Refactor the AudioDevice for iOS and improve the performance and stability

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #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/atomicops.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/system_wrappers/interface/trace.h"
+#include "webrtc/modules/audio_device/fine_audio_buffer.h"
+#include "webrtc/modules/utility/interface/helpers_ios.h"
 
 namespace webrtc {
 
 #define LOGI() LOG(LS_INFO) << "AudioDeviceIOS::"
 
+#define LOG_AND_RETURN_IF_ERROR(error, message) \
+  do {                                          \
+    OSStatus err = error;                       \
+    if (err) {                                  \
+      LOG(LS_ERROR) << message << ": " << err;  \
+      return false;                             \
+    }                                           \
+  } while (0)
+
+// Preferred hardware sample rate (unit is in Hertz). The client sample rate
+// will be set to this value as well to avoid resampling the the audio unit's
+// format converter. Note that, some devices, e.g. BT headsets, only supports
+// 8000Hz as native sample rate.
+const double kPreferredSampleRate = 48000.0;
+// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
+// size used by WebRTC. The exact actual size will differ between devices.
+// Example: using 48kHz on iPhone 6 results in a native buffer size of
+// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
+// take care of any buffering required to convert between native buffers and
+// buffers used by WebRTC. It is beneficial for the performance if the native
+// size is as close to 10ms as possible since it results in "clean" callback
+// sequence without bursts of callbacks back to back.
+const double kPreferredIOBufferDuration = 0.01;
+// Try to use mono to save resources. Also avoids channel format conversion
+// in the I/O audio unit. Initial tests have shown that it is possible to use
+// mono natively for built-in microphones and for BT headsets but not for
+// wired headsets. Wired headsets only support stereo as native channel format
+// but it is a low cost operation to do a format conversion to mono in the
+// audio unit. Hence, we will not hit a CHECK in
+// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
+// preferred number of channels and the actual number of channels.
+const int kPreferredNumberOfChannels = 1;
+// Number of bytes per audio sample for 16-bit signed integer representation.
+const UInt32 kBytesPerSample = 2;
+// Hardcoded delay estimates based on real measurements.
+// TODO(henrika): these value is not used in combination with built-in AEC.
+// Can most likely be removed.
+const UInt16 kFixedPlayoutDelayEstimate = 30;
+const UInt16 kFixedRecordDelayEstimate = 30;
+
 using ios::CheckAndLogError;
 
+// Activates an audio session suitable for full duplex VoIP sessions when
+// |activate| is true. Also sets the preferred sample rate and IO buffer
+// duration. Deactivates an active audio session if |activate| is set to false.
 static void ActivateAudioSession(AVAudioSession* session, bool activate) {
   LOG(LS_INFO) << "ActivateAudioSession(" << activate << ")";
   @autoreleasepool {
     NSError* error = nil;
     BOOL success = NO;
+    // Deactivate the audio session and return if |activate| is false.
     if (!activate) {
-      // Deactivate the audio session.
       success = [session setActive:NO error:&error];
       DCHECK(CheckAndLogError(success, error));
       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.
+    // Use a category which supports simultaneous recording and playback.
+    // By default, using this category implies that our app’s audio is
+    // nonmixable, hence activating the session will interrupt any other
+    // audio sessions which are also nonmixable.
     if (session.category != AVAudioSessionCategoryPlayAndRecord) {
       error = nil;
       success = [session setCategory:AVAudioSessionCategoryPlayAndRecord
                                error:&error];
       DCHECK(CheckAndLogError(success, error));
     }
+    // Specify mode for two-way voice communication (e.g. VoIP).
     if (session.mode != AVAudioSessionModeVoiceChat) {
       error = nil;
       success = [session setMode:AVAudioSessionModeVoiceChat error:&error];
       DCHECK(CheckAndLogError(success, error));
     }
+    // Set the session's sample rate or the hardware sample rate.
+    // It is essential that we use the same sample rate as stream format
+    // to ensure that the I/O unit does not have to do sample rate conversion.
+    error = nil;
+    success =
+        [session setPreferredSampleRate:kPreferredSampleRate error:&error];
+    DCHECK(CheckAndLogError(success, error));
+    // Set the preferred audio I/O buffer duration, in seconds.
+    // TODO(henrika): add more comments here.
+    error = nil;
+    success = [session setPreferredIOBufferDuration:kPreferredIOBufferDuration
+                                              error:&error];
+    DCHECK(CheckAndLogError(success, error));
+
+    // TODO(henrika): add observers here...
+
+    // Activate the audio session. Activation can fail if another active audio
+    // session (e.g. phone call) has higher priority than ours.
     error = nil;
     success = [session setActive:YES error:&error];
     DCHECK(CheckAndLogError(success, error));
+    CHECK(session.isInputAvailable) << "No input path is available!";
     // 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;
-    size_t frames_per_buffer =
-        static_cast<size_t>(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.
+    // Try to set the preferred number of hardware audio channels. These calls
+    // must be done after setting the audio session’s category and mode and
+    // activating the session.
+    // We try to use mono in both directions to save resources and format
+    // conversions in the audio unit. Some devices does only support stereo;
+    // e.g. wired headset on iPhone 6.
+    // TODO(henrika): add support for stereo if needed.
+    error = nil;
+    success =
+        [session setPreferredInputNumberOfChannels:kPreferredNumberOfChannels
+                                             error:&error];
+    DCHECK(CheckAndLogError(success, error));
+    error = nil;
+    success =
+        [session setPreferredOutputNumberOfChannels:kPreferredNumberOfChannels
+                                              error:&error];
+    DCHECK(CheckAndLogError(success, error));
   }
 }
 
 #if !defined(NDEBUG)
+// Helper method for printing out an AudioStreamBasicDescription structure.
+static void LogABSD(AudioStreamBasicDescription absd) {
+  char formatIDString[5];
+  UInt32 formatID = CFSwapInt32HostToBig(absd.mFormatID);
+  bcopy(&formatID, formatIDString, 4);
+  formatIDString[4] = '\0';
+  LOG(LS_INFO) << "LogABSD";
+  LOG(LS_INFO) << " sample rate: " << absd.mSampleRate;
+  LOG(LS_INFO) << " format ID: " << formatIDString;
+  LOG(LS_INFO) << " format flags: " << std::hex << absd.mFormatFlags;
+  LOG(LS_INFO) << " bytes per packet: " << absd.mBytesPerPacket;
+  LOG(LS_INFO) << " frames per packet: " << absd.mFramesPerPacket;
+  LOG(LS_INFO) << " bytes per frame: " << absd.mBytesPerFrame;
+  LOG(LS_INFO) << " channels per packet: " << absd.mChannelsPerFrame;
+  LOG(LS_INFO) << " bits per channel: " << absd.mBitsPerChannel;
+  LOG(LS_INFO) << " reserved: " << absd.mReserved;
+}
+
+// Helper method that logs essential device information strings.
 static void LogDeviceInfo() {
   LOG(LS_INFO) << "LogDeviceInfo";
   @autoreleasepool {
     LOG(LS_INFO) << " system name: " << ios::GetSystemName();
     LOG(LS_INFO) << " system version: " << ios::GetSystemVersion();
     LOG(LS_INFO) << " device type: " << ios::GetDeviceType();
     LOG(LS_INFO) << " device name: " << ios::GetDeviceName();
   }
 }
-#endif
+#endif  // !defined(NDEBUG)
 
 AudioDeviceIOS::AudioDeviceIOS()
-    : audio_device_buffer_(nullptr),
-      _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-      _auVoiceProcessing(nullptr),
-      _audioInterruptionObserver(nullptr),
+    : _audioDeviceBuffer(nullptr),
+      _vpioUnit(nullptr),
+      _recording(0),
+      _playing(0),
       _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) {
+      _audioInterruptionObserver(nullptr) {
   LOGI() << "ctor" << ios::GetCurrentThreadDescription();
-  memset(_playoutBuffer, 0, sizeof(_playoutBuffer));
-  memset(_recordingBuffer, 0, sizeof(_recordingBuffer));
-  memset(_recordingLength, 0, sizeof(_recordingLength));
-  memset(_recordingSeqNumber, 0, sizeof(_recordingSeqNumber));
 }
 
 AudioDeviceIOS::~AudioDeviceIOS() {
   LOGI() << "~dtor";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   Terminate();
-  delete &_critSect;
 }
 
 void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   LOGI() << "AttachAudioBuffer";
   DCHECK(audioBuffer);
-  DCHECK(thread_checker_.CalledOnValidThread());
-  audio_device_buffer_ = audioBuffer;
+  DCHECK(_threadChecker.CalledOnValidThread());
+  _audioDeviceBuffer = audioBuffer;
 }
 
 int32_t AudioDeviceIOS::Init() {
   LOGI() << "Init";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   if (_initialized) {
     return 0;
   }
 #if !defined(NDEBUG)
   LogDeviceInfo();
 #endif
-  // 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);
+  // Store the preferred sample rate and preferred number of channels already
+  // here. They have not been set and confirmed yet since ActivateAudioSession()
+  // is not called until audio is about to start. However, it makes sense to
+  // store the parameters now and then verify at a later stage.
+  _playoutParameters.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
+  _recordParameters.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
+  // Ensure that the audio device buffer (ADB) knows about the internal audio
+  // parameters. Note that, even if we are unable to get a mono audio session,
+  // we will always tell the I/O audio unit to do a channel format conversion
+  // to guarantee mono on the "input side" of the audio unit.
+  UpdateAudioDeviceBuffer();
   _initialized = true;
   return 0;
 }
 
 int32_t AudioDeviceIOS::Terminate() {
   LOGI() << "Terminate";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   if (!_initialized) {
     return 0;
   }
-  // Stop the capture thread.
-  if (_captureWorkerThread) {
-    if (!_captureWorkerThread->Stop()) {
-      LOG_F(LS_ERROR) << "Failed to stop CaptureWorkerThread!";
-      return -1;
-    }
-    _captureWorkerThread.reset();
-  }
   ShutdownPlayOrRecord();
-  _isShutDown = true;
   _initialized = false;
   return 0;
 }
 
 int32_t AudioDeviceIOS::InitPlayout() {
   LOGI() << "InitPlayout";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   DCHECK(_initialized);
   DCHECK(!_playIsInitialized);
   DCHECK(!_playing);
   if (!_recIsInitialized) {
-    if (InitPlayOrRecord() == -1) {
+    if (!InitPlayOrRecord()) {
       LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
       return -1;
     }
   }
   _playIsInitialized = true;
   return 0;
 }
 
 int32_t AudioDeviceIOS::InitRecording() {
   LOGI() << "InitRecording";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   DCHECK(_initialized);
   DCHECK(!_recIsInitialized);
   DCHECK(!_recording);
   if (!_playIsInitialized) {
-    if (InitPlayOrRecord() == -1) {
+    if (!InitPlayOrRecord()) {
       LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
       return -1;
     }
   }
   _recIsInitialized = true;
   return 0;
 }
 
 int32_t AudioDeviceIOS::StartPlayout() {
   LOGI() << "StartPlayout";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.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;
-
+  _fineAudioBuffer->ResetPlayout();
   if (!_recording) {
-    OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
+    OSStatus result = AudioOutputUnitStart(_vpioUnit);
     if (result != noErr) {
       LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
       return -1;
     }
   }
-  _playing = true;
+  rtc::AtomicOps::ReleaseStore(&_playing, 1);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StopPlayout() {
   LOGI() << "StopPlayout";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   if (!_playIsInitialized || !_playing) {
     return 0;
   }
-
-  CriticalSectionScoped lock(&_critSect);
-
   if (!_recording) {
-    // Both playout and recording has stopped, shutdown the device.
     ShutdownPlayOrRecord();
   }
   _playIsInitialized = false;
-  _playing = false;
+  rtc::AtomicOps::ReleaseStore(&_playing, 0);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StartRecording() {
   LOGI() << "StartRecording";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.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;
-
+  _fineAudioBuffer->ResetRecord();
   if (!_playing) {
-    OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
+    OSStatus result = AudioOutputUnitStart(_vpioUnit);
     if (result != noErr) {
       LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
       return -1;
     }
   }
-  _recording = true;
+  rtc::AtomicOps::ReleaseStore(&_recording, 1);
   return 0;
 }
 
 int32_t AudioDeviceIOS::StopRecording() {
   LOGI() << "StopRecording";
-  DCHECK(thread_checker_.CalledOnValidThread());
+  DCHECK(_threadChecker.CalledOnValidThread());
   if (!_recIsInitialized || !_recording) {
     return 0;
   }
-
-  CriticalSectionScoped lock(&_critSect);
-
   if (!_playing) {
-    // Both playout and recording has stopped, shutdown the device.
     ShutdownPlayOrRecord();
   }
   _recIsInitialized = false;
-  _recording = false;
+  rtc::AtomicOps::ReleaseStore(&_recording, 0);
   return 0;
 }
 
 // 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;
@@ skipping 19 matching lines @@
 
 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::PlayoutDelay(uint16_t& delayMS) const {
-  delayMS = _playoutDelay;
+  delayMS = kFixedPlayoutDelayEstimate;
   return 0;
 }
 
 int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
-  delayMS = _recordingDelay;
-  return 0;
-}
-
-int32_t AudioDeviceIOS::PlayoutBuffer(AudioDeviceModule::BufferType& type,
-                                      uint16_t& sizeMS) const {
-  type = AudioDeviceModule::kAdaptiveBufferSize;
-  sizeMS = _playoutDelay;
+  delayMS = kFixedRecordDelayEstimate;
   return 0;
 }
 
 int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
-  CHECK(playout_parameters_.is_valid());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  *params = playout_parameters_;
+  LOGI() << "GetPlayoutAudioParameters";
+  DCHECK(_playoutParameters.is_valid());
+  DCHECK(_threadChecker.CalledOnValidThread());
+  *params = _playoutParameters;
   return 0;
 }
 
 int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
-  CHECK(record_parameters_.is_valid());
-  DCHECK(thread_checker_.CalledOnValidThread());
-  *params = record_parameters_;
-  return 0;
-}
-
-// ============================================================================
-//                                 Private Methods
-// ============================================================================
-
-int32_t AudioDeviceIOS::InitPlayOrRecord() {
-  LOGI() << "AudioDeviceIOS::InitPlayOrRecord";
-  DCHECK(!_auVoiceProcessing);
-
-  OSStatus result = -1;
-
-  // 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(nullptr, &desc);
-  if (nullptr == comp) {
-    LOG_F(LS_ERROR) << "Could not find audio component for Audio Unit";
-    return -1;
+  LOGI() << "GetRecordAudioParameters";
+  DCHECK(_recordParameters.is_valid());
+  DCHECK(_threadChecker.CalledOnValidThread());
+  *params = _recordParameters;
+  return 0;
+}
+
+void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
+  LOGI() << "UpdateAudioDevicebuffer";
+  // AttachAudioBuffer() is called at construction by the main class but check
+  // just in case.
+  DCHECK(_audioDeviceBuffer) << "AttachAudioBuffer must be called first";
+  // Inform the audio device buffer (ADB) about the new audio format.
+  _audioDeviceBuffer->SetPlayoutSampleRate(_playoutParameters.sample_rate());
+  _audioDeviceBuffer->SetPlayoutChannels(_playoutParameters.channels());
+  _audioDeviceBuffer->SetRecordingSampleRate(_recordParameters.sample_rate());
+  _audioDeviceBuffer->SetRecordingChannels(_recordParameters.channels());
+}
+
+void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
+  LOGI() << "SetupAudioBuffersForActiveAudioSession";
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  // Verify the current values once the audio session has been activated.
+  LOG(LS_INFO) << " sample rate: " << session.sampleRate;
+  LOG(LS_INFO) << " IO buffer duration: " << session.IOBufferDuration;
+  LOG(LS_INFO) << " output channels: " << session.outputNumberOfChannels;
+  LOG(LS_INFO) << " input channels: " << session.inputNumberOfChannels;
+  LOG(LS_INFO) << " output latency: " << session.outputLatency;
+  LOG(LS_INFO) << " input latency: " << session.inputLatency;
+  // Log a warning message for the case when we are unable to set the preferred
+  // hardware sample rate but continue and use the non-ideal sample rate after
+  // reinitializing the audio parameters.
+  if (session.sampleRate != _playoutParameters.sample_rate()) {
+    LOG(LS_WARNING)
+        << "Failed to enable an audio session with the preferred sample rate!";
   }
 
-  result = AudioComponentInstanceNew(comp, &_auVoiceProcessing);
-  if (0 != result) {
-    LOG_F(LS_ERROR) << "Failed to create Audio Unit instance: " << result;
-    return -1;
+  // At this stage, we also know the exact IO buffer duration and can add
+  // that info to the existing audio parameters where it is converted into
+  // number of audio frames.
+  // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
+  // Hence, 128 is the size we expect to see in upcoming render callbacks.
+  _playoutParameters.reset(session.sampleRate, _playoutParameters.channels(),
+                           session.IOBufferDuration);
+  DCHECK(_playoutParameters.is_complete());
+  _recordParameters.reset(session.sampleRate, _recordParameters.channels(),
+                          session.IOBufferDuration);
+  DCHECK(_recordParameters.is_complete());
+  LOG(LS_INFO) << " frames per I/O buffer: "
+               << _playoutParameters.frames_per_buffer();
+  LOG(LS_INFO) << " bytes per I/O buffer: "
+               << _playoutParameters.GetBytesPerBuffer();
+  DCHECK_EQ(_playoutParameters.GetBytesPerBuffer(),
+            _recordParameters.GetBytesPerBuffer());
+
+  // Update the ADB parameters since the sample rate might have changed.
+  UpdateAudioDeviceBuffer();
+
+  // Create a modified audio buffer class which allows us to ask for,
+  // or deliver, any number of samples (and not only multiple of 10ms) to match
+  // the native audio unit buffer size.
+  DCHECK(_audioDeviceBuffer);
+  _fineAudioBuffer.reset(new FineAudioBuffer(
+      _audioDeviceBuffer, _playoutParameters.GetBytesPerBuffer(),
+      _playoutParameters.sample_rate()));
+
+  // The extra/temporary playoutbuffer must be of this size to avoid
+  // unnecessary memcpy while caching data between successive callbacks.
+  const int requiredPlayoutBufferSize =
+      _fineAudioBuffer->RequiredPlayoutBufferSizeBytes();
+  LOG(LS_INFO) << " required playout buffer size: "
+               << requiredPlayoutBufferSize;
+  _playoutAudioBuffer.reset(new SInt8[requiredPlayoutBufferSize]);
+
+  // Allocate AudioBuffers to be used as storage for the received audio.
+  // The AudioBufferList structure works as a placeholder for the
+  // AudioBuffer structure, which holds a pointer to the actual data buffer
+  // in |_recordAudioBuffer|. Recorded audio will be rendered into this memory
+  // at each input callback when calling AudioUnitRender().
+  const int dataByteSize = _recordParameters.GetBytesPerBuffer();
+  _recordAudioBuffer.reset(new SInt8[dataByteSize]);
+  _audioRecordBufferList.mNumberBuffers = 1;
+  AudioBuffer* audioBuffer = &_audioRecordBufferList.mBuffers[0];
+  audioBuffer->mNumberChannels = _recordParameters.channels();
+  audioBuffer->mDataByteSize = dataByteSize;
+  audioBuffer->mData = _recordAudioBuffer.get();
+}
+
+bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
+  LOGI() << "SetupAndInitializeVoiceProcessingAudioUnit";
+  DCHECK(!_vpioUnit);
+  // Create an audio component description to identify the Voice-Processing
+  // I/O audio unit.
+  AudioComponentDescription vpioUnitDescription;
+  vpioUnitDescription.componentType = kAudioUnitType_Output;
+  vpioUnitDescription.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
+  vpioUnitDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
+  vpioUnitDescription.componentFlags = 0;
+  vpioUnitDescription.componentFlagsMask = 0;
+  // Obtain an audio unit instance given the description.
+  AudioComponent foundVpioUnitRef =
+      AudioComponentFindNext(nullptr, &vpioUnitDescription);
+
+  // Create a Voice-Processing IO audio unit.
+  LOG_AND_RETURN_IF_ERROR(
+      AudioComponentInstanceNew(foundVpioUnitRef, &_vpioUnit),
+      "Failed to create a VoiceProcessingIO audio unit");
+
+  // A VP I/O unit's bus 1 connects to input hardware (microphone). Enable
+  // input on the input scope of the input element.
+  AudioUnitElement inputBus = 1;
+  UInt32 enableInput = 1;
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_EnableIO,
+                           kAudioUnitScope_Input, inputBus, &enableInput,
+                           sizeof(enableInput)),
+      "Failed to enable input on input scope of input element");
+
+  // A VP I/O unit's bus 0 connects to output hardware (speaker). Enable
+  // output on the output scope of the output element.
+  AudioUnitElement outputBus = 0;
+  UInt32 enableOutput = 1;
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_EnableIO,
+                           kAudioUnitScope_Output, outputBus, &enableOutput,
+                           sizeof(enableOutput)),
+      "Failed to enable output on output scope of output element");
+
+  // Set the application formats for input and output:
+  // - use same format in both directions
+  // - avoid resampling in the I/O unit by using the hardware sample rate
+  // - linear PCM => noncompressed audio data format with one frame per packet
+  // - no need to specify interleaving since only mono is supported
+  AudioStreamBasicDescription applicationFormat = {0};
+  UInt32 size = sizeof(applicationFormat);
+  DCHECK_EQ(_playoutParameters.sample_rate(), _recordParameters.sample_rate());
+  DCHECK_EQ(1, kPreferredNumberOfChannels);
+  applicationFormat.mSampleRate = _playoutParameters.sample_rate();
+  applicationFormat.mFormatID = kAudioFormatLinearPCM;
+  applicationFormat.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+  applicationFormat.mBytesPerPacket = kBytesPerSample;
+  applicationFormat.mFramesPerPacket = 1;  // uncompressed
+  applicationFormat.mBytesPerFrame = kBytesPerSample;
+  applicationFormat.mChannelsPerFrame = kPreferredNumberOfChannels;
+  applicationFormat.mBitsPerChannel = 8 * kBytesPerSample;
+#if !defined(NDEBUG)
+  LogABSD(applicationFormat);
+#endif
+
+  // Set the application format on the output scope of the input element/bus.
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Output, inputBus, &applicationFormat,
+                           size),
+      "Failed to set application format on output scope of input element");
+
+  // Set the application format on the input scope of the output element/bus.
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Input, outputBus, &applicationFormat,
+                           size),
+      "Failed to set application format on input scope of output element");
+
+  // Specify the callback function that provides audio samples to the audio
+  // unit.
+  AURenderCallbackStruct renderCallback;
+  renderCallback.inputProc = GetPlayoutData;
+  renderCallback.inputProcRefCon = this;
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_SetRenderCallback,
+                           kAudioUnitScope_Input, outputBus, &renderCallback,
+                           sizeof(renderCallback)),
+      "Failed to specify the render callback on the output element");
+
+  // Disable AU buffer allocation for the recorder, we allocate our own.
+  // TODO(henrika): not sure that it actually saves resource to make this call.
+  UInt32 flag = 0;
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_ShouldAllocateBuffer,
+                           kAudioUnitScope_Output, inputBus, &flag,
+                           sizeof(flag)),
+      "Failed to disable buffer allocation on the input element");
+
+  // Specify the callback to be called by the I/O thread to us when input audio
+  // is available. The recorded samples can then be obtained by calling the
+  // AudioUnitRender() method.
+  AURenderCallbackStruct inputCallback;
+  inputCallback.inputProc = RecordedDataIsAvailable;
+  inputCallback.inputProcRefCon = this;
+  LOG_AND_RETURN_IF_ERROR(
+      AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_SetInputCallback,
+                           kAudioUnitScope_Global, inputBus, &inputCallback,
+                           sizeof(inputCallback)),
+      "Failed to specify the input callback on the input element");
+
+  // Initialize the Voice-Processing I/O unit instance.
+  LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(_vpioUnit),
+                          "Failed to initialize the Voice-Processing I/O unit");
+  return true;
+}
+
+bool AudioDeviceIOS::InitPlayOrRecord() {
+  LOGI() << "InitPlayOrRecord";
+  AVAudioSession* session = [AVAudioSession sharedInstance];
+  // Activate the audio session and ask for a set of preferred audio parameters.
+  ActivateAudioSession(session, true);
+
+  // Ensure that we got what what we asked for in our active audio session.
+  SetupAudioBuffersForActiveAudioSession();
+
+  // Create, setup and initialize a new Voice-Processing I/O unit.
+  if (!SetupAndInitializeVoiceProcessingAudioUnit()) {
+    return false;
   }
 
-  // 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];
@@ skipping 10 matching lines @@
                   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);
+                    AudioOutputUnitStop(_vpioUnit);
+                    AudioOutputUnitStart(_vpioUnit);
                     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;
-
-  return 0;
+  return true;
 }
 
-int32_t AudioDeviceIOS::ShutdownPlayOrRecord() {
+bool AudioDeviceIOS::ShutdownPlayOrRecord() {
   LOGI() << "ShutdownPlayOrRecord";
-
   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.
+  // Close and delete the voice-processing I/O unit.
   OSStatus result = -1;
-  if (nullptr != _auVoiceProcessing) {
-    result = AudioOutputUnitStop(_auVoiceProcessing);
-    if (0 != result) {
+  if (nullptr != _vpioUnit) {
+    result = AudioOutputUnitStop(_vpioUnit);
+    if (result != noErr) {
       LOG_F(LS_ERROR) << "AudioOutputUnitStop failed: " << result;
     }
-    result = AudioComponentInstanceDispose(_auVoiceProcessing);
-    if (0 != result) {
+    result = AudioComponentInstanceDispose(_vpioUnit);
+    if (result != noErr) {
       LOG_F(LS_ERROR) << "AudioComponentInstanceDispose failed: " << result;
     }
-    _auVoiceProcessing = nullptr;
+    _vpioUnit = nullptr;
   }
-
   // All I/O should be stopped or paused prior to deactivating the audio
   // session, hence we deactivate as last action.
   AVAudioSession* session = [AVAudioSession sharedInstance];
   ActivateAudioSession(session, false);
-  return 0;
+  return true;
 }
 
-// ============================================================================
-//                                  Thread Methods
-// ============================================================================
-
-OSStatus AudioDeviceIOS::RecordProcess(
+OSStatus AudioDeviceIOS::RecordedDataIsAvailable(
     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);
+  DCHECK_EQ(1u, inBusNumber);
+  DCHECK(!ioData);  // no buffer should be allocated for input at this stage
+  AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(inRefCon);
+  return audio_device_ios->OnRecordedDataIsAvailable(
+      ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames);
 }
 
-OSStatus AudioDeviceIOS::RecordProcessImpl(
+OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
     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;
+    UInt32 inBusNumber,
+    UInt32 inNumberFrames) {
+  DCHECK_EQ(_recordParameters.frames_per_buffer(), inNumberFrames);
+  OSStatus result = noErr;
+  // Simply return if recording is not enabled.
+  if (!rtc::AtomicOps::AcquireLoad(&_recording))
+    return result;
+  // Obtain the recorded audio samples by initiating a rendering cycle.
+  // Since it happens on the input bus, the |ioData| parameter is a reference
+  // to the preallocated audio buffer list that the audio unit renders into.
+  // TODO(henrika): should error handling be improved?
+  AudioBufferList* ioData = &_audioRecordBufferList;
+  result = AudioUnitRender(_vpioUnit, ioActionFlags, inTimeStamp, inBusNumber,
+                           inNumberFrames, ioData);
+  if (result != noErr) {
+    LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
+    return result;
   }
-
-  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
-        // TODO(henrika): improve error handling
-        dataPos = inNumberFrames;  // Don't try to insert more
-      }
-    }
-  }
-  delete[] dataTmp;
-  return 0;
+  // Get a pointer to the recorded audio and send it to the WebRTC ADB.
+  // Use the FineAudioBuffer instance to convert between native buffer size
+  // and the 10ms buffer size used by WebRTC.
+  const UInt32 dataSizeInBytes = ioData->mBuffers[0].mDataByteSize;
+  CHECK_EQ(dataSizeInBytes / kBytesPerSample, inNumberFrames);
+  SInt8* data = static_cast<SInt8*>(ioData->mBuffers[0].mData);
+  _fineAudioBuffer->DeliverRecordedData(data, dataSizeInBytes,
+                                        kFixedPlayoutDelayEstimate,
+                                        kFixedRecordDelayEstimate);
+  return noErr;
 }
 
-OSStatus AudioDeviceIOS::PlayoutProcess(
+OSStatus AudioDeviceIOS::GetPlayoutData(
     void* inRefCon,
     AudioUnitRenderActionFlags* ioActionFlags,
     const AudioTimeStamp* inTimeStamp,
     UInt32 inBusNumber,
     UInt32 inNumberFrames,
     AudioBufferList* ioData) {
-  AudioDeviceIOS* ptrThis = static_cast<AudioDeviceIOS*>(inRefCon);
-  return ptrThis->PlayoutProcessImpl(inNumberFrames, ioData);
+  DCHECK_EQ(0u, inBusNumber);
+  DCHECK(ioData);
+  AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(inRefCon);
+  return audio_device_ios->OnGetPlayoutData(ioActionFlags, 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;
-      }
-
-      // Update loop/index counter, if we copied less than noSamp10ms
-      // samples we shall quit loop anyway
-      dataPos += noSamp10ms;
-    }
-    delete[] dataTmp;
+OSStatus AudioDeviceIOS::OnGetPlayoutData(
+    AudioUnitRenderActionFlags* ioActionFlags,
+    UInt32 inNumberFrames,
+    AudioBufferList* ioData) {
+  // Verify 16-bit, noninterleaved mono PCM signal format.
+  DCHECK_EQ(1u, ioData->mNumberBuffers);
+  DCHECK_EQ(1u, ioData->mBuffers[0].mNumberChannels);
+  // Get pointer to internal audio buffer to which new audio data shall be
+  // written.
+  const UInt32 dataSizeInBytes = ioData->mBuffers[0].mDataByteSize;
+  CHECK_EQ(dataSizeInBytes / kBytesPerSample, inNumberFrames);
+  SInt8* destination = static_cast<SInt8*>(ioData->mBuffers[0].mData);
+  // Produce silence and give audio unit a hint about it if playout is not
+  // activated.
+  if (!rtc::AtomicOps::AcquireLoad(&_playing)) {
+    *ioActionFlags |= kAudioUnitRenderAction_OutputIsSilence;
+    memset(destination, 0, dataSizeInBytes);
+    return noErr;
   }
-  return 0;
-}
-
-// TODO(henrika): can either be removed or simplified.
-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) {
-      LOG_F(LS_ERROR) << "AU latency error: " << 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) {
-      LOG_F(LS_ERROR) << "AU latency error: " << 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;
-          }
-        }
-      }
-
-      // Insert data into the Audio Device Buffer if found any
-      if (foundBuf) {
-        // Update recording delay
-        UpdateRecordingDelay();
-
-        // Set the recorded buffer
-        audio_device_buffer_->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
-        audio_device_buffer_->SetVQEData(_playoutDelay, _recordingDelay, 0);
-
-        // Deliver recorded samples at specified sample rate, mic level
-        // etc. to the observer using callback
-        audio_device_buffer_->DeliverRecordedData();
-
-        // Make buffer available
-        _recordingSeqNumber[lowestSeqBufPos] = 0;
-        _recordingBufferTotalSize -= _recordingLength[lowestSeqBufPos];
-        // Must be done last to avoid interrupt problems between threads
-        _recordingLength[lowestSeqBufPos] = 0;
-      }
-    }
-  }
-
-  {
-    // 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;
+  // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
+  // the native I/O audio unit) to a preallocated intermediate buffer and
+  // copy the result to the audio buffer in the |ioData| destination.
+  SInt8* source = _playoutAudioBuffer.get();
+  _fineAudioBuffer->GetPlayoutData(source);
+  memcpy(destination, source, dataSizeInBytes);
+  return noErr;
 }
 
 }  // namespace webrtc