_FlushRingBuffer(_paRenderBuffer);
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    _renderDelayOffsetSamples = 0;
-    _renderDelayUs = 0;
-    _renderLatencyUs = 0;
-    _renderDeviceIsAlive = 1;
-    _doStop = false;
-
-    // The internal microphone of a MacBook Pro is located under the left speaker
-    // grille. When the internal speakers are in use, we want to fully stereo
-    // pan to the right.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyDataSource,
-                kAudioDevicePropertyScopeOutput, 0 };
-    if (_macBookPro)
-    {
-        _macBookProPanRight = false;
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            UInt32 dataSource = 0;
-            size = sizeof(dataSource);
-            WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
-                    &propertyAddress, 0, NULL, &size, &dataSource));
-
-            if (dataSource == 'ispk')
-            {
-                _macBookProPanRight = true;
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id,
-                             "MacBook Pro using internal speakers; stereo"
-                             " panning right");
-            } else
-            {
-                WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
-                             _id, "MacBook Pro not using internal speakers");
-            }
-
-            // Add a listener to determine if the status changes.
-            WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
-    }
-
-    // Get current stream description
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
-    size = sizeof(_outStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_outStreamFormat));
-
-    if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable output stream format -> mFormatID",
-                 (const char *) &_outStreamFormat.mFormatID);
-        return -1;
-    }
-
-    if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on output device (mChannelsPerFrame = %d)",
-            _outStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Non-interleaved audio data is not supported.",
-                     "AudioHardware streams should not have this format.");
-        return -1;
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Ouput stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 _outStreamFormat.mSampleRate,
-                 _outStreamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _outStreamFormat.mBytesPerPacket,
-                 _outStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _outStreamFormat.mBytesPerFrame,
-                 _outStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 _outStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_outStreamFormat.mFormatID);
-
-    // Our preferred format to work with.
-    if (_outStreamFormat.mChannelsPerFrame < 2)
-    {
-        // Disable stereo playout when we only have one channel on the device.
-        _playChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo playout unavailable on this device");
-    }
-    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
-
-    // Listen for format changes.
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                           &propertyAddress,
-                                                           &objectListenerProc,
-                                                           this));
-
-    // Listen for processor overloads.
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
-                                                      &propertyAddress,
-                                                      &objectListenerProc,
-                                                      this));
-
-    if (_twoDevices || !_recIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
-                                                          deviceIOProc,
-                                                          this,
-                                                          &_deviceIOProcID));
-    }
-
-    _playIsInitialized = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::InitRecording()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_recording)
-    {
-        return -1;
-    }
-
-    if (!_inputDeviceIsSpecified)
-    {
-        return -1;
-    }
-
-    if (_recIsInitialized)
-    {
-        return 0;
-    }
-
-    // Initialize the microphone (devices might have been added or removed)
-    if (InitMicrophone() == -1)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitMicrophone() failed");
-    }
-
-    if (!SpeakerIsInitialized())
-    {
-        // Make this call to check if we are using
-        // one or two devices (_twoDevices)
-        bool available = false;
-        if (SpeakerIsAvailable(available) == -1)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  SpeakerIsAvailable() failed");
-        }
-    }
-
-    OSStatus err = noErr;
-    UInt32 size = 0;
-
-    PaUtil_FlushRingBuffer(_paCaptureBuffer);
-
-    _captureDelayUs = 0;
-    _captureLatencyUs = 0;
-    _captureDeviceIsAlive = 1;
-    _doStopRec = false;
-
-    // Get current stream description
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
-    size = sizeof(_inStreamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &_inStreamFormat));
-
-    if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &_inStreamFormat.mFormatID);
-        return -1;
-    }
-
-    if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Too many channels on input device (mChannelsPerFrame = %d)",
-            _inStreamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
-        _inStreamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-    if (io_block_size_samples > _captureBufSizeSamples)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "Input IO block size (%d) is larger than ring buffer (%u)",
-            io_block_size_samples, _captureBufSizeSamples);
-        return -1;
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " Input stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mSampleRate = %f, mChannelsPerFrame = %u",
-                 _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 _inStreamFormat.mBytesPerPacket,
-                 _inStreamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 _inStreamFormat.mBytesPerFrame,
-                 _inStreamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 " mFormatFlags = %u",
-                 _inStreamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &_inStreamFormat.mFormatID);
-
-    // Our preferred format to work with
-    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-    {
-        _inDesiredFormat.mChannelsPerFrame = 2;
-    } else
-    {
-        // Disable stereo recording when we only have one channel on the device.
-        _inDesiredFormat.mChannelsPerFrame = 1;
-        _recChannels = 1;
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Stereo recording unavailable on this device");
-    }
-
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters
-        _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-    }
-
-    _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
-    _inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mFramesPerPacket = 1;
-    _inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
-        * sizeof(SInt16);
-    _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
-
-    _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-        | kLinearPCMFormatFlagIsPacked;
-#ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
-    _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
-            &_captureConverter));
-
-    // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
-    // TODO(xians): investigate this block.
-    UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
-        * 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
-        * sizeof(Float32));
-    if (_inStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = ((UInt32)(bufByteCount
-                / _inStreamFormat.mFramesPerPacket) + 1)
-                * _inStreamFormat.mFramesPerPacket;
-        }
-    }
-
-    // Ensure the buffer size is within the acceptable range provided by the device.
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
-    }
-
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, size, &bufByteCount));
-
-    // Get capture device latency
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs = (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
-
-    // Get capture stream latency
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
-            &propertyAddress, 0, NULL, &size, &latency));
-    _captureLatencyUs += (UInt32)((1.0e6 * latency)
-        / _inStreamFormat.mSampleRate);
-
-    // Listen for format changes
-    // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
-    propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    // Listen for processor overloads
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                inDeviceIOProc, this, &_inDeviceIOProcID));
-    } else if (!_playIsInitialized)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
-                deviceIOProc, this, &_deviceIOProcID));
-    }
-
-    // Mark recording side as initialized
-    _recIsInitialized = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StartRecording()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized)
-    {
-        return -1;
-    }
-
-    if (_recording)
-    {
-        return 0;
-    }
-
-    if (!_initialized)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Recording worker thread has not been started");
-        return -1;
-    }
-
-    RTC_DCHECK(!capture_worker_thread_.get());
-    capture_worker_thread_.reset(
-        new rtc::PlatformThread(RunCapture, this, "CaptureWorkerThread"));
-    RTC_DCHECK(capture_worker_thread_.get());
-    capture_worker_thread_->Start();
-    capture_worker_thread_->SetPriority(rtc::kRealtimePriority);
-
-    OSStatus err = noErr;
-    if (_twoDevices)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
-    } else if (!_playing)
-    {
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
-    }
-
-    _recording = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StopRecording()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized)
-    {
-        return 0;
-    }
-
-    OSStatus err = noErr;
-
-    // Stop device
-    int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
-    if (_twoDevices)
-    {
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStopRec = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEventRec.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the capture IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
-                                                   _inDeviceIOProcID));
-                WEBRTC_CA_LOG_WARN(
-                    AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                               _inDeviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStopRec = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped");
-        }
-    }
-    else
-    {
-        // We signal a stop for a shared device even when rendering has
-        // not yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_recording|) before accessing
-        // resources we free below (e.g. the capture converter).
-        //
-        // In the case of a shared devcie, the IOProc will verify
-        // rendering has ended before stopping itself.
-        if (_recording && captureDeviceIsAlive == 1)
-        {
-            _recording = false;
-            _doStop = true; // Signal to io proc to stop audio device
-            _critSect.Leave(); // Cannot be under lock, risk of deadlock
-            if (kEventTimeout == _stopEvent.Wait(2000))
-            {
-                CriticalSectionScoped critScoped(&_critSect);
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             " Timed out stopping the shared IOProc. "
-                             "We may have failed to detect a device removal.");
-
-                // We assume rendering on a shared device has stopped as well if
-                // the IOProc times out.
-                WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                                   _deviceIOProcID));
-                WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                              _deviceIOProcID));
-            }
-            _critSect.Enter();
-            _doStop = false;
-            WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                         " Recording stopped (shared)");
-        }
-    }
-
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_captureDeviceIsAlive, 0);
-
-    if (capture_worker_thread_.get()) {
-        _critSect.Leave();
-        capture_worker_thread_->Stop();
-        capture_worker_thread_.reset();
-        _critSect.Enter();
-    }
-
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
-
-    // Remove listeners.
-    AudioObjectPropertyAddress
-        propertyAddress = { kAudioDevicePropertyStreamFormat,
-                kAudioDevicePropertyScopeInput, 0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    _recIsInitialized = false;
-    _recording = false;
-
-    return 0;
-}
-
-bool AudioDeviceMac::RecordingIsInitialized() const
-{
-    return (_recIsInitialized);
-}
-
-bool AudioDeviceMac::Recording() const
-{
-    return (_recording);
-}
-
-bool AudioDeviceMac::PlayoutIsInitialized() const
-{
-    return (_playIsInitialized);
-}
-
-int32_t AudioDeviceMac::StartPlayout()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized)
-    {
-        return -1;
-    }
-
-    if (_playing)
-    {
-        return 0;
-    }
-
-    RTC_DCHECK(!render_worker_thread_.get());
-    render_worker_thread_.reset(
-        new rtc::PlatformThread(RunRender, this, "RenderWorkerThread"));
-    render_worker_thread_->Start();
-    render_worker_thread_->SetPriority(rtc::kRealtimePriority);
-
-    if (_twoDevices || !_recording)
-    {
-        OSStatus err = noErr;
-        WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
-    }
-    _playing = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::StopPlayout()
-{
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized)
-    {
-        return 0;
-    }
-
-    OSStatus err = noErr;
-
-    int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
-    if (_playing && renderDeviceIsAlive == 1)
-    {
-        // We signal a stop for a shared device even when capturing has not
-        // yet ended. This is to ensure the IOProc will return early as
-        // intended (by checking |_playing|) before accessing resources we
-        // free below (e.g. the render converter).
-        //
-        // In the case of a shared device, the IOProc will verify capturing
-        // has ended before stopping itself.
-        _playing = false;
-        _doStop = true; // Signal to io proc to stop audio device
-        _critSect.Leave(); // Cannot be under lock, risk of deadlock
-        if (kEventTimeout == _stopEvent.Wait(2000))
-        {
-            CriticalSectionScoped critScoped(&_critSect);
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " Timed out stopping the render IOProc. "
-                         "We may have failed to detect a device removal.");
-
-            // We assume capturing on a shared device has stopped as well if the
-            // IOProc times out.
-            WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
-                                               _deviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                          _deviceIOProcID));
-        }
-        _critSect.Enter();
-        _doStop = false;
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Playout stopped");
-    }
-
-    // Setting this signal will allow the worker thread to be stopped.
-    AtomicSet32(&_renderDeviceIsAlive, 0);
-    if (render_worker_thread_.get()) {
-        _critSect.Leave();
-        render_worker_thread_->Stop();
-        render_worker_thread_.reset();
-        _critSect.Enter();
-    }
-
-    WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
-
-    // Remove listeners.
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
-            0 };
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    propertyAddress.mSelector = kAudioDeviceProcessorOverload;
-    WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-            &propertyAddress, &objectListenerProc, this));
-
-    if (_macBookPro)
-    {
-        Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
-                                                     &propertyAddress);
-        if (hasProperty)
-        {
-            propertyAddress.mSelector = kAudioDevicePropertyDataSource;
-            WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
-                    &propertyAddress, &objectListenerProc, this));
-        }
-    }
-
-    _playIsInitialized = false;
-    _playing = false;
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const
-{
-    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (renderDelayUs + _renderLatencyUs) +
-                                     0.5);
-    return 0;
-}
-
-int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const
-{
-    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-    delayMS = static_cast<uint16_t> (1e-3 * (captureDelayUs +
-                                             _captureLatencyUs) + 0.5);
-    return 0;
-}
-
-bool AudioDeviceMac::Playing() const
-{
-    return (_playing);
+    render_worker_thread_->Stop();
+    render_worker_thread_.reset();
+    _critSect.Enter();
+  }
+
+  WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
+
+  // Remove listeners.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput, 0};
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  propertyAddress.mSelector = kAudioDeviceProcessorOverload;
+  WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+      _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+
+  if (_macBookPro) {
+    Boolean hasProperty =
+        AudioObjectHasProperty(_outputDeviceID, &propertyAddress);
+    if (hasProperty) {
+      propertyAddress.mSelector = kAudioDevicePropertyDataSource;
+      WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(
+          _outputDeviceID, &propertyAddress, &objectListenerProc, this));
+    }
+  }
+
+  _playIsInitialized = false;
+  _playing = false;
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const {
+  int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+  return 0;
+}
+
+int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const {
+  int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+  delayMS =
+      static_cast<uint16_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+  return 0;
+}
+
+bool AudioDeviceMac::Playing() const {
+  return (_playing);
 }
 
 int32_t AudioDeviceMac::SetPlayoutBuffer(
     const AudioDeviceModule::BufferType type,
-    uint16_t sizeMS)
-{
-
-    if (type != AudioDeviceModule::kFixedBufferSize)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " Adaptive buffer size not supported on this platform");
-        return -1;
-    }
-
-    _playBufType = type;
-    _playBufDelayFixed = sizeMS;
-    return 0;
-}
-
-int32_t AudioDeviceMac::PlayoutBuffer(
-    AudioDeviceModule::BufferType& type,
-    uint16_t& sizeMS) const
-{
-
-    type = _playBufType;
-    sizeMS = _playBufDelayFixed;
-
-    return 0;
+    uint16_t sizeMS) {
+  if (type != AudioDeviceModule::kFixedBufferSize) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " Adaptive buffer size not supported on this platform");
+    return -1;
+  }
+
+  _playBufType = type;
+  _playBufDelayFixed = sizeMS;
+  return 0;
+}
+
+int32_t AudioDeviceMac::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                      uint16_t& sizeMS) const {
+  type = _playBufType;
+  sizeMS = _playBufDelayFixed;
+
+  return 0;
 }
 
 // Not implemented for Mac.
-int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const
-{
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-
-    return -1;
-}
-
-bool AudioDeviceMac::PlayoutWarning() const
-{
-    return (_playWarning > 0);
-}
-
-bool AudioDeviceMac::PlayoutError() const
-{
-    return (_playError > 0);
-}
-
-bool AudioDeviceMac::RecordingWarning() const
-{
-    return (_recWarning > 0);
-}
-
-bool AudioDeviceMac::RecordingError() const
-{
-    return (_recError > 0);
-}
-
-void AudioDeviceMac::ClearPlayoutWarning()
-{
-    _playWarning = 0;
-}
-
-void AudioDeviceMac::ClearPlayoutError()
-{
-    _playError = 0;
-}
-
-void AudioDeviceMac::ClearRecordingWarning()
-{
-    _recWarning = 0;
-}
-
-void AudioDeviceMac::ClearRecordingError()
-{
-    _recError = 0;
+int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const {
+  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+               "  API call not supported on this platform");
+
+  return -1;
+}
+
+bool AudioDeviceMac::PlayoutWarning() const {
+  return (_playWarning > 0);
+}
+
+bool AudioDeviceMac::PlayoutError() const {
+  return (_playError > 0);
+}
+
+bool AudioDeviceMac::RecordingWarning() const {
+  return (_recWarning > 0);
+}
+
+bool AudioDeviceMac::RecordingError() const {
+  return (_recError > 0);
+}
+
+void AudioDeviceMac::ClearPlayoutWarning() {
+  _playWarning = 0;
+}
+
+void AudioDeviceMac::ClearPlayoutError() {
+  _playError = 0;
+}
+
+void AudioDeviceMac::ClearRecordingWarning() {
+  _recWarning = 0;
+}
+
+void AudioDeviceMac::ClearRecordingError() {
+  _recError = 0;
 }
 
 // ============================================================================
 //                                 Private Methods
 // ============================================================================
 
-int32_t
-AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
-                                 AudioDeviceID scopedDeviceIds[],
-                                 const uint32_t deviceListLength)
-{
-    OSStatus err = noErr;
-
+int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
+                                         AudioDeviceID scopedDeviceIds[],
+                                         const uint32_t deviceListLength) {
+  OSStatus err = noErr;
+
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+  UInt32 size = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size));
+  if (size == 0) {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "No devices");
+    return 0;
+  }
+
+  AudioDeviceID* deviceIds = (AudioDeviceID*)malloc(size);
+  UInt32 numberDevices = size / sizeof(AudioDeviceID);
+  AudioBufferList* bufferList = NULL;
+  UInt32 numberScopedDevices = 0;
+
+  // First check if there is a default device and list it
+  UInt32 hardwareProperty = 0;
+  if (scope == kAudioDevicePropertyScopeOutput) {
+    hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+  } else {
+    hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+  }
+
+  AudioObjectPropertyAddress propertyAddressDefault = {
+      hardwareProperty, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  AudioDeviceID usedID;
+  UInt32 uintSize = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                                     &propertyAddressDefault, 0,
+                                                     NULL, &uintSize, &usedID));
+  if (usedID != kAudioDeviceUnknown) {
+    scopedDeviceIds[numberScopedDevices] = usedID;
+    numberScopedDevices++;
+  } else {
+    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                 "GetNumberDevices(): Default device unknown");
+  }
+
+  // Then list the rest of the devices
+  bool listOK = true;
+
+  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds));
+  if (err != noErr) {
+    listOK = false;
+  } else {
+    propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
+    propertyAddress.mScope = scope;
+    propertyAddress.mElement = 0;
+    for (UInt32 i = 0; i < numberDevices; i++) {
+      // Check for input channels
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(
+          deviceIds[i], &propertyAddress, 0, NULL, &size));
+      if (err == kAudioHardwareBadDeviceError) {
+        // This device doesn't actually exist; continue iterating.
+        continue;
+      } else if (err != noErr) {
+        listOK = false;
+        break;
+      }
+
+      bufferList = (AudioBufferList*)malloc(size);
+      WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
+          deviceIds[i], &propertyAddress, 0, NULL, &size, bufferList));
+      if (err != noErr) {
+        listOK = false;
+        break;
+      }
+
+      if (bufferList->mNumberBuffers > 0) {
+        if (numberScopedDevices >= deviceListLength) {
+          WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                       "Device list is not long enough");
+          listOK = false;
+          break;
+        }
+
+        scopedDeviceIds[numberScopedDevices] = deviceIds[i];
+        numberScopedDevices++;
+      }
+
+      free(bufferList);
+      bufferList = NULL;
+    }  // for
+  }
+
+  if (!listOK) {
+    if (deviceIds) {
+      free(deviceIds);
+      deviceIds = NULL;
+    }
+
+    if (bufferList) {
+      free(bufferList);
+      bufferList = NULL;
+    }
+
+    return -1;
+  }
+
+  // Happy ending
+  if (deviceIds) {
+    free(deviceIds);
+    deviceIds = NULL;
+  }
+
+  return numberScopedDevices;
+}
+
+int32_t AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
+                                      const uint16_t index,
+                                      char* name) {
+  OSStatus err = noErr;
+  UInt32 len = kAdmMaxDeviceNameSize;
+  AudioDeviceID deviceIds[MaxNumberDevices];
+
+  int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "No devices");
+    return -1;
+  }
+
+  // If the number is below the number of devices, assume it's "WEBRTC ID"
+  // otherwise assume it's a CoreAudio ID
+  AudioDeviceID usedID;
+
+  // Check if there is a default device
+  bool isDefaultDevice = false;
+  if (index == 0) {
+    UInt32 hardwareProperty = 0;
+    if (scope == kAudioDevicePropertyScopeOutput) {
+      hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
+    } else {
+      hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
+    }
     AudioObjectPropertyAddress propertyAddress = {
-            kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
-            kAudioObjectPropertyElementMaster };
-    UInt32 size = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size));
-    if (size == 0)
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "No devices");
-        return 0;
-    }
-
-    AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
-    UInt32 numberDevices = size / sizeof(AudioDeviceID);
-    AudioBufferList* bufferList = NULL;
-    UInt32 numberScopedDevices = 0;
-
-    // First check if there is a default device and list it
-    UInt32 hardwareProperty = 0;
-    if (scope == kAudioDevicePropertyScopeOutput)
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-    } else
-    {
-        hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-    }
-
-    AudioObjectPropertyAddress
-        propertyAddressDefault = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-
-    AudioDeviceID usedID;
-    UInt32 uintSize = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddressDefault, 0, NULL, &uintSize, &usedID));
-    if (usedID != kAudioDeviceUnknown)
-    {
-        scopedDeviceIds[numberScopedDevices] = usedID;
-        numberScopedDevices++;
-    } else
-    {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "GetNumberDevices(): Default device unknown");
-    }
-
-    // Then list the rest of the devices
-    bool listOK = true;
-
-    WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-            &propertyAddress, 0, NULL, &size, deviceIds));
-    if (err != noErr)
-    {
-        listOK = false;
-    } else
-    {
-        propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
-        propertyAddress.mScope = scope;
-        propertyAddress.mElement = 0;
-        for (UInt32 i = 0; i < numberDevices; i++)
-        {
-            // Check for input channels
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size));
-            if (err == kAudioHardwareBadDeviceError)
-            {
-                // This device doesn't actually exist; continue iterating.
-                continue;
-            } else if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
-
-            bufferList = (AudioBufferList*) malloc(size);
-            WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
-                    &propertyAddress, 0, NULL, &size, bufferList));
-            if (err != noErr)
-            {
-                listOK = false;
-                break;
-            }
-
-            if (bufferList->mNumberBuffers > 0)
-            {
-                if (numberScopedDevices >= deviceListLength)
-                {
-                    WEBRTC_TRACE(kTraceError,
-                                 kTraceAudioDevice, _id,
-                                 "Device list is not long enough");
-                    listOK = false;
-                    break;
-                }
-
-                scopedDeviceIds[numberScopedDevices] = deviceIds[i];
-                numberScopedDevices++;
-            }
-
-            free(bufferList);
-            bufferList = NULL;
-        }  // for
-    }
-
-    if (!listOK)
-    {
-        if (deviceIds)
-        {
-            free(deviceIds);
-            deviceIds = NULL;
-        }
-
-        if (bufferList)
-        {
-            free(bufferList);
-            bufferList = NULL;
-        }
-
-        return -1;
-    }
-
-    // Happy ending
-    if (deviceIds)
-    {
-        free(deviceIds);
-        deviceIds = NULL;
-    }
-
-    return numberScopedDevices;
-}
-
-int32_t
-AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
-                              const uint16_t index,
-                              char* name)
-{
-    OSStatus err = noErr;
-    UInt32 len = kAdmMaxDeviceNameSize;
-    AudioDeviceID deviceIds[MaxNumberDevices];
-
-    int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "No devices");
-        return -1;
-    }
-
-    // If the number is below the number of devices, assume it's "WEBRTC ID"
-    // otherwise assume it's a CoreAudio ID
-    AudioDeviceID usedID;
-
-    // Check if there is a default device
-    bool isDefaultDevice = false;
-    if (index == 0)
-    {
-        UInt32 hardwareProperty = 0;
-        if (scope == kAudioDevicePropertyScopeOutput)
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
-        } else
-        {
-            hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
-        }
-        AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &usedID));
-        if (usedID == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "GetDeviceName(): Default device unknown");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
-
-    AudioObjectPropertyAddress propertyAddress = {
-            kAudioDevicePropertyDeviceName, scope, 0 };
-
-    if (isDefaultDevice)
-    {
-        char devName[len];
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, devName));
-
-        sprintf(name, "default (%s)", devName);
-    } else
-    {
-        if (index < numberDevices)
-        {
-            usedID = deviceIds[index];
-        } else
-        {
-            usedID = index;
-        }
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
-                &propertyAddress, 0, NULL, &len, name));
-    }
-
-    return 0;
+        hardwareProperty, kAudioObjectPropertyScopeGlobal,
+        kAudioObjectPropertyElementMaster};
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &usedID));
+    if (usedID == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "GetDeviceName(): Default device unknown");
+    } else {
+      isDefaultDevice = true;
+    }
+  }
+
+  AudioObjectPropertyAddress propertyAddress = {kAudioDevicePropertyDeviceName,
+                                                scope, 0};
+
+  if (isDefaultDevice) {
+    char devName[len];
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, devName));
+
+    sprintf(name, "default (%s)", devName);
+  } else {
+    if (index < numberDevices) {
+      usedID = deviceIds[index];
+    } else {
+      usedID = index;
+    }
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID, &propertyAddress,
+                                                       0, NULL, &len, name));
+  }
+
+  return 0;
 }
 
 int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
                                    AudioDeviceID& deviceId,
-                                   const bool isInput)
-{
-    OSStatus err = noErr;
-    UInt32 size = 0;
-    AudioObjectPropertyScope deviceScope;
-    AudioObjectPropertySelector defaultDeviceSelector;
-    AudioDeviceID deviceIds[MaxNumberDevices];
-
-    if (isInput)
-    {
-        deviceScope = kAudioDevicePropertyScopeInput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
-    } else
-    {
-        deviceScope = kAudioDevicePropertyScopeOutput;
-        defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
-    }
-
-    AudioObjectPropertyAddress
-        propertyAddress = { defaultDeviceSelector,
-                kAudioObjectPropertyScopeGlobal,
-                kAudioObjectPropertyElementMaster };
-
-    // Get the actual device IDs
-    int numberDevices = GetNumberDevices(deviceScope, deviceIds,
-                                         MaxNumberDevices);
-    if (numberDevices < 0)
-    {
-        return -1;
-    } else if (numberDevices == 0)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "InitDevice(): No devices");
-        return -1;
-    }
-
-    bool isDefaultDevice = false;
-    deviceId = kAudioDeviceUnknown;
-    if (userDeviceIndex == 0)
-    {
-        // Try to use default system device
-        size = sizeof(AudioDeviceID);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
-                &propertyAddress, 0, NULL, &size, &deviceId));
-        if (deviceId == kAudioDeviceUnknown)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         " No default device exists");
-        } else
-        {
-            isDefaultDevice = true;
-        }
-    }
-
-    if (!isDefaultDevice)
-    {
-        deviceId = deviceIds[userDeviceIndex];
-    }
-
-    // Obtain device name and manufacturer for logging.
-    // Also use this as a test to ensure a user-set device ID is valid.
-    char devName[128];
-    char devManf[128];
-    memset(devName, 0, sizeof(devName));
-    memset(devManf, 0, sizeof(devManf));
-
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
-    propertyAddress.mScope = deviceScope;
-    propertyAddress.mElement = 0;
-    size = sizeof(devName);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devName));
-
-    propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
-    size = sizeof(devManf);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
-            &propertyAddress, 0, NULL, &size, devManf));
-
-    if (isInput)
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Input device: %s %s", devManf, devName);
-    } else
-    {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     " Output device: %s %s", devManf, devName);
-    }
-
-    return 0;
-}
-
-OSStatus AudioDeviceMac::SetDesiredPlayoutFormat()
-{
-    // Our preferred format to work with.
-    _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
-    _outDesiredFormat.mChannelsPerFrame = _playChannels;
-
-    if (_ptrAudioBuffer)
-    {
-        // Update audio buffer with the selected parameters.
-        _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
-        _ptrAudioBuffer->SetPlayoutChannels((uint8_t) _playChannels);
-    }
-
-    _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT *
-        ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
-
-    _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame *
-                                        sizeof(SInt16);
-    // In uncompressed audio, a packet is one frame.
-    _outDesiredFormat.mFramesPerPacket = 1;
-    _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame *
-                                       sizeof(SInt16);
-    _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
-
-    _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
-                                     kLinearPCMFormatFlagIsPacked;
+                                   const bool isInput) {
+  OSStatus err = noErr;
+  UInt32 size = 0;
+  AudioObjectPropertyScope deviceScope;
+  AudioObjectPropertySelector defaultDeviceSelector;
+  AudioDeviceID deviceIds[MaxNumberDevices];
+
+  if (isInput) {
+    deviceScope = kAudioDevicePropertyScopeInput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
+  } else {
+    deviceScope = kAudioDevicePropertyScopeOutput;
+    defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
+  }
+
+  AudioObjectPropertyAddress propertyAddress = {
+      defaultDeviceSelector, kAudioObjectPropertyScopeGlobal,
+      kAudioObjectPropertyElementMaster};
+
+  // Get the actual device IDs
+  int numberDevices =
+      GetNumberDevices(deviceScope, deviceIds, MaxNumberDevices);
+  if (numberDevices < 0) {
+    return -1;
+  } else if (numberDevices == 0) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "InitDevice(): No devices");
+    return -1;
+  }
+
+  bool isDefaultDevice = false;
+  deviceId = kAudioDeviceUnknown;
+  if (userDeviceIndex == 0) {
+    // Try to use default system device
+    size = sizeof(AudioDeviceID);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, &deviceId));
+    if (deviceId == kAudioDeviceUnknown) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   " No default device exists");
+    } else {
+      isDefaultDevice = true;
+    }
+  }
+
+  if (!isDefaultDevice) {
+    deviceId = deviceIds[userDeviceIndex];
+  }
+
+  // Obtain device name and manufacturer for logging.
+  // Also use this as a test to ensure a user-set device ID is valid.
+  char devName[128];
+  char devManf[128];
+  memset(devName, 0, sizeof(devName));
+  memset(devManf, 0, sizeof(devManf));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
+  propertyAddress.mScope = deviceScope;
+  propertyAddress.mElement = 0;
+  size = sizeof(devName);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devName));
+
+  propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
+  size = sizeof(devManf);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId, &propertyAddress,
+                                                     0, NULL, &size, devManf));
+
+  if (isInput) {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Input device: %s %s",
+                 devManf, devName);
+  } else {
+    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " Output device: %s %s",
+                 devManf, devName);
+  }
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::SetDesiredPlayoutFormat() {
+  // Our preferred format to work with.
+  _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
+  _outDesiredFormat.mChannelsPerFrame = _playChannels;
+
+  if (_ptrAudioBuffer) {
+    // Update audio buffer with the selected parameters.
+    _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
+    _ptrAudioBuffer->SetPlayoutChannels((uint8_t)_playChannels);
+  }
+
+  _renderDelayOffsetSamples = _renderBufSizeSamples -
+                              N_BUFFERS_OUT * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES *
+                                  _outDesiredFormat.mChannelsPerFrame;
+
+  _outDesiredFormat.mBytesPerPacket =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  // In uncompressed audio, a packet is one frame.
+  _outDesiredFormat.mFramesPerPacket = 1;
+  _outDesiredFormat.mBytesPerFrame =
+      _outDesiredFormat.mChannelsPerFrame * sizeof(SInt16);
+  _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
+
+  _outDesiredFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
 #ifdef WEBRTC_ARCH_BIG_ENDIAN
-    _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
+  _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
 #endif
-    _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
-    OSStatus err = noErr;
-    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat,
-                                              &_outStreamFormat,
-                                              &_renderConverter));
-
-    // Try to set buffer size to desired value (_playBufDelayFixed).
-    UInt32 bufByteCount = static_cast<UInt32> ((_outStreamFormat.mSampleRate /
-                          1000.0) *
-                          _playBufDelayFixed *
-                          _outStreamFormat.mChannelsPerFrame *
-                          sizeof(Float32));
-    if (_outStreamFormat.mFramesPerPacket != 0)
-    {
-        if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
-        {
-            bufByteCount = (static_cast<UInt32> (bufByteCount /
-                           _outStreamFormat.mFramesPerPacket) + 1) *
-                           _outStreamFormat.mFramesPerPacket;
-        }
-    }
-
-    // Ensure the buffer size is within the range provided by the device.
-    AudioObjectPropertyAddress propertyAddress =
-        {kAudioDevicePropertyDataSource,
-         kAudioDevicePropertyScopeOutput,
-         0};
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
-    AudioValueRange range;
-    UInt32 size = sizeof(range);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &range));
-    if (range.mMinimum > bufByteCount)
-    {
-        bufByteCount = range.mMinimum;
-    } else if (range.mMaximum < bufByteCount)
-    {
-        bufByteCount = range.mMaximum;
-    }
-
-    propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
-    size = sizeof(bufByteCount);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       size,
-                                                       &bufByteCount));
-
-    // Get render device latency.
-    propertyAddress.mSelector = kAudioDevicePropertyLatency;
-    UInt32 latency = 0;
-    size = sizeof(UInt32);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs = static_cast<uint32_t> ((1.0e6 * latency) /
-                       _outStreamFormat.mSampleRate);
-
-    // Get render stream latency.
-    propertyAddress.mSelector = kAudioDevicePropertyStreams;
-    AudioStreamID stream = 0;
-    size = sizeof(AudioStreamID);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &stream));
-    propertyAddress.mSelector = kAudioStreamPropertyLatency;
-    size = sizeof(UInt32);
-    latency = 0;
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
-                                                       &propertyAddress,
-                                                       0,
-                                                       NULL,
-                                                       &size,
-                                                       &latency));
-    _renderLatencyUs += static_cast<uint32_t> ((1.0e6 * latency) /
-                        _outStreamFormat.mSampleRate);
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  initial playout status: _renderDelayOffsetSamples=%d,"
-                 " _renderDelayUs=%d, _renderLatencyUs=%d",
-                 _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
-    return 0;
+  _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
+
+  OSStatus err = noErr;
+  WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(
+      &_outDesiredFormat, &_outStreamFormat, &_renderConverter));
+
+  // Try to set buffer size to desired value (_playBufDelayFixed).
+  UInt32 bufByteCount = static_cast<UInt32>(
+      (_outStreamFormat.mSampleRate / 1000.0) * _playBufDelayFixed *
+      _outStreamFormat.mChannelsPerFrame * sizeof(Float32));
+  if (_outStreamFormat.mFramesPerPacket != 0) {
+    if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0) {
+      bufByteCount = (static_cast<UInt32>(bufByteCount /
+                                          _outStreamFormat.mFramesPerPacket) +
+                      1) *
+                     _outStreamFormat.mFramesPerPacket;
+    }
+  }
+
+  // Ensure the buffer size is within the range provided by the device.
+  AudioObjectPropertyAddress propertyAddress = {
+      kAudioDevicePropertyDataSource, kAudioDevicePropertyScopeOutput, 0};
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
+  AudioValueRange range;
+  UInt32 size = sizeof(range);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &range));
+  if (range.mMinimum > bufByteCount) {
+    bufByteCount = range.mMinimum;
+  } else if (range.mMaximum < bufByteCount) {
+    bufByteCount = range.mMaximum;
+  }
+
+  propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
+  size = sizeof(bufByteCount);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, size, &bufByteCount));
+
+  // Get render device latency.
+  propertyAddress.mSelector = kAudioDevicePropertyLatency;
+  UInt32 latency = 0;
+  size = sizeof(UInt32);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs =
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  // Get render stream latency.
+  propertyAddress.mSelector = kAudioDevicePropertyStreams;
+  AudioStreamID stream = 0;
+  size = sizeof(AudioStreamID);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &stream));
+  propertyAddress.mSelector = kAudioStreamPropertyLatency;
+  size = sizeof(UInt32);
+  latency = 0;
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      _outputDeviceID, &propertyAddress, 0, NULL, &size, &latency));
+  _renderLatencyUs +=
+      static_cast<uint32_t>((1.0e6 * latency) / _outStreamFormat.mSampleRate);
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "  initial playout status: _renderDelayOffsetSamples=%d,"
+               " _renderDelayUs=%d, _renderLatencyUs=%d",
+               _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
+  return 0;
 }
 
 OSStatus AudioDeviceMac::objectListenerProc(
     AudioObjectID objectId,
     UInt32 numberAddresses,
     const AudioObjectPropertyAddress addresses[],
-    void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
-
-    ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
-
-    // AudioObjectPropertyListenerProc functions are supposed to return 0
-    return 0;
+    void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
+
+  // AudioObjectPropertyListenerProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::implObjectListenerProc(
     const AudioObjectID objectId,
     const UInt32 numberAddresses,
-    const AudioObjectPropertyAddress addresses[])
-{
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "AudioDeviceMac::implObjectListenerProc()");
-
-    for (UInt32 i = 0; i < numberAddresses; i++)
-    {
-        if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
-        {
-            HandleDeviceChange();
-        } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
-        {
-            HandleStreamFormatChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
-        {
-            HandleDataSourceChange(objectId, addresses[i]);
-        } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
-        {
-            HandleProcessorOverload(addresses[i]);
-        }
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceMac::HandleDeviceChange()
-{
-    OSStatus err = noErr;
-
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "kAudioHardwarePropertyDevices");
-
-    // A device has changed. Check if our registered devices have been removed.
-    // Ensure the devices have been initialized, meaning the IDs are valid.
-    if (MicrophoneIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeInput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
-
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Capture device is not alive (probably removed)");
-            AtomicSet32(&_captureDeviceIsAlive, 0);
-            _mixerManager.CloseMicrophone();
-            if (_recError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending recording error exists");
-            }
-            _recError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
-    }
-
-    if (SpeakerIsInitialized())
-    {
-        AudioObjectPropertyAddress propertyAddress = {
-                kAudioDevicePropertyDeviceIsAlive,
-                kAudioDevicePropertyScopeOutput, 0 };
-        UInt32 deviceIsAlive = 1;
-        UInt32 size = sizeof(UInt32);
-        err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
-                                         NULL, &size, &deviceIsAlive);
-
-        if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
-        {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "Render device is not alive (probably removed)");
-            AtomicSet32(&_renderDeviceIsAlive, 0);
-            _mixerManager.CloseSpeaker();
-            if (_playError == 1)
-            {
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                             _id, "  pending playout error exists");
-            }
-            _playError = 1; // triggers callback from module process thread
-        } else if (err != noErr)
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioDeviceGetProperty()", (const char*) &err);
-            return -1;
-        }
-    }
-
-    return 0;
+    const AudioObjectPropertyAddress addresses[]) {
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "AudioDeviceMac::implObjectListenerProc()");
+
+  for (UInt32 i = 0; i < numberAddresses; i++) {
+    if (addresses[i].mSelector == kAudioHardwarePropertyDevices) {
+      HandleDeviceChange();
+    } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat) {
+      HandleStreamFormatChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource) {
+      HandleDataSourceChange(objectId, addresses[i]);
+    } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload) {
+      HandleProcessorOverload(addresses[i]);
+    }
+  }
+
+  return 0;
+}
+
+int32_t AudioDeviceMac::HandleDeviceChange() {
+  OSStatus err = noErr;
+
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+               "kAudioHardwarePropertyDevices");
+
+  // A device has changed. Check if our registered devices have been removed.
+  // Ensure the devices have been initialized, meaning the IDs are valid.
+  if (MicrophoneIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeInput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Capture device is not alive (probably removed)");
+      AtomicSet32(&_captureDeviceIsAlive, 0);
+      _mixerManager.CloseMicrophone();
+      if (_recError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending recording error exists");
+      }
+      _recError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
+    }
+  }
+
+  if (SpeakerIsInitialized()) {
+    AudioObjectPropertyAddress propertyAddress = {
+        kAudioDevicePropertyDeviceIsAlive, kAudioDevicePropertyScopeOutput, 0};
+    UInt32 deviceIsAlive = 1;
+    UInt32 size = sizeof(UInt32);
+    err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0, NULL,
+                                     &size, &deviceIsAlive);
+
+    if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0) {
+      WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                   "Render device is not alive (probably removed)");
+      AtomicSet32(&_renderDeviceIsAlive, 0);
+      _mixerManager.CloseSpeaker();
+      if (_playError == 1) {
+        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                     "  pending playout error exists");
+      }
+      _playError = 1;  // triggers callback from module process thread
+    } else if (err != noErr) {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioDeviceGetProperty()", (const char*)&err);
+      return -1;
+    }
+  }
+
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleStreamFormatChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
-
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "Stream format changed");
-
-    if (objectId != _inputDeviceID && objectId != _outputDeviceID)
-    {
-        return 0;
-    }
-
-    // Get the new device format
-    AudioStreamBasicDescription streamFormat;
-    UInt32 size = sizeof(streamFormat);
-    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-            &propertyAddress, 0, NULL, &size, &streamFormat));
-
-    if (streamFormat.mFormatID != kAudioFormatLinearPCM)
-    {
-        logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                 "Unacceptable input stream format -> mFormatID",
-                 (const char *) &streamFormat.mFormatID);
-        return -1;
-    }
-
-    if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "Too many channels on device (mChannelsPerFrame = %d)",
-                     streamFormat.mChannelsPerFrame);
-        return -1;
-    }
-
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "Stream format:");
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mSampleRate = %f, mChannelsPerFrame = %u",
-                 streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerPacket = %u, mFramesPerPacket = %u",
-                 streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mBytesPerFrame = %u, mBitsPerChannel = %u",
-                 streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "mFormatFlags = %u",
-                 streamFormat.mFormatFlags);
-    logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
-             (const char *) &streamFormat.mFormatID);
-
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        const int io_block_size_samples = streamFormat.mChannelsPerFrame *
-            streamFormat.mSampleRate / 100 * N_BLOCKS_IO;
-        if (io_block_size_samples > _captureBufSizeSamples)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                "Input IO block size (%d) is larger than ring buffer (%u)",
-                io_block_size_samples, _captureBufSizeSamples);
-            return -1;
-
-        }
-
-        memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
-        {
-            _inDesiredFormat.mChannelsPerFrame = 2;
-        } else
-        {
-            // Disable stereo recording when we only have one channel on the device.
-            _inDesiredFormat.mChannelsPerFrame = 1;
-            _recChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo recording unavailable on this device");
-        }
-
-        if (_ptrAudioBuffer)
-        {
-            // Update audio buffer with the selected parameters
-            _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
-            _ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
-        }
-
-        // Recreate the converter with the new format
-        // TODO(xians): make this thread safe
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
-
-        WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
-                &_captureConverter));
-    } else
-    {
-        memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
-
-        // Our preferred format to work with
-        if (_outStreamFormat.mChannelsPerFrame < 2)
-        {
-            _playChannels = 1;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "Stereo playout unavailable on this device");
-        }
-        WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
-    }
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
+
+  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Stream format changed");
+
+  if (objectId != _inputDeviceID && objectId != _outputDeviceID) {
     return 0;
+  }
+
+  // Get the new device format
+  AudioStreamBasicDescription streamFormat;
+  UInt32 size = sizeof(streamFormat);
+  WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+      objectId, &propertyAddress, 0, NULL, &size, &streamFormat));
+
+  if (streamFormat.mFormatID != kAudioFormatLinearPCM) {
+    logCAMsg(kTraceError, kTraceAudioDevice, _id,
+             "Unacceptable input stream format -> mFormatID",
+             (const char*)&streamFormat.mFormatID);
+    return -1;
+  }
+
+  if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "Too many channels on device (mChannelsPerFrame = %d)",
+                 streamFormat.mChannelsPerFrame);
+    return -1;
+  }
+
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Stream format:");
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mSampleRate = %f, mChannelsPerFrame = %u",
+               streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerPacket = %u, mFramesPerPacket = %u",
+               streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+               "mBytesPerFrame = %u, mBitsPerChannel = %u",
+               streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
+  WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "mFormatFlags = %u",
+               streamFormat.mFormatFlags);
+  logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
+           (const char*)&streamFormat.mFormatID);
+
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    const int io_block_size_samples = streamFormat.mChannelsPerFrame *
+                                      streamFormat.mSampleRate / 100 *
+                                      N_BLOCKS_IO;
+    if (io_block_size_samples > _captureBufSizeSamples) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "Input IO block size (%d) is larger than ring buffer (%u)",
+                   io_block_size_samples, _captureBufSizeSamples);
+      return -1;
+    }
+
+    memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) {
+      _inDesiredFormat.mChannelsPerFrame = 2;
+    } else {
+      // Disable stereo recording when we only have one channel on the device.
+      _inDesiredFormat.mChannelsPerFrame = 1;
+      _recChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo recording unavailable on this device");
+    }
+
+    if (_ptrAudioBuffer) {
+      // Update audio buffer with the selected parameters
+      _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
+      _ptrAudioBuffer->SetRecordingChannels((uint8_t)_recChannels);
+    }
+
+    // Recreate the converter with the new format
+    // TODO(xians): make this thread safe
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
+
+    WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
+                                              &_captureConverter));
+  } else {
+    memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
+
+    // Our preferred format to work with
+    if (_outStreamFormat.mChannelsPerFrame < 2) {
+      _playChannels = 1;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "Stereo playout unavailable on this device");
+    }
+    WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
+  }
+  return 0;
 }
 
 int32_t AudioDeviceMac::HandleDataSourceChange(
     const AudioObjectID objectId,
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    OSStatus err = noErr;
-
-    if (_macBookPro && propertyAddress.mScope
-        == kAudioDevicePropertyScopeOutput)
-    {
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "Data source changed");
-
-        _macBookProPanRight = false;
-        UInt32 dataSource = 0;
-        UInt32 size = sizeof(UInt32);
-        WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
-                &propertyAddress, 0, NULL, &size, &dataSource));
-        if (dataSource == 'ispk')
-        {
-            _macBookProPanRight = true;
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro using internal speakers; stereo panning right");
-        } else
-        {
-            WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                         "MacBook Pro not using internal speakers");
-        }
-    }
-
-    return 0;
+    const AudioObjectPropertyAddress propertyAddress) {
+  OSStatus err = noErr;
+
+  if (_macBookPro &&
+      propertyAddress.mScope == kAudioDevicePropertyScopeOutput) {
+    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, "Data source changed");
+
+    _macBookProPanRight = false;
+    UInt32 dataSource = 0;
+    UInt32 size = sizeof(UInt32);
+    WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(
+        objectId, &propertyAddress, 0, NULL, &size, &dataSource));
+    if (dataSource == 'ispk') {
+      _macBookProPanRight = true;
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro using internal speakers; stereo panning right");
+    } else {
+      WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+                   "MacBook Pro not using internal speakers");
+    }
+  }
+
+  return 0;
 }
 int32_t AudioDeviceMac::HandleProcessorOverload(
-    const AudioObjectPropertyAddress propertyAddress)
-{
-    // TODO(xians): we probably want to notify the user in some way of the
-    // overload. However, the Windows interpretations of these errors seem to
-    // be more severe than what ProcessorOverload is thrown for.
-    //
-    // We don't log the notification, as it's sent from the HAL's IO thread. We
-    // don't want to slow it down even further.
-    if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
-        // overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::ERecordingProblem);
-    } else
-    {
-        //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-        // "Render processor overload");
-        //_callback->ProblemIsReported(
-        // SndCardStreamObserver::EPlaybackProblem);
-    }
-
-    return 0;
+    const AudioObjectPropertyAddress propertyAddress) {
+  // TODO(xians): we probably want to notify the user in some way of the
+  // overload. However, the Windows interpretations of these errors seem to
+  // be more severe than what ProcessorOverload is thrown for.
+  //
+  // We don't log the notification, as it's sent from the HAL's IO thread. We
+  // don't want to slow it down even further.
+  if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
+    // overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::ERecordingProblem);
+  } else {
+    // WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+    // "Render processor overload");
+    //_callback->ProblemIsReported(
+    // SndCardStreamObserver::EPlaybackProblem);
+  }
+
+  return 0;
 }
 
 // ============================================================================
 //                                  Thread Methods
 // ============================================================================
 
-OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID,
+                                      const AudioTimeStamp*,
                                       const AudioBufferList* inputData,
                                       const AudioTimeStamp* inputTime,
                                       AudioBufferList* outputData,
                                       const AudioTimeStamp* outputTime,
-                                      void *clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
-
-    ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
-
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+                                      void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
+
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
-                                          UInt32 *numberDataPackets,
-                                          AudioBufferList *data,
-                                          AudioStreamPacketDescription **,
-                                          void *userData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
-    RTC_DCHECK(ptrThis != NULL);
-
-    return ptrThis->implOutConverterProc(numberDataPackets, data);
-}
-
-OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
+                                          UInt32* numberDataPackets,
+                                          AudioBufferList* data,
+                                          AudioStreamPacketDescription**,
+                                          void* userData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)userData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  return ptrThis->implOutConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID,
+                                        const AudioTimeStamp*,
                                         const AudioBufferList* inputData,
                                         const AudioTimeStamp* inputTime,
                                         AudioBufferList*,
-                                        const AudioTimeStamp*, void* clientData)
-{
-    AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
-    RTC_DCHECK(ptrThis != NULL);
-
-    ptrThis->implInDeviceIOProc(inputData, inputTime);
-
-    // AudioDeviceIOProc functions are supposed to return 0
-    return 0;
+                                        const AudioTimeStamp*,
+                                        void* clientData) {
+  AudioDeviceMac* ptrThis = (AudioDeviceMac*)clientData;
+  RTC_DCHECK(ptrThis != NULL);
+
+  ptrThis->implInDeviceIOProc(inputData, inputTime);
+
+  // AudioDeviceIOProc functions are supposed to return 0
+  return 0;
 }
 
 OSStatus AudioDeviceMac::inConverterProc(
     AudioConverterRef,
-    UInt32 *numberDataPackets,
-    AudioBufferList *data,
-    AudioStreamPacketDescription ** /*dataPacketDescription*/,
-    void *userData)
-{
-    AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
-    RTC_DCHECK(ptrThis != NULL);
-
-    return ptrThis->implInConverterProc(numberDataPackets, data);
-}
-
-OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
-                                          const AudioTimeStamp *inputTime,
-                                          AudioBufferList *outputData,
-                                          const AudioTimeStamp *outputTime)
-{
-    OSStatus err = noErr;
-    UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-    if (!_twoDevices && _recording)
-    {
-        implInDeviceIOProc(inputData, inputTime);
-    }
-
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStop)
-    {
-        _critSect.Enter();
-        if (_doStop)
-        {
-            if (_twoDevices || (!_recording && !_playing))
-            {
-               // In the case of a shared device, the single driving ioProc
-               // is stopped here
-               WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
-                                                 _deviceIOProcID));
-               WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
-                                                             _deviceIOProcID));
-               if (err == noErr)
-               {
-                  WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                               _id, " Playout or shared device stopped");
-               }
-            }
-
-            _doStop = false;
-            _stopEvent.Set();
-            _critSect.Leave();
-            return 0;
+    UInt32* numberDataPackets,
+    AudioBufferList* data,
+    AudioStreamPacketDescription** /*dataPacketDescription*/,
+    void* userData) {
+  AudioDeviceMac* ptrThis = static_cast<AudioDeviceMac*>(userData);
+  RTC_DCHECK(ptrThis != NULL);
+
+  return ptrThis->implInConverterProc(numberDataPackets, data);
+}
+
+OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList* inputData,
+                                          const AudioTimeStamp* inputTime,
+                                          AudioBufferList* outputData,
+                                          const AudioTimeStamp* outputTime) {
+  OSStatus err = noErr;
+  UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  if (!_twoDevices && _recording) {
+    implInDeviceIOProc(inputData, inputTime);
+  }
+
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStop) {
+    _critSect.Enter();
+    if (_doStop) {
+      if (_twoDevices || (!_recording && !_playing)) {
+        // In the case of a shared device, the single driving ioProc
+        // is stopped here
+        WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID, _deviceIOProcID));
+        WEBRTC_CA_LOG_WARN(
+            AudioDeviceDestroyIOProcID(_outputDeviceID, _deviceIOProcID));
+        if (err == noErr) {
+          WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                       " Playout or shared device stopped");
         }
-        _critSect.Leave();
-    }
-
-    if (!_playing)
-    {
-        // This can be the case when a shared device is capturing but not
-        // rendering. We allow the checks above before returning to avoid a
-        // timeout when capturing is stopped.
-        return 0;
-    }
-
-    RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
-    UInt32 size = outputData->mBuffers->mDataByteSize
-        / _outStreamFormat.mBytesPerFrame;
-
-    // TODO(xians): signal an error somehow?
-    err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
-                                          this, &size, outputData, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " Error in AudioConverterFillComplexBuffer()");
-            return 1;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return 1;
+      }
+
+      _doStop = false;
+      _stopEvent.Set();
+      _critSect.Leave();
+      return 0;
+    }
+    _critSect.Leave();
+  }
+
+  if (!_playing) {
+    // This can be the case when a shared device is capturing but not
+    // rendering. We allow the checks above before returning to avoid a
+    // timeout when capturing is stopped.
+    return 0;
+  }
+
+  RTC_DCHECK(_outStreamFormat.mBytesPerFrame != 0);
+  UInt32 size =
+      outputData->mBuffers->mDataByteSize / _outStreamFormat.mBytesPerFrame;
+
+  // TODO(xians): signal an error somehow?
+  err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
+                                        this, &size, outputData, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " Error in AudioConverterFillComplexBuffer()");
+      return 1;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return 1;
+    }
+  }
+
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
+
+  int32_t renderDelayUs =
+      static_cast<int32_t>(1e-3 * (outputTimeNs - nowNs) + 0.5);
+  renderDelayUs += static_cast<int32_t>(
+      (1.0e6 * bufSizeSamples) / _outDesiredFormat.mChannelsPerFrame /
+          _outDesiredFormat.mSampleRate +
+      0.5);
+
+  AtomicSet32(&_renderDelayUs, renderDelayUs);
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implOutConverterProc(UInt32* numberDataPackets,
+                                              AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _outDesiredFormat.mChannelsPerFrame;
+
+  data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
+  // Always give the converter as much as it wants, zero padding as required.
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _outDesiredFormat.mBytesPerPacket;
+  data->mBuffers->mData = _renderConvertData;
+  memset(_renderConvertData, 0, sizeof(_renderConvertData));
+
+  PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+    return 1;
+  }
+
+  return 0;
+}
+
+OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList* inputData,
+                                            const AudioTimeStamp* inputTime) {
+  OSStatus err = noErr;
+  UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
+  UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
+
+  // Check if we should close down audio device
+  // Double-checked locking optimization to remove locking overhead
+  if (_doStopRec) {
+    _critSect.Enter();
+    if (_doStopRec) {
+      // This will be signalled only when a shared device is not in use.
+      WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
+      WEBRTC_CA_LOG_WARN(
+          AudioDeviceDestroyIOProcID(_inputDeviceID, _inDeviceIOProcID));
+      if (err == noErr) {
+        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
+                     " Recording device stopped");
+      }
+
+      _doStopRec = false;
+      _stopEventRec.Set();
+      _critSect.Leave();
+      return 0;
+    }
+    _critSect.Leave();
+  }
+
+  if (!_recording) {
+    // Allow above checks to avoid a timeout on stopping capture.
+    return 0;
+  }
+
+  PaRingBufferSize bufSizeSamples =
+      PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
+
+  int32_t captureDelayUs =
+      static_cast<int32_t>(1e-3 * (nowNs - inputTimeNs) + 0.5);
+  captureDelayUs += static_cast<int32_t>((1.0e6 * bufSizeSamples) /
+                                             _inStreamFormat.mChannelsPerFrame /
+                                             _inStreamFormat.mSampleRate +
+                                         0.5);
+
+  AtomicSet32(&_captureDelayUs, captureDelayUs);
+
+  RTC_DCHECK(inputData->mNumberBuffers == 1);
+  PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize *
+                                _inStreamFormat.mChannelsPerFrame /
+                                _inStreamFormat.mBytesPerPacket;
+  PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
+                         numSamples);
+
+  kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
+  if (kernErr != KERN_SUCCESS) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 " semaphore_signal_all() error: %d", kernErr);
+  }
+
+  return err;
+}
+
+OSStatus AudioDeviceMac::implInConverterProc(UInt32* numberDataPackets,
+                                             AudioBufferList* data) {
+  RTC_DCHECK(data->mNumberBuffers == 1);
+  PaRingBufferSize numSamples =
+      *numberDataPackets * _inStreamFormat.mChannelsPerFrame;
+
+  while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
+      if (signal == 0) {
+        // The capture device is no longer alive; stop the worker thread.
+        *numberDataPackets = 0;
+        return 1;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_wait() error: %d", kernErr);
+    }
+  }
+
+  // Pass the read pointer directly to the converter to avoid a memcpy.
+  void* dummyPtr;
+  PaRingBufferSize dummySize;
+  PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
+                                  &data->mBuffers->mData, &numSamples,
+                                  &dummyPtr, &dummySize);
+  PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
+
+  data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
+  *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
+  data->mBuffers->mDataByteSize =
+      *numberDataPackets * _inStreamFormat.mBytesPerPacket;
+
+  return 0;
+}
+
+bool AudioDeviceMac::RunRender(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->RenderWorkerThread();
+}
+
+bool AudioDeviceMac::RenderWorkerThread() {
+  PaRingBufferSize numSamples =
+      ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
+  while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer) -
+             _renderDelayOffsetSamples <
+         numSamples) {
+    mach_timespec_t timeout;
+    timeout.tv_sec = 0;
+    timeout.tv_nsec = TIMER_PERIOD_MS;
+
+    kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
+    if (kernErr == KERN_OPERATION_TIMED_OUT) {
+      int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
+      if (signal == 0) {
+        // The render device is no longer alive; stop the worker thread.
+        return false;
+      }
+    } else if (kernErr != KERN_SUCCESS) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   " semaphore_timedwait() error: %d", kernErr);
+    }
+  }
+
+  int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
+
+  if (!_ptrAudioBuffer) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  capture AudioBuffer is invalid");
+    return false;
+  }
+
+  // Ask for new PCM data to be played out using the AudioDeviceBuffer.
+  uint32_t nSamples =
+      _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
+
+  nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
+  if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES) {
+    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                 "  invalid number of output samples(%d)", nSamples);
+  }
+
+  uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
+
+  SInt16* pPlayBuffer = (SInt16*)&playBuffer;
+  if (_macBookProPanRight && (_playChannels == 2)) {
+    // Mix entirely into the right channel and zero the left channel.
+    SInt32 sampleInt32 = 0;
+    for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx += 2) {
+      sampleInt32 = pPlayBuffer[sampleIdx];
+      sampleInt32 += pPlayBuffer[sampleIdx + 1];
+      sampleInt32 /= 2;
+
+      if (sampleInt32 > 32767) {
+        sampleInt32 = 32767;
+      } else if (sampleInt32 < -32768) {
+        sampleInt32 = -32768;
+      }
+
+      pPlayBuffer[sampleIdx] = 0;
+      pPlayBuffer[sampleIdx + 1] = static_cast<SInt16>(sampleInt32);
+    }
+  }
+
+  PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
+
+  return true;
+}
+
+bool AudioDeviceMac::RunCapture(void* ptrThis) {
+  return static_cast<AudioDeviceMac*>(ptrThis)->CaptureWorkerThread();
+}
+
+bool AudioDeviceMac::CaptureWorkerThread() {
+  OSStatus err = noErr;
+  UInt32 noRecSamples =
+      ENGINE_REC_BUF_SIZE_IN_SAMPLES * _inDesiredFormat.mChannelsPerFrame;
+  SInt16 recordBuffer[noRecSamples];
+  UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
+
+  AudioBufferList engineBuffer;
+  engineBuffer.mNumberBuffers = 1;  // Interleaved channels.
+  engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
+  engineBuffer.mBuffers->mDataByteSize =
+      _inDesiredFormat.mBytesPerPacket * noRecSamples;
+  engineBuffer.mBuffers->mData = recordBuffer;
+
+  err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
+                                        this, &size, &engineBuffer, NULL);
+  if (err != noErr) {
+    if (err == 1) {
+      // This is our own error.
+      return false;
+    } else {
+      logCAMsg(kTraceError, kTraceAudioDevice, _id,
+               "Error in AudioConverterFillComplexBuffer()", (const char*)&err);
+      return false;
+    }
+  }
+
+  // TODO(xians): what if the returned size is incorrect?
+  if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES) {
+    uint32_t currentMicLevel(0);
+    uint32_t newMicLevel(0);
+    int32_t msecOnPlaySide;
+    int32_t msecOnRecordSide;
+
+    int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
+    int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
+
+    msecOnPlaySide =
+        static_cast<int32_t>(1e-3 * (renderDelayUs + _renderLatencyUs) + 0.5);
+    msecOnRecordSide =
+        static_cast<int32_t>(1e-3 * (captureDelayUs + _captureLatencyUs) + 0.5);
+
+    if (!_ptrAudioBuffer) {
+      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+                   "  capture AudioBuffer is invalid");
+      return false;
+    }
+
+    // store the recorded buffer (no action will be taken if the
+    // #recorded samples is not a full buffer)
+    _ptrAudioBuffer->SetRecordedBuffer((int8_t*)&recordBuffer, (uint32_t)size);
+
+    if (AGC()) {
+      // Use mod to ensure we check the volume on the first pass.
+      if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
+        get_mic_volume_counter_ms_ = 0;
+        // store current mic level in the audio buffer if AGC is enabled
+        if (MicrophoneVolume(currentMicLevel) == 0) {
+          // this call does not affect the actual microphone volume
+          _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
         }
-    }
-
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
-
-    int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
-        + 0.5);
-    renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-        / _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
-        + 0.5);
-
-    AtomicSet32(&_renderDelayUs, renderDelayUs);
-
-    return 0;
-}
-
-OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
-                                              AudioBufferList *data)
-{
-  RTC_DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _outDesiredFormat.mChannelsPerFrame;
-
-    data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
-    // Always give the converter as much as it wants, zero padding as required.
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _outDesiredFormat.mBytesPerPacket;
-    data->mBuffers->mData = _renderConvertData;
-    memset(_renderConvertData, 0, sizeof(_renderConvertData));
-
-    PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
-
-    kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
-        return 1;
-    }
-
-    return 0;
-}
-
-OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
-                                            const AudioTimeStamp *inputTime)
-{
-    OSStatus err = noErr;
-    UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
-    UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
-    // Check if we should close down audio device
-    // Double-checked locking optimization to remove locking overhead
-    if (_doStopRec)
-    {
-        _critSect.Enter();
-        if (_doStopRec)
-        {
-            // This will be signalled only when a shared device is not in use.
-            WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
-            WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
-                                                          _inDeviceIOProcID));
-            if (err == noErr)
-            {
-                WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                             _id, " Recording device stopped");
-            }
-
-            _doStopRec = false;
-            _stopEventRec.Set();
-            _critSect.Leave();
-            return 0;
+      }
+      get_mic_volume_counter_ms_ += kBufferSizeMs;
+    }
+
+    _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
+
+    _ptrAudioBuffer->SetTypingStatus(KeyPressed());
+
+    // deliver recorded samples at specified sample rate, mic level etc.
+    // to the observer using callback
+    _ptrAudioBuffer->DeliverRecordedData();
+
+    if (AGC()) {
+      newMicLevel = _ptrAudioBuffer->NewMicLevel();
+      if (newMicLevel != 0) {
+        // The VQE will only deliver non-zero microphone levels when
+        // a change is needed.
+        // Set this new mic level (received from the observer as return
+        // value in the callback).
+        WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
+                     "  AGC change of volume: old=%u => new=%u",
+                     currentMicLevel, newMicLevel);
+        if (SetMicrophoneVolume(newMicLevel) == -1) {
+          WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
+                       "  the required modification of the microphone "
+                       "volume failed");
         }
-        _critSect.Leave();
-    }
-
-    if (!_recording)
-    {
-        // Allow above checks to avoid a timeout on stopping capture.
-        return 0;
-    }
-
-    PaRingBufferSize bufSizeSamples =
-        PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
-
-    int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
-        + 0.5);
-    captureDelayUs
-        += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
-            / _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
-            + 0.5);
-
-    AtomicSet32(&_captureDelayUs, captureDelayUs);
-
-    RTC_DCHECK(inputData->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize
-        * _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
-    PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
-                           numSamples);
-
-    kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
-    if (kernErr != KERN_SUCCESS)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     " semaphore_signal_all() error: %d", kernErr);
-    }
-
-    return err;
-}
-
-OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
-                                             AudioBufferList *data)
-{
-  RTC_DCHECK(data->mNumberBuffers == 1);
-    PaRingBufferSize numSamples = *numberDataPackets
-        * _inStreamFormat.mChannelsPerFrame;
-
-    while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The capture device is no longer alive; stop the worker thread.
-                *numberDataPackets = 0;
-                return 1;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_wait() error: %d", kernErr);
-        }
-    }
-
-    // Pass the read pointer directly to the converter to avoid a memcpy.
-    void* dummyPtr;
-    PaRingBufferSize dummySize;
-    PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
-                                    &data->mBuffers->mData, &numSamples,
-                                    &dummyPtr, &dummySize);
-    PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
-
-    data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
-    *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
-    data->mBuffers->mDataByteSize = *numberDataPackets
-        * _inStreamFormat.mBytesPerPacket;
-
-    return 0;
-}
-
-bool AudioDeviceMac::RunRender(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
-}
-
-bool AudioDeviceMac::RenderWorkerThread()
-{
-    PaRingBufferSize numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
-        * _outDesiredFormat.mChannelsPerFrame;
-    while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
-        - _renderDelayOffsetSamples < numSamples)
-    {
-        mach_timespec_t timeout;
-        timeout.tv_sec = 0;
-        timeout.tv_nsec = TIMER_PERIOD_MS;
-
-        kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
-        if (kernErr == KERN_OPERATION_TIMED_OUT)
-        {
-            int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
-            if (signal == 0)
-            {
-                // The render device is no longer alive; stop the worker thread.
-                return false;
-            }
-        } else if (kernErr != KERN_SUCCESS)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         " semaphore_timedwait() error: %d", kernErr);
-        }
-    }
-
-    int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
-
-    if (!_ptrAudioBuffer)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  capture AudioBuffer is invalid");
-        return false;
-    }
-
-    // Ask for new PCM data to be played out using the AudioDeviceBuffer.
-    uint32_t nSamples =
-        _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
-
-    nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
-    if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
-    {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  invalid number of output samples(%d)", nSamples);
-    }
-
-    uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
-
-    SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
-    if (_macBookProPanRight && (_playChannels == 2))
-    {
-        // Mix entirely into the right channel and zero the left channel.
-        SInt32 sampleInt32 = 0;
-        for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
-            += 2)
-        {
-            sampleInt32 = pPlayBuffer[sampleIdx];
-            sampleInt32 += pPlayBuffer[sampleIdx + 1];
-            sampleInt32 /= 2;
-
-            if (sampleInt32 > 32767)
-            {
-                sampleInt32 = 32767;
-            } else if (sampleInt32 < -32768)
-            {
-                sampleInt32 = -32768;
-            }
-
-            pPlayBuffer[sampleIdx] = 0;
-            pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
-        }
-    }
-
-    PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
-
-    return true;
-}
-
-bool AudioDeviceMac::RunCapture(void* ptrThis)
-{
-    return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
-}
-
-bool AudioDeviceMac::CaptureWorkerThread()
-{
-    OSStatus err = noErr;
-    UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
-        * _inDesiredFormat.mChannelsPerFrame;
-    SInt16 recordBuffer[noRecSamples];
-    UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
-
-    AudioBufferList engineBuffer;
-    engineBuffer.mNumberBuffers = 1; // Interleaved channels.
-    engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
-    engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
-        * noRecSamples;
-    engineBuffer.mBuffers->mData = recordBuffer;
-
-    err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
-                                          this, &size, &engineBuffer, NULL);
-    if (err != noErr)
-    {
-        if (err == 1)
-        {
-            // This is our own error.
-            return false;
-        } else
-        {
-            logCAMsg(kTraceError, kTraceAudioDevice, _id,
-                     "Error in AudioConverterFillComplexBuffer()",
-                     (const char *) &err);
-            return false;
-        }
-    }
-
-    // TODO(xians): what if the returned size is incorrect?
-    if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
-    {
-        uint32_t currentMicLevel(0);
-        uint32_t newMicLevel(0);
-        int32_t msecOnPlaySide;
-        int32_t msecOnRecordSide;
-
-        int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
-        int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-
-        msecOnPlaySide = static_cast<int32_t> (1e-3 * (renderDelayUs +
-                                                       _renderLatencyUs) + 0.5);
-        msecOnRecordSide = static_cast<int32_t> (1e-3 * (captureDelayUs +
-                                                         _captureLatencyUs) +
-                                                 0.5);
-
-        if (!_ptrAudioBuffer)
-        {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "  capture AudioBuffer is invalid");
-            return false;
-        }
-
-        // store the recorded buffer (no action will be taken if the
-        // #recorded samples is not a full buffer)
-        _ptrAudioBuffer->SetRecordedBuffer((int8_t*) &recordBuffer,
-                                           (uint32_t) size);
-
-        if (AGC())
-        {
-            // Use mod to ensure we check the volume on the first pass.
-            if (get_mic_volume_counter_ms_ % kGetMicVolumeIntervalMs == 0) {
-                get_mic_volume_counter_ms_ = 0;
-                // store current mic level in the audio buffer if AGC is enabled
-                if (MicrophoneVolume(currentMicLevel) == 0)
-                {
-                    // this call does not affect the actual microphone volume
-                    _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
-                }
-            }
-            get_mic_volume_counter_ms_ += kBufferSizeMs;
-        }
-
-        _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
-
-        _ptrAudioBuffer->SetTypingStatus(KeyPressed());
-
-        // deliver recorded samples at specified sample rate, mic level etc.
-        // to the observer using callback
-        _ptrAudioBuffer->DeliverRecordedData();
-
-        if (AGC())
-        {
-            newMicLevel = _ptrAudioBuffer->NewMicLevel();
-            if (newMicLevel != 0)
-            {
-                // The VQE will only deliver non-zero microphone levels when
-                // a change is needed.
-                // Set this new mic level (received from the observer as return
-                // value in the callback).
-                WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
-                             _id, "  AGC change of volume: old=%u => new=%u",
-                             currentMicLevel, newMicLevel);
-                if (SetMicrophoneVolume(newMicLevel) == -1)
-                {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  the required modification of the microphone "
-                                 "volume failed");
-                }
-            }
-        }
-    }
-
-    return true;
+      }
+    }
+  }
+
+  return true;
 }
 
 bool AudioDeviceMac::KeyPressed() {
   bool key_down = false;
   // Loop through all Mac virtual key constant values.
-  for (unsigned int key_index = 0;
-                    key_index < arraysize(prev_key_state_);
-                    ++key_index) {
-    bool keyState = CGEventSourceKeyState(
-                             kCGEventSourceStateHIDSystemState,
-                             key_index);
+  for (unsigned int key_index = 0; key_index < arraysize(prev_key_state_);
+       ++key_index) {
+    bool keyState =
+        CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, key_index);
     // A false -> true change in keymap means a key is pressed.
     key_down |= (keyState && !prev_key_state_[key_index]);
     // Save current state.
     prev_key_state_[key_index] = keyState;
   }
   return key_down;
 }
 }  // namespace webrtc