Cleanup of iOS AudioDevice implementation

webrtc/modules/audio_device/ios/audio_device_ios.mm

Index: webrtc/modules/audio_device/ios/audio_device_ios.mm
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.mm b/webrtc/modules/audio_device/ios/audio_device_ios.mm
index 47503a96d70546ef07b822627f6ee69548bdc379..ecb56946f9e2d41151ea06993cda8498ca956f51 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.mm
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.mm
@@ -8,1905 +8,1029 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
 #import <AVFoundation/AVFoundation.h>
 #import <Foundation/Foundation.h>
 
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
+#include "webrtc/modules/utility/interface/helpers_ios.h"
 
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
 #include "webrtc/system_wrappers/interface/trace.h"
 
 namespace webrtc {
-AudioDeviceIOS::AudioDeviceIOS(const int32_t id)
-    :
-    _ptrAudioBuffer(NULL),
-    _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-    _id(id),
-    _auVoiceProcessing(NULL),
-    _audioInterruptionObserver(NULL),
-    _initialized(false),
-    _isShutDown(false),
-    _recording(false),
-    _playing(false),
-    _recIsInitialized(false),
-    _playIsInitialized(false),
-    _recordingDeviceIsSpecified(false),
-    _playoutDeviceIsSpecified(false),
-    _micIsInitialized(false),
-    _speakerIsInitialized(false),
-    _AGC(false),
-    _adbSampFreq(0),
-    _recordingDelay(0),
-    _playoutDelay(0),
-    _playoutDelayMeasurementCounter(9999),
-    _recordingDelayHWAndOS(0),
-    _recordingDelayMeasurementCounter(9999),
-    _playWarning(0),
-    _playError(0),
-    _recWarning(0),
-    _recError(0),
-    _playoutBufferUsed(0),
-    _recordingCurrentSeq(0),
-    _recordingBufferTotalSize(0) {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
-                 "%s created", __FUNCTION__);
-
-    memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-    memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
-    memset(_recordingLength, 0, sizeof(_recordingLength));
-    memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
-}
 
-AudioDeviceIOS::~AudioDeviceIOS() {
-    WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
-                 "%s destroyed", __FUNCTION__);
+#define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
 
-    Terminate();
+using ios::CheckAndLogError;
 
-    delete &_critSect;
+static void ActivateAudioSession(AVAudioSession* session, bool activate) {
+  LOG(LS_INFO) << "ActivateAudioSession(" << activate << ")";
+  @autoreleasepool {
+    NSError* errorObject = nil;

[tkchin_webrtc, 2015/07/08 19:41:14]

nit: usually just
NSError* error = nil;

[henrika_webrtc, 2015/07/09 12:58:01]

Done.

+    BOOL success = NO;
+    if (!activate) {
+      // Deactivate the audio session.
+      success = [session setActive:NO error:&errorObject];
+      DCHECK(CheckAndLogError(success, errorObject));
+      return;
+    }
+    // Activate an audio session and set category and mode. Only make changes
+    // if needed since setting them to the value they already have will clear
+    // transient properties (such as PortOverride) that some other component
+    // have set up.
+    if (session.category != AVAudioSessionCategoryPlayAndRecord) {
+      success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
+                               error:&errorObject];

[tkchin_webrtc, 2015/07/08 19:41:14]

nit: I think it is good practice to explicitly set error = nil before reusing,
here and below.

[henrika_webrtc, 2015/07/09 12:58:02]

In fact, I tried CheckAndLogError(BOOL success, NSError** error) to be able to
set *error to nil inside and then call with CheckAndLogError(success, &error),
but for some reason linking failed then. The goal was to avoid adding several
lines of "error = nil;" lines. 

+      DCHECK(CheckAndLogError(success, errorObject));
+    }
+    if (session.mode != AVAudioSessionModeVoiceChat) {
+      success =
+          [session setMode:AVAudioSessionModeVoiceChat error:&errorObject];
+      DCHECK(CheckAndLogError(success, errorObject));
+    }
+    success = [session setActive:YES error:&errorObject];
+    DCHECK(CheckAndLogError(success, errorObject));
+    // Ensure that category and mode are actually activated.
+    DCHECK(
+        [session.category isEqualToString:AVAudioSessionCategoryPlayAndRecord]);
+    DCHECK([session.mode isEqualToString:AVAudioSessionModeVoiceChat]);
+  }
+}
+
+// Query hardware characteristics, such as input and output latency, input and
+// output channel count, hardware sample rate, hardware volume setting, and
+// whether audio input is available. To obtain meaningful values for hardware
+// characteristics,the audio session must be initialized and active before we
+// query the values.
+// TODO(henrika): Note that these characteristics can change at runtime. For
+// instance, input sample rate may change when a user plugs in a headset.
+static void GetHardwareAudioParameters(AudioParameters* playout_parameters,
+                                       AudioParameters* record_parameters) {
+  LOG(LS_INFO) << "GetHardwareAudioParameters";
+  @autoreleasepool {
+    // Implicit initialization happens when we obtain a reference to the
+    // AVAudioSession object.
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    // Always get values when the audio session is active.
+    ActivateAudioSession(session, true);
+    CHECK(session.isInputAvailable) << "No input path is available!";
+    // Get current hardware parameters.
+    double sample_rate = (double)session.sampleRate;
+    double io_buffer_duration = (double)session.IOBufferDuration;
+    int output_channels = (int)session.outputNumberOfChannels;
+    int input_channels = (int)session.inputNumberOfChannels;
+    int frames_per_buffer =
+        static_cast<int>(sample_rate * io_buffer_duration + 0.5);
+    // Copy hardware parameters to output parameters.
+    playout_parameters->reset(sample_rate, output_channels, frames_per_buffer);
+    record_parameters->reset(sample_rate, input_channels, frames_per_buffer);
+    // Add logging for debugging purposes.
+    LOG(LS_INFO) << " sample rate: " << sample_rate;
+    LOG(LS_INFO) << " IO buffer duration: " << io_buffer_duration;
+    LOG(LS_INFO) << " frames_per_buffer: " << frames_per_buffer;
+    LOG(LS_INFO) << " output channels: " << output_channels;
+    LOG(LS_INFO) << " input channels: " << input_channels;
+    LOG(LS_INFO) << " output latency: " << (double)[session outputLatency];
+    LOG(LS_INFO) << " input latency: " << (double)[session inputLatency];
+    // Don't keep the audio session active. Instead, deactivate when needed.
+    ActivateAudioSession(session, false);
+    // TODO(henrika): to be extra safe, we can do more here. E.g., set
+    // preferred values for sample rate, channels etc., re-activate an audio
+    // session and verify the actual values again. Then we know for sure that
+    // the current values will in fact be correct. Or, we can skip all this
+    // and check setting when audio is started. Probably better.
+  }
+}
+
+AudioDeviceIOS::AudioDeviceIOS()
+    : audio_device_buffer_(nullptr),
+      _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
+      _auVoiceProcessing(nullptr),
+      _audioInterruptionObserver(nullptr),
+      _initialized(false),
+      _isShutDown(false),
+      _recording(false),
+      _playing(false),
+      _recIsInitialized(false),
+      _playIsInitialized(false),
+      _adbSampFreq(0),
+      _recordingDelay(0),
+      _playoutDelay(0),
+      _playoutDelayMeasurementCounter(9999),
+      _recordingDelayHWAndOS(0),
+      _recordingDelayMeasurementCounter(9999),
+      _playoutBufferUsed(0),
+      _recordingCurrentSeq(0),
+      _recordingBufferTotalSize(0) {
+  LOGI() << "ctor" << ios::GetCurrentThreadDescription();
+  memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
+  memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
+  memset(_recordingLength, 0, sizeof(_recordingLength));
+  memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
 }
 
-
-// ============================================================================
-//                                     API
-// ============================================================================
-
-void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    _ptrAudioBuffer = audioBuffer;
-
-    // inform the AudioBuffer about default settings for this implementation
-    _ptrAudioBuffer->SetRecordingSampleRate(ENGINE_REC_BUF_SIZE_IN_SAMPLES);
-    _ptrAudioBuffer->SetPlayoutSampleRate(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
-    _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
-    _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
+AudioDeviceIOS::~AudioDeviceIOS() {
+  LOGI() << "~dtor";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  Terminate();
+  delete &_critSect;
 }
 
-int32_t AudioDeviceIOS::ActiveAudioLayer(
-    AudioDeviceModule::AudioLayer& audioLayer) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-    return 0;
+void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  LOGI() << "AttachAudioBuffer";
+  DCHECK(audioBuffer);
+  DCHECK(thread_checker_.CalledOnValidThread());
+  audio_device_buffer_ = audioBuffer;
 }
 
 int32_t AudioDeviceIOS::Init() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (_initialized) {
-        return 0;
-    }
-
-    _isShutDown = false;
-
-    // Create and start capture thread
-    if (!_captureWorkerThread) {
-        _captureWorkerThread = ThreadWrapper::CreateThread(
-            RunCapture, this, "CaptureWorkerThread");
-        bool res = _captureWorkerThread->Start();
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "CaptureWorkerThread started (res=%d)", res);
-        _captureWorkerThread->SetPriority(kRealtimePriority);
-    } else {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                     _id, "Thread already created");
-    }
-    _playWarning = 0;
-    _playError = 0;
-    _recWarning = 0;
-    _recError = 0;
-
-    _initialized = true;
-
+  LOGI() << "Init";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (_initialized) {
     return 0;
+  }
+  // Query hardware audio parameters and cache the results. These parameters
+  // will be used as preferred values later when streaming starts.
+  // Note that I override these "optimal" value below since I don't want to
+  // modify the existing behavior yet.
+  GetHardwareAudioParameters(&playout_parameters_, &record_parameters_);
+  // TODO(henrika): these parameters are currently hard coded to match the
+  // existing implementation where we always use 16kHz as preferred sample
+  // rate and mono only. Goal is to improve this scheme and make it more
+  // flexible. In addition, a better native buffer size shall be derived.
+  // Using 10ms as default here (only used by unit test so far).
+  // We should also implemented observers for notification of any change in
+  // these parameters.
+  playout_parameters_.reset(16000, 1, 160);
+  record_parameters_.reset(16000, 1, 160);
+
+  // AttachAudioBuffer() is called at construction by the main class but check
+  // just in case.
+  DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
+  // Inform the audio device buffer (ADB) about the new audio format.
+  // TODO(henrika): try to improve this section.
+  audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
+  audio_device_buffer_->SetRecordingSampleRate(
+      record_parameters_.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
+
+  DCHECK(!_captureWorkerThread);
+  // Create and start the capture thread.
+  // TODO(henrika): do we need this thread?
+  _isShutDown = false;
+  _captureWorkerThread =
+      ThreadWrapper::CreateThread(RunCapture, this, "CaptureWorkerThread");
+  if (!_captureWorkerThread->Start()) {
+    LOG_F(LS_ERROR) << "Failed to start CaptureWorkerThread!";
+    return -1;
+  }
+  _captureWorkerThread->SetPriority(kRealtimePriority);
+  _initialized = true;
+  return 0;
 }
 
 int32_t AudioDeviceIOS::Terminate() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    if (!_initialized) {
-        return 0;
-    }
-
-
-    // Stop capture thread
-    if (_captureWorkerThread) {
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "Stopping CaptureWorkerThread");
-        bool res = _captureWorkerThread->Stop();
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
-                     _id, "CaptureWorkerThread stopped (res=%d)", res);
-        _captureWorkerThread.reset();
-    }
-
-    // Shut down Audio Unit
-    ShutdownPlayOrRecord();
-
-    _isShutDown = true;
-    _initialized = false;
-    _speakerIsInitialized = false;
-    _micIsInitialized = false;
-    _playoutDeviceIsSpecified = false;
-    _recordingDeviceIsSpecified = false;
+  LOGI() << "Terminate";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (!_initialized) {
     return 0;
-}
-
-bool AudioDeviceIOS::Initialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return (_initialized);
-}
-
-int32_t AudioDeviceIOS::InitSpeaker() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Not initialized");
-        return -1;
-    }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Cannot init speaker when playing");
-        return -1;
-    }
-
-    if (!_playoutDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Playout device is not specified");
-        return -1;
+  }
+  // Stop the capture thread.
+  if (_captureWorkerThread) {
+    if (!_captureWorkerThread->Stop()) {
+      LOG_F(LS_ERROR) << "Failed to stop CaptureWorkerThread!";
+      return -1;
     }
-
-    // Do nothing
-    _speakerIsInitialized = true;
-
-    return 0;
+    _captureWorkerThread.reset();
+  }
+  ShutdownPlayOrRecord();
+  _isShutDown = true;
+  _initialized = false;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::InitMicrophone() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Not initialized");
-        return -1;
-    }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Cannot init mic when recording");
-        return -1;
-    }
-
-    if (!_recordingDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
-                     _id, "  Recording device is not specified");
-        return -1;
+int32_t AudioDeviceIOS::InitPlayout() {
+  LOGI() << "InitPlayout";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_initialized);
+  DCHECK(!_playIsInitialized);
+  DCHECK(!_playing);
+  if (!_recIsInitialized) {
+    if (InitPlayOrRecord() == -1) {
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
+      return -1;
     }
-
-    // Do nothing
-
-    _micIsInitialized = true;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::SpeakerIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return _speakerIsInitialized;
-}
-
-bool AudioDeviceIOS::MicrophoneIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return _micIsInitialized;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Speaker volume not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetSpeakerVolume(volume=%u)", volume);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::SetWaveOutVolume(uint16_t volumeLeft,
-                                     uint16_t volumeRight) {
-    WEBRTC_TRACE(
-        kTraceModuleCall,
-        kTraceAudioDevice,
-        _id,
-        "AudioDeviceIOS::SetWaveOutVolume(volumeLeft=%u, volumeRight=%u)",
-        volumeLeft, volumeRight);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-
-    return -1;
-}
-
-int32_t
-AudioDeviceIOS::WaveOutVolume(uint16_t& /*volumeLeft*/,
-                                 uint16_t& /*volumeRight*/) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MinSpeakerVolume(
-    uint32_t& minVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::SpeakerVolumeStepSize(uint16_t& stepSize) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Speaker mute not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic mute not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+  }
+  _playIsInitialized = true;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::MicrophoneBoostIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic boost not supported on iOS
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::SetMicrophoneBoost(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetMicrophoneBoost(enable=%u)", enable);
-
-    if (!_micIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Microphone not initialized");
-        return -1;
-    }
-
-    if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetMicrophoneBoost cannot be enabled on this platform");
-        return -1;
+int32_t AudioDeviceIOS::InitRecording() {
+  LOGI() << "InitPlayout";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_initialized);
+  DCHECK(!_recIsInitialized);
+  DCHECK(!_recording);
+  if (!_playIsInitialized) {
+    if (InitPlayOrRecord() == -1) {
+      LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
+      return -1;
     }
-
-    return 0;
+  }
+  _recIsInitialized = true;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::MicrophoneBoost(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    if (!_micIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Microphone not initialized");
-        return -1;
+int32_t AudioDeviceIOS::StartPlayout() {
+  LOGI() << "StartPlayout";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_playIsInitialized);
+  DCHECK(!_playing);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
+  _playoutBufferUsed = 0;
+  _playoutDelay = 0;
+  // Make sure first call to update delay function will update delay
+  _playoutDelayMeasurementCounter = 9999;
+
+  if (!_recording) {
+    OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+      return -1;
     }
-
-    enabled = false;
-
-    return 0;
+  }
+  _playing = true;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+int32_t AudioDeviceIOS::StopPlayout() {
+  LOGI() << "StopPlayout";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (!_playIsInitialized || !_playing) {
+    return 0;
+  }
 
-    available = false;  // Stereo recording not supported on iOS
+  CriticalSectionScoped lock(&_critSect);
 
-    return 0;
+  if (!_recording) {
+    // Both playout and recording has stopped, shutdown the device.
+    ShutdownPlayOrRecord();
+  }
+  _playIsInitialized = false;
+  _playing = false;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetStereoRecording(enable=%u)", enable);
-
-    if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Stereo recording is not supported on this platform");
-        return -1;
+int32_t AudioDeviceIOS::StartRecording() {
+  LOGI() << "StartRecording";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_recIsInitialized);
+  DCHECK(!_recording);
+
+  CriticalSectionScoped lock(&_critSect);
+
+  memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
+  memset(_recordingLength, 0, sizeof(_recordingLength));
+  memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
+
+  _recordingCurrentSeq = 0;
+  _recordingBufferTotalSize = 0;
+  _recordingDelay = 0;
+  _recordingDelayHWAndOS = 0;
+  // Make sure first call to update delay function will update delay
+  _recordingDelayMeasurementCounter = 9999;
+
+  if (!_playing) {
+    OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
+    if (result != noErr) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+      return -1;
     }
-    return 0;
+  }
+  _recording = true;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    enabled = false;
+int32_t AudioDeviceIOS::StopRecording() {
+  LOGI() << "StopRecording";
+  DCHECK(thread_checker_.CalledOnValidThread());
+  if (!_recIsInitialized || !_recording) {
     return 0;
-}
+  }
 
-int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+  CriticalSectionScoped lock(&_critSect);
 
-    available = false;  // Stereo playout not supported on iOS
-
-    return 0;
+  if (!_playing) {
+    // Both playout and recording has stopped, shutdown the device.
+    ShutdownPlayOrRecord();
+  }
+  _recIsInitialized = false;
+  _recording = false;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetStereoPlayout(enable=%u)", enable);
-
+// Change the default receiver playout route to speaker.
+int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) {
+  LOGI() << "SetLoudspeakerStatus(" << enable << ")";
+
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  NSString* category = session.category;
+  AVAudioSessionCategoryOptions options = session.categoryOptions;
+  // Respect old category options if category is
+  // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options
+  // might not be valid for this category.
+  if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
     if (enable) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     " Stereo playout is not supported on this platform");
-        return -1;
+      options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
+    } else {
+      options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker;
     }
-    return 0;
+  } else {
+    options = AVAudioSessionCategoryOptionDefaultToSpeaker;
+  }
+  NSError* error = nil;
+  BOOL success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
+                          withOptions:options
+                                error:&error];
+  ios::CheckAndLogError(success, error);
+  return (error == nil) ? 0 : -1;
 }
 
-int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    enabled = false;
-    return 0;
+int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool& enabled) const {
+  LOGI() << "GetLoudspeakerStatus";
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  AVAudioSessionCategoryOptions options = session.categoryOptions;
+  enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::SetAGC(bool enable) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetAGC(enable=%d)", enable);
-
-    _AGC = enable;
-
-    return 0;
+int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
+  delayMS = _playoutDelay;
+  return 0;
 }
 
-bool AudioDeviceIOS::AGC() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    return _AGC;
+int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
+  delayMS = _recordingDelay;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-
-    available = false;  // Mic volume not supported on IOS
-
-    return 0;
+int32_t AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type,
+                                      uint16_t& sizeMS) const {
+  type = AudioDeviceModule::kAdaptiveBufferSize;
+  sizeMS = _playoutDelay;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetMicrophoneVolume(volume=%u)", volume);
+int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
+  CHECK(playout_parameters_.is_valid());
+  DCHECK(thread_checker_.CalledOnValidThread());
+  *params = playout_parameters_;
+  return 0;
+}
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
+int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
+  CHECK(record_parameters_.is_valid());
+  DCHECK(thread_checker_.CalledOnValidThread());
+  *params = record_parameters_;
+  return 0;
 }
 
-int32_t
-    AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+// ============================================================================
+//                                 Private Methods
+// ============================================================================
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
+int32_t AudioDeviceIOS::InitPlayOrRecord() {
+  LOGI() << "AudioDeviceIOS::InitPlayOrRecord";
+  DCHECK(!_auVoiceProcessing);
 
-int32_t
-    AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+  OSStatus result = -1;
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
+  // Create Voice Processing Audio Unit
+  AudioComponentDescription desc;
+  AudioComponent comp;
 
-int32_t
-    AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+  desc.componentType = kAudioUnitType_Output;
+  desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
+  desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+  desc.componentFlags = 0;
+  desc.componentFlagsMask = 0;
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
+  comp = AudioComponentFindNext(nullptr, &desc);
+  if (nullptr == comp) {
+    LOG_F(LS_ERROR) << "Could not find audio component for Audio Unit";
     return -1;
-}
-
-int32_t
-    AudioDeviceIOS::MicrophoneVolumeStepSize(
-                                            uint16_t& stepSize) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+  }
 
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
+  result = AudioComponentInstanceNew(comp, &_auVoiceProcessing);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to create Audio Unit instance: " << result;
     return -1;
-}
+  }
+
+  // TODO(henrika): I think we should set the preferred channel configuration
+  // in both directions as well to be safe.
+
+  // Set preferred hardware sample rate to 16 kHz.
+  // TODO(henrika): improve this selection of sample rate. Why do we currently
+  // use a hard coded value? How can we fail and still continue?
+  NSError* error = nil;
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  Float64 preferredSampleRate(playout_parameters_.sample_rate());
+  [session setPreferredSampleRate:preferredSampleRate error:&error];
+  if (error != nil) {
+    const char* errorString = [[error localizedDescription] UTF8String];
+    LOG_F(LS_ERROR) << "setPreferredSampleRate failed: " << errorString;
+  }
+
+  // TODO(henrika): we can reduce latency by setting the IOBufferDuration
+  // here. Default size for 16kHz is 0.016 sec or 16 msec on an iPhone 6.
+
+  // Activate the audio session.
+  ActivateAudioSession(session, true);
+
+  UInt32 enableIO = 1;
+  result = AudioUnitSetProperty(_auVoiceProcessing,
+                                kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Input,
+                                1,  // input bus
+                                &enableIO, sizeof(enableIO));
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to enable IO on input: " << result;
+  }
+
+  result = AudioUnitSetProperty(_auVoiceProcessing,
+                                kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Output,
+                                0,  // output bus
+                                &enableIO, sizeof(enableIO));
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to enable IO on output: " << result;
+  }
+
+  // Disable AU buffer allocation for the recorder, we allocate our own.
+  // TODO(henrika): understand this part better.
+  UInt32 flag = 0;
+  result = AudioUnitSetProperty(_auVoiceProcessing,
+                                kAudioUnitProperty_ShouldAllocateBuffer,
+                                kAudioUnitScope_Output, 1, &flag, sizeof(flag));
+  if (0 != result) {
+    LOG_F(LS_WARNING) << "Failed to disable AU buffer allocation: " << result;
+    // Should work anyway
+  }
+
+  // Set recording callback.
+  AURenderCallbackStruct auCbS;
+  memset(&auCbS, 0, sizeof(auCbS));
+  auCbS.inputProc = RecordProcess;
+  auCbS.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      _auVoiceProcessing, kAudioOutputUnitProperty_SetInputCallback,
+      kAudioUnitScope_Global, 1, &auCbS, sizeof(auCbS));
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to set AU record callback: " << result;
+  }
+
+  // Set playout callback.
+  memset(&auCbS, 0, sizeof(auCbS));
+  auCbS.inputProc = PlayoutProcess;
+  auCbS.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      _auVoiceProcessing, kAudioUnitProperty_SetRenderCallback,
+      kAudioUnitScope_Global, 0, &auCbS, sizeof(auCbS));
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to set AU output callback: " << result;
+  }
+
+  // Get stream format for out/0
+  AudioStreamBasicDescription playoutDesc;
+  UInt32 size = sizeof(playoutDesc);
+  result =
+      AudioUnitGetProperty(_auVoiceProcessing, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Output, 0, &playoutDesc, &size);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to get AU output stream format: " << result;
+  }
+
+  playoutDesc.mSampleRate = preferredSampleRate;
+  LOG(LS_INFO) << "Audio Unit playout opened in sampling rate: "
+               << playoutDesc.mSampleRate;
+
+  // Store the sampling frequency to use towards the Audio Device Buffer
+  // todo: Add 48 kHz (increase buffer sizes). Other fs?
+  // TODO(henrika): Figure out if we really need this complex handling.
+  if ((playoutDesc.mSampleRate > 44090.0) &&
+      (playoutDesc.mSampleRate < 44110.0)) {
+    _adbSampFreq = 44100;
+  } else if ((playoutDesc.mSampleRate > 15990.0) &&
+             (playoutDesc.mSampleRate < 16010.0)) {
+    _adbSampFreq = 16000;
+  } else if ((playoutDesc.mSampleRate > 7990.0) &&
+             (playoutDesc.mSampleRate < 8010.0)) {
+    _adbSampFreq = 8000;
+  } else {
+    _adbSampFreq = 0;
+    FATAL() << "Invalid sample rate";
+  }
+
+  // Set the audio device buffer sampling rates (use same for play and record).
+  // TODO(henrika): this is not a good place to set these things up.
+  DCHECK(audio_device_buffer_);
+  DCHECK_EQ(_adbSampFreq, playout_parameters_.sample_rate());
+  audio_device_buffer_->SetRecordingSampleRate(_adbSampFreq);
+  audio_device_buffer_->SetPlayoutSampleRate(_adbSampFreq);
+
+  // Set stream format for out/0.
+  playoutDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
+                             kLinearPCMFormatFlagIsPacked |
+                             kLinearPCMFormatFlagIsNonInterleaved;
+  playoutDesc.mBytesPerPacket = 2;
+  playoutDesc.mFramesPerPacket = 1;
+  playoutDesc.mBytesPerFrame = 2;
+  playoutDesc.mChannelsPerFrame = 1;
+  playoutDesc.mBitsPerChannel = 16;
+  result =
+      AudioUnitSetProperty(_auVoiceProcessing, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Input, 0, &playoutDesc, size);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to set AU stream format for out/0";
+  }
+
+  // Get stream format for in/1.
+  AudioStreamBasicDescription recordingDesc;
+  size = sizeof(recordingDesc);
+  result =
+      AudioUnitGetProperty(_auVoiceProcessing, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Input, 1, &recordingDesc, &size);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to get AU stream format for in/1";
+  }
+
+  recordingDesc.mSampleRate = preferredSampleRate;
+  LOG(LS_INFO) << "Audio Unit recording opened in sampling rate: "
+               << recordingDesc.mSampleRate;
+
+  // Set stream format for out/1 (use same sampling frequency as for in/1).
+  recordingDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
+                               kLinearPCMFormatFlagIsPacked |
+                               kLinearPCMFormatFlagIsNonInterleaved;
+  recordingDesc.mBytesPerPacket = 2;
+  recordingDesc.mFramesPerPacket = 1;
+  recordingDesc.mBytesPerFrame = 2;
+  recordingDesc.mChannelsPerFrame = 1;
+  recordingDesc.mBitsPerChannel = 16;
+  result =
+      AudioUnitSetProperty(_auVoiceProcessing, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Output, 1, &recordingDesc, size);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "Failed to set AU stream format for out/1";
+  }
+
+  // Initialize here already to be able to get/set stream properties.
+  result = AudioUnitInitialize(_auVoiceProcessing);
+  if (0 != result) {
+    LOG_F(LS_ERROR) << "AudioUnitInitialize failed: " << result;
+  }
+
+  // Get hardware sample rate for logging (see if we get what we asked for).
+  // TODO(henrika): what if we don't get what we ask for?
+  double sampleRate = session.sampleRate;
+  LOG(LS_INFO) << "Current HW sample rate is: " << sampleRate
+               << ", ADB sample rate is: " << _adbSampFreq;
+  LOG(LS_INFO) << "Current HW IO buffer size is: " <<
+      [session IOBufferDuration];
+
+  // Listen to audio interruptions.
+  // TODO(henrika): learn this area better.
+  NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
+  id observer = [center
+      addObserverForName:AVAudioSessionInterruptionNotification
+                  object:nil
+                   queue:[NSOperationQueue mainQueue]
+              usingBlock:^(NSNotification* notification) {
+                NSNumber* typeNumber =
+                    [notification userInfo][AVAudioSessionInterruptionTypeKey];
+                AVAudioSessionInterruptionType type =
+                    (AVAudioSessionInterruptionType)[typeNumber
+                                                         unsignedIntegerValue];
+                switch (type) {
+                  case AVAudioSessionInterruptionTypeBegan:
+                    // At this point our audio session has been deactivated and
+                    // the
+                    // audio unit render callbacks no longer occur. Nothing to
+                    // do.
+                    break;
+                  case AVAudioSessionInterruptionTypeEnded: {
+                    NSError* error = nil;
+                    AVAudioSession* session = [AVAudioSession sharedInstance];
+                    [session setActive:YES error:&error];
+                    if (error != nil) {
+                      LOG_F(LS_ERROR) << "Failed to active audio session";
+                    }
+                    // Post interruption the audio unit render callbacks don't
+                    // automatically continue, so we restart the unit manually
+                    // here.
+                    AudioOutputUnitStop(_auVoiceProcessing);
+                    AudioOutputUnitStart(_auVoiceProcessing);
+                    break;
+                  }
+                }
+              }];
+  // Increment refcount on observer using ARC bridge. Instance variable is a
+  // void* instead of an id because header is included in other pure C++
+  // files.
+  _audioInterruptionObserver = (__bridge_retained void*)observer;
 
-int16_t AudioDeviceIOS::PlayoutDevices() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
+  // Deactivate the audio session.
+  ActivateAudioSession(session, false);
 
-    return (int16_t)1;
+  return 0;
 }
 
-int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetPlayoutDevice(index=%u)", index);
+int32_t AudioDeviceIOS::ShutdownPlayOrRecord() {
+  LOGI() << "ShutdownPlayOrRecord";
 
-    if (_playIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout already initialized");
-        return -1;
+  if (_audioInterruptionObserver != nullptr) {
+    NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
+    // Transfer ownership of observer back to ARC, which will dealloc the
+    // observer once it exits this scope.
+    id observer = (__bridge_transfer id)_audioInterruptionObserver;
+    [center removeObserver:observer];
+    _audioInterruptionObserver = nullptr;
+  }
+
+  // Close and delete AU.
+  OSStatus result = -1;
+  if (nullptr != _auVoiceProcessing) {
+    result = AudioOutputUnitStop(_auVoiceProcessing);
+    if (0 != result) {
+      LOG_F(LS_ERROR) << "AudioOutputUnitStop failed: " << result;
     }
-
-    if (index !=0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetPlayoutDevice invalid index");
-        return -1;
+    result = AudioComponentInstanceDispose(_auVoiceProcessing);
+    if (0 != result) {
+      LOG_F(LS_ERROR) << "AudioComponentInstanceDispose failed: " << result;
     }
-    _playoutDeviceIsSpecified = true;
+    _auVoiceProcessing = nullptr;
+  }
 
-    return 0;
-}
-
-int32_t
-    AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
+  return 0;
 }
 
-int32_t
-    AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
-                                         char name[kAdmMaxDeviceNameSize],
-                                         char guid[kAdmMaxGuidSize]) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::PlayoutDeviceName(index=%u)", index);
-
-    if (index != 0) {
-        return -1;
-    }
-    // return empty strings
-    memset(name, 0, kAdmMaxDeviceNameSize);
-    if (guid != NULL) {
-        memset(guid, 0, kAdmMaxGuidSize);
-    }
+// ============================================================================
+//                                  Thread Methods
+// ============================================================================
 
+OSStatus AudioDeviceIOS::RecordProcess(
+    void* inRefCon,
+    AudioUnitRenderActionFlags* ioActionFlags,
+    const AudioTimeStamp* inTimeStamp,
+    UInt32 inBusNumber,
+    UInt32 inNumberFrames,
+    AudioBufferList* ioData) {
+  AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
+  return ptrThis->RecordProcessImpl(ioActionFlags, inTimeStamp, inBusNumber,
+                                    inNumberFrames);
+}
+
+OSStatus AudioDeviceIOS::RecordProcessImpl(
+    AudioUnitRenderActionFlags* ioActionFlags,
+    const AudioTimeStamp* inTimeStamp,
+    uint32_t inBusNumber,
+    uint32_t inNumberFrames) {
+  // Setup some basic stuff
+  // Use temp buffer not to lock up recording buffer more than necessary
+  // todo: Make dataTmp a member variable with static size that holds
+  //       max possible frames?
+  int16_t* dataTmp = new int16_t[inNumberFrames];
+  memset(dataTmp, 0, 2 * inNumberFrames);
+
+  AudioBufferList abList;
+  abList.mNumberBuffers = 1;
+  abList.mBuffers[0].mData = dataTmp;
+  abList.mBuffers[0].mDataByteSize = 2 * inNumberFrames;  // 2 bytes/sample
+  abList.mBuffers[0].mNumberChannels = 1;
+
+  // Get data from mic
+  OSStatus res = AudioUnitRender(_auVoiceProcessing, ioActionFlags, inTimeStamp,
+                                 inBusNumber, inNumberFrames, &abList);
+  if (res != 0) {
+    // TODO(henrika): improve error handling.
+    delete[] dataTmp;
     return 0;
-}
-
-int32_t
-    AudioDeviceIOS::RecordingDeviceName(uint16_t index,
-                                           char name[kAdmMaxDeviceNameSize],
-                                           char guid[kAdmMaxGuidSize]) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::RecordingDeviceName(index=%u)", index);
+  }
+
+  if (_recording) {
+    // Insert all data in temp buffer into recording buffers
+    // There is zero or one buffer partially full at any given time,
+    // all others are full or empty
+    // Full means filled with noSamp10ms samples.
+
+    const unsigned int noSamp10ms = _adbSampFreq / 100;
+    unsigned int dataPos = 0;
+    uint16_t bufPos = 0;
+    int16_t insertPos = -1;
+    unsigned int nCopy = 0;  // Number of samples to copy
+
+    while (dataPos < inNumberFrames) {
+      // Loop over all recording buffers or
+      // until we find the partially full buffer
+      // First choice is to insert into partially full buffer,
+      // second choice is to insert into empty buffer
+      bufPos = 0;
+      insertPos = -1;
+      nCopy = 0;
+      while (bufPos < N_REC_BUFFERS) {
+        if ((_recordingLength[bufPos] > 0) &&
+            (_recordingLength[bufPos] < noSamp10ms)) {
+          // Found the partially full buffer
+          insertPos = static_cast<int16_t>(bufPos);
+          // Don't need to search more, quit loop
+          bufPos = N_REC_BUFFERS;
+        } else if ((-1 == insertPos) && (0 == _recordingLength[bufPos])) {
+          // Found an empty buffer
+          insertPos = static_cast<int16_t>(bufPos);
+        }
+        ++bufPos;
+      }
 
-    if (index != 0) {
-        return -1;
-    }
-    // return empty strings
-    memset(name, 0, kAdmMaxDeviceNameSize);
-    if (guid != NULL) {
-        memset(guid, 0, kAdmMaxGuidSize);
+      // Insert data into buffer
+      if (insertPos > -1) {
+        // We found a non-full buffer, copy data to it
+        unsigned int dataToCopy = inNumberFrames - dataPos;
+        unsigned int currentRecLen = _recordingLength[insertPos];
+        unsigned int roomInBuffer = noSamp10ms - currentRecLen;
+        nCopy = (dataToCopy < roomInBuffer ? dataToCopy : roomInBuffer);
+
+        memcpy(&_recordingBuffer[insertPos][currentRecLen], &dataTmp[dataPos],
+               nCopy * sizeof(int16_t));
+        if (0 == currentRecLen) {
+          _recordingSeqNumber[insertPos] = _recordingCurrentSeq;
+          ++_recordingCurrentSeq;
+        }
+        _recordingBufferTotalSize += nCopy;
+        // Has to be done last to avoid interrupt problems between threads.
+        _recordingLength[insertPos] += nCopy;
+        dataPos += nCopy;
+      } else {
+        // Didn't find a non-full buffer
+        // TODO(henrika): improve error handling
+        dataPos = inNumberFrames;  // Don't try to insert more
+      }
     }
+  }
+  delete[] dataTmp;
+  return 0;
+}
+
+OSStatus AudioDeviceIOS::PlayoutProcess(
+    void* inRefCon,
+    AudioUnitRenderActionFlags* ioActionFlags,
+    const AudioTimeStamp* inTimeStamp,
+    UInt32 inBusNumber,
+    UInt32 inNumberFrames,
+    AudioBufferList* ioData) {
+  AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
+  return ptrThis->PlayoutProcessImpl(inNumberFrames, ioData);
+}
+
+OSStatus AudioDeviceIOS::PlayoutProcessImpl(uint32_t inNumberFrames,
+                                            AudioBufferList* ioData) {
+  int16_t* data = static_cast<int16_t*>(ioData->mBuffers[0].mData);
+  unsigned int dataSizeBytes = ioData->mBuffers[0].mDataByteSize;
+  unsigned int dataSize = dataSizeBytes / 2;  // Number of samples
+  CHECK_EQ(dataSize, inNumberFrames);
+  memset(data, 0, dataSizeBytes);  // Start with empty buffer
+
+  // Get playout data from Audio Device Buffer
+
+  if (_playing) {
+    unsigned int noSamp10ms = _adbSampFreq / 100;
+    // todo: Member variable and allocate when samp freq is determined
+    int16_t* dataTmp = new int16_t[noSamp10ms];
+    memset(dataTmp, 0, 2 * noSamp10ms);
+    unsigned int dataPos = 0;
+    int noSamplesOut = 0;
+    unsigned int nCopy = 0;
+
+    // First insert data from playout buffer if any
+    if (_playoutBufferUsed > 0) {
+      nCopy = (dataSize < _playoutBufferUsed) ? dataSize : _playoutBufferUsed;
+      DCHECK_EQ(nCopy, _playoutBufferUsed);
+      memcpy(data, _playoutBuffer, 2 * nCopy);
+      dataPos = nCopy;
+      memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
+      _playoutBufferUsed = 0;
+    }
+
+    // Now get the rest from Audio Device Buffer.
+    while (dataPos < dataSize) {
+      // Update playout delay
+      UpdatePlayoutDelay();
+
+      // Ask for new PCM data to be played out using the AudioDeviceBuffer
+      noSamplesOut = audio_device_buffer_->RequestPlayoutData(noSamp10ms);
+
+      // Get data from Audio Device Buffer
+      noSamplesOut = audio_device_buffer_->GetPlayoutData(
+          reinterpret_cast<int8_t*>(dataTmp));
+      CHECK_EQ(noSamp10ms, (unsigned int)noSamplesOut);
+
+      // Insert as much as fits in data buffer
+      nCopy =
+          (dataSize - dataPos) > noSamp10ms ? noSamp10ms : (dataSize - dataPos);
+      memcpy(&data[dataPos], dataTmp, 2 * nCopy);
+
+      // Save rest in playout buffer if any
+      if (nCopy < noSamp10ms) {
+        memcpy(_playoutBuffer, &dataTmp[nCopy], 2 * (noSamp10ms - nCopy));
+        _playoutBufferUsed = noSamp10ms - nCopy;
+      }
 
-    return 0;
+      // Update loop/index counter, if we copied less than noSamp10ms
+      // samples we shall quit loop anyway
+      dataPos += noSamp10ms;
+    }
+    delete[] dataTmp;
+  }
+  return 0;
 }
 
-int16_t AudioDeviceIOS::RecordingDevices() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+// TODO(henrika): can either be removed or simplified.
+void AudioDeviceIOS::UpdatePlayoutDelay() {
+  ++_playoutDelayMeasurementCounter;
 
-    return (int16_t)1;
-}
+  if (_playoutDelayMeasurementCounter >= 100) {
+    // Update HW and OS delay every second, unlikely to change
 
-int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetRecordingDevice(index=%u)", index);
+    // Since this is eventually rounded to integral ms, add 0.5ms
+    // here to get round-to-nearest-int behavior instead of
+    // truncation.
+    double totalDelaySeconds = 0.0005;
 
-    if (_recIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording already initialized");
-        return -1;
+    // HW output latency
+    AVAudioSession* session = [AVAudioSession sharedInstance];
+    double latency = session.outputLatency;
+    assert(latency >= 0);
+    totalDelaySeconds += latency;
+
+    // HW buffer duration
+    double ioBufferDuration = session.IOBufferDuration;
+    assert(ioBufferDuration >= 0);
+    totalDelaySeconds += ioBufferDuration;
+
+    // AU latency
+    Float64 f64(0);
+    UInt32 size = sizeof(f64);
+    OSStatus result =
+        AudioUnitGetProperty(_auVoiceProcessing, kAudioUnitProperty_Latency,
+                             kAudioUnitScope_Global, 0, &f64, &size);
+    if (0 != result) {
+      LOG_F(LS_ERROR) << "AU latency error: " << result;
     }
+    assert(f64 >= 0);
+    totalDelaySeconds += f64;
 
-    if (index !=0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  SetRecordingDevice invalid index");
-        return -1;
-    }
+    // To ms
+    _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000);
 
-    _recordingDeviceIsSpecified = true;
+    // Reset counter
+    _playoutDelayMeasurementCounter = 0;
+  }
 
-    return 0;
+  // todo: Add playout buffer?
 }
 
-int32_t
-    AudioDeviceIOS::SetRecordingDevice(
-                                        AudioDeviceModule::WindowsDeviceType) {
-    WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                 "WindowsDeviceType not supported");
-    return -1;
-}
+void AudioDeviceIOS::UpdateRecordingDelay() {
+  ++_recordingDelayMeasurementCounter;
 
-// ----------------------------------------------------------------------------
-//  SetLoudspeakerStatus
-//
-//  Change the default receiver playout route to speaker.
-//
-// ----------------------------------------------------------------------------
+  if (_recordingDelayMeasurementCounter >= 100) {
+    // Update HW and OS delay every second, unlikely to change
 
-int32_t AudioDeviceIOS::SetLoudspeakerStatus(bool enable) {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetLoudspeakerStatus(enable=%d)", enable);
+    // Since this is eventually rounded to integral ms, add 0.5ms
+    // here to get round-to-nearest-int behavior instead of
+    // truncation.
+    double totalDelaySeconds = 0.0005;
 
+    // HW input latency
     AVAudioSession* session = [AVAudioSession sharedInstance];
-    NSString* category = session.category;
-    AVAudioSessionCategoryOptions options = session.categoryOptions;
-    // Respect old category options if category is
-    // AVAudioSessionCategoryPlayAndRecord. Otherwise reset it since old options
-    // might not be valid for this category.
-    if ([category isEqualToString:AVAudioSessionCategoryPlayAndRecord]) {
-      if (enable) {
-        options |= AVAudioSessionCategoryOptionDefaultToSpeaker;
-      } else {
-        options &= ~AVAudioSessionCategoryOptionDefaultToSpeaker;
-      }
-    } else {
-      options = AVAudioSessionCategoryOptionDefaultToSpeaker;
-    }
-
-    NSError* error = nil;
-    [session setCategory:AVAudioSessionCategoryPlayAndRecord
-             withOptions:options
-                   error:&error];
-    if (error != nil) {
-      WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                   "Error changing default output route ");
-      return -1;
+    double latency = session.inputLatency;
+    assert(latency >= 0);
+    totalDelaySeconds += latency;
+
+    // HW buffer duration
+    double ioBufferDuration = session.IOBufferDuration;
+    assert(ioBufferDuration >= 0);
+    totalDelaySeconds += ioBufferDuration;
+
+    // AU latency
+    Float64 f64(0);
+    UInt32 size = sizeof(f64);
+    OSStatus result =
+        AudioUnitGetProperty(_auVoiceProcessing, kAudioUnitProperty_Latency,
+                             kAudioUnitScope_Global, 0, &f64, &size);
+    if (0 != result) {
+      LOG_F(LS_ERROR) << "AU latency error: " << result;
     }
+    assert(f64 >= 0);
+    totalDelaySeconds += f64;
 
-    return 0;
-}
+    // To ms
+    _recordingDelayHWAndOS = static_cast<uint32_t>(totalDelaySeconds / 1000);
 
-int32_t AudioDeviceIOS::GetLoudspeakerStatus(bool &enabled) const {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetLoudspeakerStatus(enabled=?)");
+    // Reset counter
+    _recordingDelayMeasurementCounter = 0;
+  }
 
-    AVAudioSession* session = [AVAudioSession sharedInstance];
-    AVAudioSessionCategoryOptions options = session.categoryOptions;
-    enabled = options & AVAudioSessionCategoryOptionDefaultToSpeaker;
+  _recordingDelay = _recordingDelayHWAndOS;
 
-    return 0;
+  // ADB recording buffer size, update every time
+  // Don't count the one next 10 ms to be sent, then convert samples => ms
+  const uint32_t noSamp10ms = _adbSampFreq / 100;
+  if (_recordingBufferTotalSize > noSamp10ms) {
+    _recordingDelay +=
+        (_recordingBufferTotalSize - noSamp10ms) / (_adbSampFreq / 1000);
+  }
 }
 
-int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    available = false;
-
-    // Try to initialize the playout side
-    int32_t res = InitPlayout();
-
-    // Cancel effect of initialization
-    StopPlayout();
-
-    if (res != -1) {
-        available = true;
-    }
-
-    return 0;
+bool AudioDeviceIOS::RunCapture(void* ptrThis) {
+  return static_cast<AudioDeviceIOS*>(ptrThis)->CaptureWorkerThread();
 }
 
-int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    available = false;
-
-    // Try to initialize the recording side
-    int32_t res = InitRecording();
+bool AudioDeviceIOS::CaptureWorkerThread() {
+  if (_recording) {
+    int bufPos = 0;
+    unsigned int lowestSeq = 0;
+    int lowestSeqBufPos = 0;
+    bool foundBuf = true;
+    const unsigned int noSamp10ms = _adbSampFreq / 100;
+
+    while (foundBuf) {
+      // Check if we have any buffer with data to insert
+      // into the Audio Device Buffer,
+      // and find the one with the lowest seq number
+      foundBuf = false;
+      for (bufPos = 0; bufPos < N_REC_BUFFERS; ++bufPos) {
+        if (noSamp10ms == _recordingLength[bufPos]) {
+          if (!foundBuf) {
+            lowestSeq = _recordingSeqNumber[bufPos];
+            lowestSeqBufPos = bufPos;
+            foundBuf = true;
+          } else if (_recordingSeqNumber[bufPos] < lowestSeq) {
+            lowestSeq = _recordingSeqNumber[bufPos];
+            lowestSeqBufPos = bufPos;
+          }
+        }
+      }
 
-    // Cancel effect of initialization
-    StopRecording();
+      // Insert data into the Audio Device Buffer if found any
+      if (foundBuf) {
+        // Update recording delay
+        UpdateRecordingDelay();
 
-    if (res != -1) {
-        available = true;
-    }
+        // Set the recorded buffer
+        audio_device_buffer_->SetRecordedBuffer(
+            reinterpret_cast<int8_t*>(_recordingBuffer[lowestSeqBufPos]),
+            _recordingLength[lowestSeqBufPos]);
 
-    return 0;
-}
+        // Don't need to set the current mic level in ADB since we only
+        // support digital AGC,
+        // and besides we cannot get or set the IOS mic level anyway.
 
-int32_t AudioDeviceIOS::InitPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
+        // Set VQE info, use clockdrift == 0
+        audio_device_buffer_->SetVQEData(_playoutDelay, _recordingDelay, 0);
 
-    CriticalSectionScoped lock(&_critSect);
+        // Deliver recorded samples at specified sample rate, mic level
+        // etc. to the observer using callback
+        audio_device_buffer_->DeliverRecordedData();
 
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "  Not initialized");
-        return -1;
-    }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Playout already started");
-        return -1;
-    }
-
-    if (_playIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout already initialized");
-        return 0;
-    }
-
-    if (!_playoutDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout device is not specified");
-        return -1;
-    }
-
-    // Initialize the speaker
-    if (InitSpeaker() == -1) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitSpeaker() failed");
-    }
-
-    _playIsInitialized = true;
-
-    if (!_recIsInitialized) {
-        // Audio init
-        if (InitPlayOrRecord() == -1) {
-            // todo: Handle error
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  InitPlayOrRecord() failed");
-        }
-    } else {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-        "  Recording already initialized - InitPlayOrRecord() not called");
-    }
-
-    return 0;
-}
-
-bool AudioDeviceIOS::PlayoutIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_playIsInitialized);
-}
-
-int32_t AudioDeviceIOS::InitRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_initialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Not initialized");
-        return -1;
-    }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Recording already started");
-        return -1;
-    }
-
-    if (_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording already initialized");
-        return 0;
-    }
-
-    if (!_recordingDeviceIsSpecified) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording device is not specified");
-        return -1;
-    }
-
-    // Initialize the microphone
-    if (InitMicrophone() == -1) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  InitMicrophone() failed");
-    }
-
-    _recIsInitialized = true;
-
-    if (!_playIsInitialized) {
-        // Audio init
-        if (InitPlayOrRecord() == -1) {
-            // todo: Handle error
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  InitPlayOrRecord() failed");
-        }
-    } else {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout already initialized - InitPlayOrRecord() " \
-                     "not called");
-    }
-
-    return 0;
-}
-
-bool AudioDeviceIOS::RecordingIsInitialized() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_recIsInitialized);
-}
-
-int32_t AudioDeviceIOS::StartRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Recording not initialized");
-        return -1;
-    }
-
-    if (_recording) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording already started");
-        return 0;
-    }
-
-    // Reset recording buffer
-    memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
-    memset(_recordingLength, 0, sizeof(_recordingLength));
-    memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
-    _recordingCurrentSeq = 0;
-    _recordingBufferTotalSize = 0;
-    _recordingDelay = 0;
-    _recordingDelayHWAndOS = 0;
-    // Make sure first call to update delay function will update delay
-    _recordingDelayMeasurementCounter = 9999;
-    _recWarning = 0;
-    _recError = 0;
-
-    if (!_playing) {
-        // Start Audio Unit
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "  Starting Audio Unit");
-        OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                         "  Error starting Audio Unit (result=%d)", result);
-            return -1;
-        }
-    }
-
-    _recording = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StopRecording() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Recording is not initialized");
-        return 0;
-    }
-
-    _recording = false;
-
-    if (!_playing) {
-        // Both playout and recording has stopped, shutdown the device
-        ShutdownPlayOrRecord();
-    }
-
-    _recIsInitialized = false;
-    _micIsInitialized = false;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::Recording() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-    return (_recording);
-}
-
-int32_t AudioDeviceIOS::StartPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    // This lock is (among other things) needed to avoid concurrency issues
-    // with capture thread
-    // shutting down Audio Unit
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Playout not initialized");
-        return -1;
-    }
-
-    if (_playing) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playing already started");
-        return 0;
-    }
-
-    // Reset playout buffer
-    memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-    _playoutBufferUsed = 0;
-    _playoutDelay = 0;
-    // Make sure first call to update delay function will update delay
-    _playoutDelayMeasurementCounter = 9999;
-    _playWarning = 0;
-    _playError = 0;
-
-    if (!_recording) {
-        // Start Audio Unit
-        WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                     "  Starting Audio Unit");
-        OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
-                         "  Error starting Audio Unit (result=%d)", result);
-            return -1;
-        }
-    }
-
-    _playing = true;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::StopPlayout() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout is not initialized");
-        return 0;
-    }
-
-    _playing = false;
-
-    if (!_recording) {
-        // Both playout and recording has stopped, signal shutdown the device
-        ShutdownPlayOrRecord();
-    }
-
-    _playIsInitialized = false;
-    _speakerIsInitialized = false;
-
-    return 0;
-}
-
-bool AudioDeviceIOS::Playing() const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "%s", __FUNCTION__);
-    return (_playing);
-}
-
-// ----------------------------------------------------------------------------
-//  ResetAudioDevice
-//
-//  Disable playout and recording, signal to capture thread to shutdown,
-//  and set enable states after shutdown to same as current.
-//  In capture thread audio device will be shutdown, then started again.
-// ----------------------------------------------------------------------------
-int32_t AudioDeviceIOS::ResetAudioDevice() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    CriticalSectionScoped lock(&_critSect);
-
-    if (!_playIsInitialized && !_recIsInitialized) {
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Playout or recording not initialized, doing nothing");
-        return 0;  // Nothing to reset
-    }
-
-    // Store the states we have before stopping to restart below
-    bool initPlay = _playIsInitialized;
-    bool play = _playing;
-    bool initRec = _recIsInitialized;
-    bool rec = _recording;
-
-    int res(0);
-
-    // Stop playout and recording
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Stopping playout and recording");
-    res += StopPlayout();
-    res += StopRecording();
-
-    // Restart
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Restarting playout and recording (%d, %d, %d, %d)",
-                 initPlay, play, initRec, rec);
-    if (initPlay) res += InitPlayout();
-    if (initRec)  res += InitRecording();
-    if (play)     res += StartPlayout();
-    if (rec)      res += StartRecording();
-
-    if (0 != res) {
-        // Logging is done in init/start/stop calls above
-        return -1;
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
-    delayMS = _playoutDelay;
-    return 0;
-}
-
-int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
-    delayMS = _recordingDelay;
-    return 0;
-}
-
-int32_t
-    AudioDeviceIOS::SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
-                                     uint16_t sizeMS) {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
-                 "AudioDeviceIOS::SetPlayoutBuffer(type=%u, sizeMS=%u)",
-                 type, sizeMS);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-int32_t
-    AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                     uint16_t& sizeMS) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    type = AudioDeviceModule::kAdaptiveBufferSize;
-
-    sizeMS = _playoutDelay;
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::CPULoad(uint16_t& /*load*/) const {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                 "  API call not supported on this platform");
-    return -1;
-}
-
-bool AudioDeviceIOS::PlayoutWarning() const {
-    return (_playWarning > 0);
-}
-
-bool AudioDeviceIOS::PlayoutError() const {
-    return (_playError > 0);
-}
-
-bool AudioDeviceIOS::RecordingWarning() const {
-    return (_recWarning > 0);
-}
-
-bool AudioDeviceIOS::RecordingError() const {
-    return (_recError > 0);
-}
-
-void AudioDeviceIOS::ClearPlayoutWarning() {
-    _playWarning = 0;
-}
-
-void AudioDeviceIOS::ClearPlayoutError() {
-    _playError = 0;
-}
-
-void AudioDeviceIOS::ClearRecordingWarning() {
-    _recWarning = 0;
-}
-
-void AudioDeviceIOS::ClearRecordingError() {
-    _recError = 0;
-}
-
-// ============================================================================
-//                                 Private Methods
-// ============================================================================
-
-int32_t AudioDeviceIOS::InitPlayOrRecord() {
-    WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    OSStatus result = -1;
-
-    // Check if already initialized
-    if (NULL != _auVoiceProcessing) {
-        // We already have initialized before and created any of the audio unit,
-        // check that all exist
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "  Already initialized");
-        // todo: Call AudioUnitReset() here and empty all buffers?
-        return 0;
-    }
-
-    // Create Voice Processing Audio Unit
-    AudioComponentDescription desc;
-    AudioComponent comp;
-
-    desc.componentType = kAudioUnitType_Output;
-    desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
-    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-    desc.componentFlags = 0;
-    desc.componentFlagsMask = 0;
-
-    comp = AudioComponentFindNext(NULL, &desc);
-    if (NULL == comp) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not find audio component for Audio Unit");
-        return -1;
-    }
-
-    result = AudioComponentInstanceNew(comp, &_auVoiceProcessing);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not create Audio Unit instance (result=%d)",
-                     result);
-        return -1;
-    }
-
-    // Set preferred hardware sample rate to 16 kHz
-    NSError* error = nil;
-    AVAudioSession* session = [AVAudioSession sharedInstance];
-    Float64 preferredSampleRate(16000.0);
-    [session setPreferredSampleRate:preferredSampleRate
-                              error:&error];
-    if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set preferred sample rate: %s", errorString);
-    }
-    error = nil;
-    // Make the setMode:error: and setCategory:error: calls only if necessary.
-    // Non-obviously, setting them to the value they already have will clear
-    // transient properties (such as PortOverride) that some other component may
-    // have set up.
-    if (session.mode != AVAudioSessionModeVoiceChat) {
-      [session setMode:AVAudioSessionModeVoiceChat error:&error];
-      if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set mode: %s", errorString);
+        // Make buffer available
+        _recordingSeqNumber[lowestSeqBufPos] = 0;
+        _recordingBufferTotalSize -= _recordingLength[lowestSeqBufPos];
+        // Must be done last to avoid interrupt problems between threads
+        _recordingLength[lowestSeqBufPos] = 0;
       }
     }
-    error = nil;
-    if (session.category != AVAudioSessionCategoryPlayAndRecord) {
-      [session setCategory:AVAudioSessionCategoryPlayAndRecord error:&error];
-      if (error != nil) {
-        const char* errorString = [[error localizedDescription] UTF8String];
-        WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                     "Could not set category: %s", errorString);
-      }
-    }
-
-    //////////////////////
-    // Setup Voice Processing Audio Unit
-
-    // Note: For Signal Processing AU element 0 is output bus, element 1 is
-    //       input bus for global scope element is irrelevant (always use
-    //       element 0)
-
-    // Enable IO on both elements
-
-    // todo: Below we just log and continue upon error. We might want
-    //       to close AU and return error for some cases.
-    // todo: Log info about setup.
-
-    UInt32 enableIO = 1;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_EnableIO,
-                                  kAudioUnitScope_Input,
-                                  1,  // input bus
-                                  &enableIO,
-                                  sizeof(enableIO));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not enable IO on input (result=%d)", result);
-    }
-
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_EnableIO,
-                                  kAudioUnitScope_Output,
-                                  0,   // output bus
-                                  &enableIO,
-                                  sizeof(enableIO));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not enable IO on output (result=%d)", result);
-    }
-
-    // Disable AU buffer allocation for the recorder, we allocate our own
-    UInt32 flag = 0;
-    result = AudioUnitSetProperty(
-        _auVoiceProcessing, kAudioUnitProperty_ShouldAllocateBuffer,
-        kAudioUnitScope_Output,  1, &flag, sizeof(flag));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Could not disable AU buffer allocation (result=%d)",
-                     result);
-        // Should work anyway
-    }
-
-    // Set recording callback
-    AURenderCallbackStruct auCbS;
-    memset(&auCbS, 0, sizeof(auCbS));
-    auCbS.inputProc = RecordProcess;
-    auCbS.inputProcRefCon = this;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioOutputUnitProperty_SetInputCallback,
-                                  kAudioUnitScope_Global, 1,
-                                  &auCbS, sizeof(auCbS));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set record callback for Audio Unit (result=%d)",
-            result);
-    }
-
-    // Set playout callback
-    memset(&auCbS, 0, sizeof(auCbS));
-    auCbS.inputProc = PlayoutProcess;
-    auCbS.inputProcRefCon = this;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_SetRenderCallback,
-                                  kAudioUnitScope_Global, 0,
-                                  &auCbS, sizeof(auCbS));
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set play callback for Audio Unit (result=%d)",
-            result);
-    }
-
-    // Get stream format for out/0
-    AudioStreamBasicDescription playoutDesc;
-    UInt32 size = sizeof(playoutDesc);
-    result = AudioUnitGetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Output, 0, &playoutDesc,
-                                  &size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not get stream format Audio Unit out/0 (result=%d)",
-            result);
-    }
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  Audio Unit playout opened in sampling rate %f",
-                 playoutDesc.mSampleRate);
-
-    playoutDesc.mSampleRate = preferredSampleRate;
-
-    // Store the sampling frequency to use towards the Audio Device Buffer
-    // todo: Add 48 kHz (increase buffer sizes). Other fs?
-    if ((playoutDesc.mSampleRate > 44090.0)
-        && (playoutDesc.mSampleRate < 44110.0)) {
-        _adbSampFreq = 44100;
-    } else if ((playoutDesc.mSampleRate > 15990.0)
-               && (playoutDesc.mSampleRate < 16010.0)) {
-        _adbSampFreq = 16000;
-    } else if ((playoutDesc.mSampleRate > 7990.0)
-               && (playoutDesc.mSampleRate < 8010.0)) {
-        _adbSampFreq = 8000;
-    } else {
-        _adbSampFreq = 0;
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Audio Unit out/0 opened in unknown sampling rate (%f)",
-            playoutDesc.mSampleRate);
-        // todo: We should bail out here.
-    }
-
-    // Set the audio device buffer sampling rate,
-    // we assume we get the same for play and record
-    if (_ptrAudioBuffer->SetRecordingSampleRate(_adbSampFreq) < 0) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set audio device buffer recording sampling rate (%d)",
-            _adbSampFreq);
-    }
-
-    if (_ptrAudioBuffer->SetPlayoutSampleRate(_adbSampFreq) < 0) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set audio device buffer playout sampling rate (%d)",
-            _adbSampFreq);
-    }
-
-    // Set stream format for in/0  (use same sampling frequency as for out/0)
-    playoutDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-                               | kLinearPCMFormatFlagIsPacked
-                               | kLinearPCMFormatFlagIsNonInterleaved;
-    playoutDesc.mBytesPerPacket = 2;
-    playoutDesc.mFramesPerPacket = 1;
-    playoutDesc.mBytesPerFrame = 2;
-    playoutDesc.mChannelsPerFrame = 1;
-    playoutDesc.mBitsPerChannel = 16;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Input, 0, &playoutDesc, size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set stream format Audio Unit in/0 (result=%d)",
-            result);
-    }
-
-    // Get stream format for in/1
-    AudioStreamBasicDescription recordingDesc;
-    size = sizeof(recordingDesc);
-    result = AudioUnitGetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Input, 1, &recordingDesc,
-                                  &size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not get stream format Audio Unit in/1 (result=%d)",
-            result);
-    }
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
-                 "  Audio Unit recording opened in sampling rate %f",
-                 recordingDesc.mSampleRate);
-
-    recordingDesc.mSampleRate = preferredSampleRate;
-
-    // Set stream format for out/1 (use same sampling frequency as for in/1)
-    recordingDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
-                                 | kLinearPCMFormatFlagIsPacked
-                                 | kLinearPCMFormatFlagIsNonInterleaved;
-
-    recordingDesc.mBytesPerPacket = 2;
-    recordingDesc.mFramesPerPacket = 1;
-    recordingDesc.mBytesPerFrame = 2;
-    recordingDesc.mChannelsPerFrame = 1;
-    recordingDesc.mBitsPerChannel = 16;
-    result = AudioUnitSetProperty(_auVoiceProcessing,
-                                  kAudioUnitProperty_StreamFormat,
-                                  kAudioUnitScope_Output, 1, &recordingDesc,
-                                  size);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-            "  Could not set stream format Audio Unit out/1 (result=%d)",
-            result);
-    }
-
-    // Initialize here already to be able to get/set stream properties.
-    result = AudioUnitInitialize(_auVoiceProcessing);
-    if (0 != result) {
-        WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                     "  Could not init Audio Unit (result=%d)", result);
-    }
-
-    // Get hardware sample rate for logging (see if we get what we asked for)
-    double sampleRate = session.sampleRate;
-    WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-                 "  Current HW sample rate is %f, ADB sample rate is %d",
-                 sampleRate, _adbSampFreq);
-
-    // Listen to audio interruptions.
-    NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
-    id observer =
-        [center addObserverForName:AVAudioSessionInterruptionNotification
-                            object:nil
-                             queue:[NSOperationQueue mainQueue]
-                        usingBlock:^(NSNotification* notification) {
-          NSNumber* typeNumber =
-              [notification userInfo][AVAudioSessionInterruptionTypeKey];
-          AVAudioSessionInterruptionType type =
-              (AVAudioSessionInterruptionType)[typeNumber unsignedIntegerValue];
-          switch (type) {
-            case AVAudioSessionInterruptionTypeBegan:
-              // At this point our audio session has been deactivated and the
-              // audio unit render callbacks no longer occur. Nothing to do.
-              break;
-            case AVAudioSessionInterruptionTypeEnded: {
-              NSError* error = nil;
-              AVAudioSession* session = [AVAudioSession sharedInstance];
-              [session setActive:YES
-                           error:&error];
-              if (error != nil) {
-                  WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                               "Error activating audio session");
-              }
-              // Post interruption the audio unit render callbacks don't
-              // automatically continue, so we restart the unit manually here.
-              AudioOutputUnitStop(_auVoiceProcessing);
-              AudioOutputUnitStart(_auVoiceProcessing);
-              break;
-            }
-          }
-        }];
-    // Increment refcount on observer using ARC bridge. Instance variable is a
-    // void* instead of an id because header is included in other pure C++
-    // files.
-    _audioInterruptionObserver = (__bridge_retained void*)observer;
-
-    // Activate audio session.
-    error = nil;
-    [session setActive:YES
-                 error:&error];
-    if (error != nil) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "Error activating audio session");
-    }
-
-    return 0;
-}
-
-int32_t AudioDeviceIOS::ShutdownPlayOrRecord() {
-    WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
-    if (_audioInterruptionObserver != NULL) {
-        NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
-        // Transfer ownership of observer back to ARC, which will dealloc the
-        // observer once it exits this scope.
-        id observer = (__bridge_transfer id)_audioInterruptionObserver;
-        [center removeObserver:observer];
-        _audioInterruptionObserver = NULL;
-    }
-
-    // Close and delete AU
-    OSStatus result = -1;
-    if (NULL != _auVoiceProcessing) {
-        result = AudioOutputUnitStop(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                "  Error stopping Audio Unit (result=%d)", result);
-        }
-        result = AudioComponentInstanceDispose(_auVoiceProcessing);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                "  Error disposing Audio Unit (result=%d)", result);
-        }
-        _auVoiceProcessing = NULL;
-    }
-
-    return 0;
-}
-
-// ============================================================================
-//                                  Thread Methods
-// ============================================================================
-
-OSStatus
-    AudioDeviceIOS::RecordProcess(void *inRefCon,
-                                  AudioUnitRenderActionFlags *ioActionFlags,
-                                  const AudioTimeStamp *inTimeStamp,
-                                  UInt32 inBusNumber,
-                                  UInt32 inNumberFrames,
-                                  AudioBufferList *ioData) {
-    AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
-
-    return ptrThis->RecordProcessImpl(ioActionFlags,
-                                      inTimeStamp,
-                                      inBusNumber,
-                                      inNumberFrames);
-}
-
-
-OSStatus
-    AudioDeviceIOS::RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags,
-                                      const AudioTimeStamp *inTimeStamp,
-                                      uint32_t inBusNumber,
-                                      uint32_t inNumberFrames) {
-    // Setup some basic stuff
-    // Use temp buffer not to lock up recording buffer more than necessary
-    // todo: Make dataTmp a member variable with static size that holds
-    //       max possible frames?
-    int16_t* dataTmp = new int16_t[inNumberFrames];
-    memset(dataTmp, 0, 2*inNumberFrames);
-
-    AudioBufferList abList;
-    abList.mNumberBuffers = 1;
-    abList.mBuffers[0].mData = dataTmp;
-    abList.mBuffers[0].mDataByteSize = 2*inNumberFrames;  // 2 bytes/sample
-    abList.mBuffers[0].mNumberChannels = 1;
-
-    // Get data from mic
-    OSStatus res = AudioUnitRender(_auVoiceProcessing,
-                                   ioActionFlags, inTimeStamp,
-                                   inBusNumber, inNumberFrames, &abList);
-    if (res != 0) {
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "  Error getting rec data, error = %d", res);
-
-        if (_recWarning > 0) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  Pending rec warning exists");
-        }
-        _recWarning = 1;
-
-        delete [] dataTmp;
-        return 0;
-    }
-
-    if (_recording) {
-        // Insert all data in temp buffer into recording buffers
-        // There is zero or one buffer partially full at any given time,
-        // all others are full or empty
-        // Full means filled with noSamp10ms samples.
-
-        const unsigned int noSamp10ms = _adbSampFreq / 100;
-        unsigned int dataPos = 0;
-        uint16_t bufPos = 0;
-        int16_t insertPos = -1;
-        unsigned int nCopy = 0;  // Number of samples to copy
-
-        while (dataPos < inNumberFrames) {
-            // Loop over all recording buffers or
-            // until we find the partially full buffer
-            // First choice is to insert into partially full buffer,
-            // second choice is to insert into empty buffer
-            bufPos = 0;
-            insertPos = -1;
-            nCopy = 0;
-            while (bufPos < N_REC_BUFFERS) {
-                if ((_recordingLength[bufPos] > 0)
-                    && (_recordingLength[bufPos] < noSamp10ms)) {
-                    // Found the partially full buffer
-                    insertPos = static_cast<int16_t>(bufPos);
-                    // Don't need to search more, quit loop
-                    bufPos = N_REC_BUFFERS;
-                } else if ((-1 == insertPos)
-                           && (0 == _recordingLength[bufPos])) {
-                    // Found an empty buffer
-                    insertPos = static_cast<int16_t>(bufPos);
-                }
-                ++bufPos;
-            }
-
-            // Insert data into buffer
-            if (insertPos > -1) {
-                // We found a non-full buffer, copy data to it
-                unsigned int dataToCopy = inNumberFrames - dataPos;
-                unsigned int currentRecLen = _recordingLength[insertPos];
-                unsigned int roomInBuffer = noSamp10ms - currentRecLen;
-                nCopy = (dataToCopy < roomInBuffer ? dataToCopy : roomInBuffer);
-
-                memcpy(&_recordingBuffer[insertPos][currentRecLen],
-                       &dataTmp[dataPos], nCopy*sizeof(int16_t));
-                if (0 == currentRecLen) {
-                    _recordingSeqNumber[insertPos] = _recordingCurrentSeq;
-                    ++_recordingCurrentSeq;
-                }
-                _recordingBufferTotalSize += nCopy;
-                // Has to be done last to avoid interrupt problems
-                // between threads
-                _recordingLength[insertPos] += nCopy;
-                dataPos += nCopy;
-            } else {
-                // Didn't find a non-full buffer
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "  Could not insert into recording buffer");
-                if (_recWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending rec warning exists");
-                }
-                _recWarning = 1;
-                dataPos = inNumberFrames;  // Don't try to insert more
-            }
-        }
-    }
-
-    delete [] dataTmp;
-
-    return 0;
-}
-
-OSStatus
-    AudioDeviceIOS::PlayoutProcess(void *inRefCon,
-                                   AudioUnitRenderActionFlags *ioActionFlags,
-                                   const AudioTimeStamp *inTimeStamp,
-                                   UInt32 inBusNumber,
-                                   UInt32 inNumberFrames,
-                                   AudioBufferList *ioData) {
-    AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
-
-    return ptrThis->PlayoutProcessImpl(inNumberFrames, ioData);
-}
-
-OSStatus
-    AudioDeviceIOS::PlayoutProcessImpl(uint32_t inNumberFrames,
-                                       AudioBufferList *ioData) {
-    // Setup some basic stuff
-//    assert(sizeof(short) == 2); // Assumption for implementation
-
-    int16_t* data =
-        static_cast<int16_t*>(ioData->mBuffers[0].mData);
-    unsigned int dataSizeBytes = ioData->mBuffers[0].mDataByteSize;
-    unsigned int dataSize = dataSizeBytes/2;  // Number of samples
-        if (dataSize != inNumberFrames) {  // Should always be the same
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                     "dataSize (%u) != inNumberFrames (%u)",
-                     dataSize, (unsigned int)inNumberFrames);
-        if (_playWarning > 0) {
-            WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                         "  Pending play warning exists");
-        }
-        _playWarning = 1;
-    }
-    memset(data, 0, dataSizeBytes);  // Start with empty buffer
-
-
-    // Get playout data from Audio Device Buffer
-
-    if (_playing) {
-        unsigned int noSamp10ms = _adbSampFreq / 100;
-        // todo: Member variable and allocate when samp freq is determined
-        int16_t* dataTmp = new int16_t[noSamp10ms];
-        memset(dataTmp, 0, 2*noSamp10ms);
-        unsigned int dataPos = 0;
-        int noSamplesOut = 0;
-        unsigned int nCopy = 0;
-
-        // First insert data from playout buffer if any
-        if (_playoutBufferUsed > 0) {
-            nCopy = (dataSize < _playoutBufferUsed) ?
-                    dataSize : _playoutBufferUsed;
-            if (nCopy != _playoutBufferUsed) {
-                // todo: If dataSize < _playoutBufferUsed
-                //       (should normally never be)
-                //       we must move the remaining data
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "nCopy (%u) != _playoutBufferUsed (%u)",
-                             nCopy, _playoutBufferUsed);
-                if (_playWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending play warning exists");
-                }
-                _playWarning = 1;
-            }
-            memcpy(data, _playoutBuffer, 2*nCopy);
-            dataPos = nCopy;
-            memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-            _playoutBufferUsed = 0;
-        }
-
-        // Now get the rest from Audio Device Buffer
-        while (dataPos < dataSize) {
-            // Update playout delay
-            UpdatePlayoutDelay();
-
-            // Ask for new PCM data to be played out using the AudioDeviceBuffer
-            noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(noSamp10ms);
-
-            // Get data from Audio Device Buffer
-            noSamplesOut =
-                _ptrAudioBuffer->GetPlayoutData(
-                    reinterpret_cast<int8_t*>(dataTmp));
-            // Cast OK since only equality comparison
-            if (noSamp10ms != (unsigned int)noSamplesOut) {
-                // Should never happen
-                WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                             "noSamp10ms (%u) != noSamplesOut (%d)",
-                             noSamp10ms, noSamplesOut);
-
-                if (_playWarning > 0) {
-                    WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
-                                 "  Pending play warning exists");
-                }
-                _playWarning = 1;
-            }
-
-            // Insert as much as fits in data buffer
-            nCopy = (dataSize-dataPos) > noSamp10ms ?
-                    noSamp10ms : (dataSize-dataPos);
-            memcpy(&data[dataPos], dataTmp, 2*nCopy);
-
-            // Save rest in playout buffer if any
-            if (nCopy < noSamp10ms) {
-                memcpy(_playoutBuffer, &dataTmp[nCopy], 2*(noSamp10ms-nCopy));
-                _playoutBufferUsed = noSamp10ms - nCopy;
-            }
-
-            // Update loop/index counter, if we copied less than noSamp10ms
-            // samples we shall quit loop anyway
-            dataPos += noSamp10ms;
-        }
-
-        delete [] dataTmp;
-    }
-
-    return 0;
-}
-
-void AudioDeviceIOS::UpdatePlayoutDelay() {
-    ++_playoutDelayMeasurementCounter;
-
-    if (_playoutDelayMeasurementCounter >= 100) {
-        // Update HW and OS delay every second, unlikely to change
-
-        // Since this is eventually rounded to integral ms, add 0.5ms
-        // here to get round-to-nearest-int behavior instead of
-        // truncation.
-        double totalDelaySeconds = 0.0005;
-
-        // HW output latency
-        AVAudioSession* session = [AVAudioSession sharedInstance];
-        double latency = session.outputLatency;
-        assert(latency >= 0);
-        totalDelaySeconds += latency;
-
-        // HW buffer duration
-        double ioBufferDuration = session.IOBufferDuration;
-        assert(ioBufferDuration >= 0);
-        totalDelaySeconds += ioBufferDuration;
-
-        // AU latency
-        Float64 f64(0);
-        UInt32 size = sizeof(f64);
-        OSStatus result = AudioUnitGetProperty(
-            _auVoiceProcessing, kAudioUnitProperty_Latency,
-            kAudioUnitScope_Global, 0, &f64, &size);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "error AU latency (result=%d)", result);
-        }
-        assert(f64 >= 0);
-        totalDelaySeconds += f64;
-
-        // To ms
-        _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000);
-
-        // Reset counter
-        _playoutDelayMeasurementCounter = 0;
-    }
-
-    // todo: Add playout buffer?
-}
-
-void AudioDeviceIOS::UpdateRecordingDelay() {
-    ++_recordingDelayMeasurementCounter;
-
-    if (_recordingDelayMeasurementCounter >= 100) {
-        // Update HW and OS delay every second, unlikely to change
-
-        // Since this is eventually rounded to integral ms, add 0.5ms
-        // here to get round-to-nearest-int behavior instead of
-        // truncation.
-        double totalDelaySeconds = 0.0005;
-
-        // HW input latency
-        AVAudioSession* session = [AVAudioSession sharedInstance];
-        double latency = session.inputLatency;
-        assert(latency >= 0);
-        totalDelaySeconds += latency;
-
-        // HW buffer duration
-        double ioBufferDuration = session.IOBufferDuration;
-        assert(ioBufferDuration >= 0);
-        totalDelaySeconds += ioBufferDuration;
-
-        // AU latency
-        Float64 f64(0);
-        UInt32 size = sizeof(f64);
-        OSStatus result = AudioUnitGetProperty(
-             _auVoiceProcessing, kAudioUnitProperty_Latency,
-             kAudioUnitScope_Global, 0, &f64, &size);
-        if (0 != result) {
-            WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
-                         "error AU latency (result=%d)", result);
-        }
-        assert(f64 >= 0);
-        totalDelaySeconds += f64;
-
-        // To ms
-        _recordingDelayHWAndOS =
-            static_cast<uint32_t>(totalDelaySeconds / 1000);
-
-        // Reset counter
-        _recordingDelayMeasurementCounter = 0;
-    }
-
-    _recordingDelay = _recordingDelayHWAndOS;
-
-    // ADB recording buffer size, update every time
-    // Don't count the one next 10 ms to be sent, then convert samples => ms
-    const uint32_t noSamp10ms = _adbSampFreq / 100;
-    if (_recordingBufferTotalSize > noSamp10ms) {
-        _recordingDelay +=
-            (_recordingBufferTotalSize - noSamp10ms) / (_adbSampFreq / 1000);
-    }
-}
-
-bool AudioDeviceIOS::RunCapture(void* ptrThis) {
-    return static_cast<AudioDeviceIOS*>(ptrThis)->CaptureWorkerThread();
-}
-
-bool AudioDeviceIOS::CaptureWorkerThread() {
-    if (_recording) {
-        int bufPos = 0;
-        unsigned int lowestSeq = 0;
-        int lowestSeqBufPos = 0;
-        bool foundBuf = true;
-        const unsigned int noSamp10ms = _adbSampFreq / 100;
-
-        while (foundBuf) {
-            // Check if we have any buffer with data to insert
-            // into the Audio Device Buffer,
-            // and find the one with the lowest seq number
-            foundBuf = false;
-            for (bufPos = 0; bufPos < N_REC_BUFFERS; ++bufPos) {
-                if (noSamp10ms == _recordingLength[bufPos]) {
-                    if (!foundBuf) {
-                        lowestSeq = _recordingSeqNumber[bufPos];
-                        lowestSeqBufPos = bufPos;
-                        foundBuf = true;
-                    } else if (_recordingSeqNumber[bufPos] < lowestSeq) {
-                        lowestSeq = _recordingSeqNumber[bufPos];
-                        lowestSeqBufPos = bufPos;
-                    }
-                }
-            }  // for
-
-            // Insert data into the Audio Device Buffer if found any
-            if (foundBuf) {
-                // Update recording delay
-                UpdateRecordingDelay();
-
-                // Set the recorded buffer
-                _ptrAudioBuffer->SetRecordedBuffer(
-                    reinterpret_cast<int8_t*>(
-                        _recordingBuffer[lowestSeqBufPos]),
-                        _recordingLength[lowestSeqBufPos]);
-
-                // Don't need to set the current mic level in ADB since we only
-                // support digital AGC,
-                // and besides we cannot get or set the IOS mic level anyway.
-
-                // Set VQE info, use clockdrift == 0
-                _ptrAudioBuffer->SetVQEData(_playoutDelay, _recordingDelay, 0);
-
-                // Deliver recorded samples at specified sample rate, mic level
-                // etc. to the observer using callback
-                _ptrAudioBuffer->DeliverRecordedData();
-
-                // Make buffer available
-                _recordingSeqNumber[lowestSeqBufPos] = 0;
-                _recordingBufferTotalSize -= _recordingLength[lowestSeqBufPos];
-                // Must be done last to avoid interrupt problems between threads
-                _recordingLength[lowestSeqBufPos] = 0;
-            }
-        }  // while (foundBuf)
-    }  // if (_recording)
-
-    {
-        // Normal case
-        // Sleep thread (5ms) to let other threads get to work
-        // todo: Is 5 ms optimal? Sleep shorter if inserted into the Audio
-        //       Device Buffer?
-        timespec t;
-        t.tv_sec = 0;
-        t.tv_nsec = 5*1000*1000;
-        nanosleep(&t, NULL);
-    }
-
-    return true;
+  }
+
+  {
+    // Normal case
+    // Sleep thread (5ms) to let other threads get to work
+    // todo: Is 5 ms optimal? Sleep shorter if inserted into the Audio
+    //       Device Buffer?
+    timespec t;
+    t.tv_sec = 0;
+    t.tv_nsec = 5 * 1000 * 1000;
+    nanosleep(&t, nullptr);
+  }
+  return true;
 }
 
 }  // namespace webrtc