/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "modules/audio_device/audio_device_impl.h"

#include <stddef.h>

#include "api/make_ref_counted.h"
#include "api/scoped_refptr.h"
#include "modules/audio_device/audio_device_config.h"  // IWYU pragma: keep
#include "modules/audio_device/audio_device_generic.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/metrics.h"

#if defined(_WIN32)
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
#include "modules/audio_device/win/audio_device_core_win.h"
#endif
#elif defined(WEBRTC_ANDROID)
#include <stdlib.h>
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
#include "modules/audio_device/android/aaudio_player.h"
#include "modules/audio_device/android/aaudio_recorder.h"
#endif
#include "modules/audio_device/android/audio_device_template.h"
#include "modules/audio_device/android/audio_manager.h"
#include "modules/audio_device/android/audio_record_jni.h"
#include "modules/audio_device/android/audio_track_jni.h"
#include "modules/audio_device/android/opensles_player.h"
#include "modules/audio_device/android/opensles_recorder.h"
#elif defined(WEBRTC_LINUX)
#if defined(WEBRTC_ENABLE_LINUX_ALSA)
#include "modules/audio_device/linux/audio_device_alsa_linux.h"
#endif
#if defined(WEBRTC_ENABLE_LINUX_PULSE)
#include "modules/audio_device/linux/audio_device_pulse_linux.h"
#endif
#elif defined(WEBRTC_IOS)
#include "sdk/objc/native/src/audio/audio_device_ios.h"
#elif defined(WEBRTC_MAC)
#include "modules/audio_device/mac/audio_device_mac.h"
#endif
#if defined(WEBRTC_DUMMY_FILE_DEVICES)
#include "modules/audio_device/dummy/file_audio_device.h"
#include "modules/audio_device/dummy/file_audio_device_factory.h"
#endif
#include "modules/audio_device/dummy/audio_device_dummy.h"

#define CHECKinitialized_() \
  {                         \
    if (!initialized_) {    \
      return -1;            \
    }                       \
  }

#define CHECKinitialized__BOOL() \
  {                              \
    if (!initialized_) {         \
      return false;              \
    }                            \
  }

namespace webrtc {

rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
    AudioLayer audio_layer,
    TaskQueueFactory* task_queue_factory) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
  return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
}

// static
rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
    AudioLayer audio_layer,
    TaskQueueFactory* task_queue_factory) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;

  // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
  // dedicated factory method which should be used instead.
  if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
    RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() "
                         "factory method instead for this option.";
    return nullptr;
  }

  // Create the generic reference counted (platform independent) implementation.
  auto audio_device = rtc::make_ref_counted<AudioDeviceModuleImpl>(
      audio_layer, task_queue_factory);

  // Ensure that the current platform is supported.
  if (audio_device->CheckPlatform() == -1) {
    return nullptr;
  }

  // Create the platform-dependent implementation.
  if (audio_device->CreatePlatformSpecificObjects() == -1) {
    return nullptr;
  }

  // Ensure that the generic audio buffer can communicate with the platform
  // specific parts.
  if (audio_device->AttachAudioBuffer() == -1) {
    return nullptr;
  }

  return audio_device;
}

AudioDeviceModuleImpl::AudioDeviceModuleImpl(
    AudioLayer audio_layer,
    TaskQueueFactory* task_queue_factory)
    : audio_layer_(audio_layer), audio_device_buffer_(task_queue_factory) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
}

int32_t AudioDeviceModuleImpl::CheckPlatform() {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
  // Ensure that the current platform is supported
  PlatformType platform(kPlatformNotSupported);
#if defined(_WIN32)
  platform = kPlatformWin32;
  RTC_LOG(LS_INFO) << "current platform is Win32";
#elif defined(WEBRTC_ANDROID)
  platform = kPlatformAndroid;
  RTC_LOG(LS_INFO) << "current platform is Android";
#elif defined(WEBRTC_LINUX)
  platform = kPlatformLinux;
  RTC_LOG(LS_INFO) << "current platform is Linux";
#elif defined(WEBRTC_IOS)
  platform = kPlatformIOS;
  RTC_LOG(LS_INFO) << "current platform is IOS";
#elif defined(WEBRTC_MAC)
  platform = kPlatformMac;
  RTC_LOG(LS_INFO) << "current platform is Mac";
#endif
  if (platform == kPlatformNotSupported) {
    RTC_LOG(LS_ERROR)
        << "current platform is not supported => this module will self "
           "destruct!";
    return -1;
  }
  platform_type_ = platform;
  return 0;
}

int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
// Dummy ADM implementations if build flags are set.
#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
  audio_device_.reset(new AudioDeviceDummy());
  RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
#elif defined(WEBRTC_DUMMY_FILE_DEVICES)
  audio_device_.reset(FileAudioDeviceFactory::CreateFileAudioDevice());
  if (audio_device_) {
    RTC_LOG(LS_INFO) << "Will use file-playing dummy device.";
  } else {
    // Create a dummy device instead.
    audio_device_.reset(new AudioDeviceDummy());
    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized";
  }

// Real (non-dummy) ADM implementations.
#else
  AudioLayer audio_layer(PlatformAudioLayer());
// Windows ADM implementation.
#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
  if ((audio_layer == kWindowsCoreAudio) ||
      (audio_layer == kPlatformDefaultAudio)) {
    RTC_LOG(LS_INFO) << "Attempting to use the Windows Core Audio APIs...";
    if (AudioDeviceWindowsCore::CoreAudioIsSupported()) {
      audio_device_.reset(new AudioDeviceWindowsCore());
      RTC_LOG(LS_INFO) << "Windows Core Audio APIs will be utilized";
    }
  }
#endif  // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)

#if defined(WEBRTC_ANDROID)
  // Create an Android audio manager.
  audio_manager_android_.reset(new AudioManager());
  // Select best possible combination of audio layers.
  if (audio_layer == kPlatformDefaultAudio) {
    if (audio_manager_android_->IsAAudioSupported()) {
      // Use of AAudio for both playout and recording has highest priority.
      audio_layer = kAndroidAAudioAudio;
    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
               audio_manager_android_->IsLowLatencyRecordSupported()) {
      // Use OpenSL ES for both playout and recording.
      audio_layer = kAndroidOpenSLESAudio;
    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
               !audio_manager_android_->IsLowLatencyRecordSupported()) {
      // Use OpenSL ES for output on devices that only supports the
      // low-latency output audio path.
      audio_layer = kAndroidJavaInputAndOpenSLESOutputAudio;
    } else {
      // Use Java-based audio in both directions when low-latency output is
      // not supported.
      audio_layer = kAndroidJavaAudio;
    }
  }
  AudioManager* audio_manager = audio_manager_android_.get();
  if (audio_layer == kAndroidJavaAudio) {
    // Java audio for both input and output audio.
    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AudioTrackJni>(
        audio_layer, audio_manager));
  } else if (audio_layer == kAndroidOpenSLESAudio) {
    // OpenSL ES based audio for both input and output audio.
    audio_device_.reset(
        new AudioDeviceTemplate<OpenSLESRecorder, OpenSLESPlayer>(
            audio_layer, audio_manager));
  } else if (audio_layer == kAndroidJavaInputAndOpenSLESOutputAudio) {
    // Java audio for input and OpenSL ES for output audio (i.e. mixed APIs).
    // This combination provides low-latency output audio and at the same
    // time support for HW AEC using the AudioRecord Java API.
    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
        audio_layer, audio_manager));
  } else if (audio_layer == kAndroidAAudioAudio) {
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
    // AAudio based audio for both input and output.
    audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
        audio_layer, audio_manager));
#endif
  } else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
#if defined(WEBRTC_AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
    // Java audio for input and AAudio for output audio (i.e. mixed APIs).
    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
        audio_layer, audio_manager));
#endif
  } else {
    RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
    audio_device_.reset(nullptr);
  }
// END #if defined(WEBRTC_ANDROID)

// Linux ADM implementation.
// Note that, WEBRTC_ENABLE_LINUX_ALSA is always defined by default when
// WEBRTC_LINUX is defined. WEBRTC_ENABLE_LINUX_PULSE depends on the
// 'rtc_include_pulse_audio' build flag.
// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
// PulseAudio is the default selection.
#elif defined(WEBRTC_LINUX)
#if !defined(WEBRTC_ENABLE_LINUX_PULSE)
  // Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
  // - kPlatformDefaultAudio => ALSA, and
  // - kLinuxAlsaAudio => ALSA, and
  // - kLinuxPulseAudio => Invalid selection.
  RTC_LOG(LS_WARNING) << "PulseAudio is disabled using build flag.";
  if ((audio_layer == kLinuxAlsaAudio) ||
      (audio_layer == kPlatformDefaultAudio)) {
    audio_device_.reset(new AudioDeviceLinuxALSA());
    RTC_LOG(LS_INFO) << "Linux ALSA APIs will be utilized.";
  }
#else
  // Build flag 'rtc_include_pulse_audio' is set to true (default). In this
  // mode:
  // - kPlatformDefaultAudio => PulseAudio, and
  // - kLinuxPulseAudio => PulseAudio, and
  // - kLinuxAlsaAudio => ALSA (supported but not default).
  RTC_LOG(LS_INFO) << "PulseAudio support is enabled.";
  if ((audio_layer == kLinuxPulseAudio) ||
      (audio_layer == kPlatformDefaultAudio)) {
    // Linux PulseAudio implementation is default.
    audio_device_.reset(new AudioDeviceLinuxPulse());
    RTC_LOG(LS_INFO) << "Linux PulseAudio APIs will be utilized";
  } else if (audio_layer == kLinuxAlsaAudio) {
    audio_device_.reset(new AudioDeviceLinuxALSA());
    RTC_LOG(LS_WARNING) << "Linux ALSA APIs will be utilized.";
  }
#endif  // #if !defined(WEBRTC_ENABLE_LINUX_PULSE)
#endif  // #if defined(WEBRTC_LINUX)

// iOS ADM implementation.
#if defined(WEBRTC_IOS)
  if (audio_layer == kPlatformDefaultAudio) {
    audio_device_.reset(
        new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
    RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
  }
// END #if defined(WEBRTC_IOS)

// Mac OS X ADM implementation.
#elif defined(WEBRTC_MAC)
  if (audio_layer == kPlatformDefaultAudio) {
    audio_device_.reset(new AudioDeviceMac());
    RTC_LOG(LS_INFO) << "Mac OS X Audio APIs will be utilized.";
  }
#endif  // WEBRTC_MAC

  // Dummy ADM implementation.
  if (audio_layer == kDummyAudio) {
    audio_device_.reset(new AudioDeviceDummy());
    RTC_LOG(LS_INFO) << "Dummy Audio APIs will be utilized.";
  }
#endif  // if defined(WEBRTC_DUMMY_AUDIO_BUILD)

  if (!audio_device_) {
    RTC_LOG(LS_ERROR)
        << "Failed to create the platform specific ADM implementation.";
    return -1;
  }
  return 0;
}

int32_t AudioDeviceModuleImpl::AttachAudioBuffer() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  audio_device_->AttachAudioBuffer(&audio_device_buffer_);
  return 0;
}

AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
}

int32_t AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  AudioLayer activeAudio;
  if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
    return -1;
  }
  *audioLayer = activeAudio;
  return 0;
}

int32_t AudioDeviceModuleImpl::Init() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  if (initialized_)
    return 0;
  RTC_CHECK(audio_device_);
  AudioDeviceGeneric::InitStatus status = audio_device_->Init();
  RTC_HISTOGRAM_ENUMERATION(
      "WebRTC.Audio.InitializationResult", static_cast<int>(status),
      static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
  if (status != AudioDeviceGeneric::InitStatus::OK) {
    RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
    return -1;
  }
  initialized_ = true;
  return 0;
}

int32_t AudioDeviceModuleImpl::Terminate() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  if (!initialized_)
    return 0;
  if (audio_device_->Terminate() == -1) {
    return -1;
  }
  initialized_ = false;
  return 0;
}

bool AudioDeviceModuleImpl::Initialized() const {
  RTC_LOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
  return initialized_;
}

int32_t AudioDeviceModuleImpl::InitSpeaker() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  return audio_device_->InitSpeaker();
}

int32_t AudioDeviceModuleImpl::InitMicrophone() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  return audio_device_->InitMicrophone();
}

int32_t AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetSpeakerVolume(uint32_t volume) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
  CHECKinitialized_();
  return audio_device_->SetSpeakerVolume(volume);
}

int32_t AudioDeviceModuleImpl::SpeakerVolume(uint32_t* volume) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  uint32_t level = 0;
  if (audio_device_->SpeakerVolume(level) == -1) {
    return -1;
  }
  *volume = level;
  RTC_LOG(LS_INFO) << "output: " << *volume;
  return 0;
}

bool AudioDeviceModuleImpl::SpeakerIsInitialized() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  bool isInitialized = audio_device_->SpeakerIsInitialized();
  RTC_LOG(LS_INFO) << "output: " << isInitialized;
  return isInitialized;
}

bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  bool isInitialized = audio_device_->MicrophoneIsInitialized();
  RTC_LOG(LS_INFO) << "output: " << isInitialized;
  return isInitialized;
}

int32_t AudioDeviceModuleImpl::MaxSpeakerVolume(uint32_t* maxVolume) const {
  CHECKinitialized_();
  uint32_t maxVol = 0;
  if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
    return -1;
  }
  *maxVolume = maxVol;
  return 0;
}

int32_t AudioDeviceModuleImpl::MinSpeakerVolume(uint32_t* minVolume) const {
  CHECKinitialized_();
  uint32_t minVol = 0;
  if (audio_device_->MinSpeakerVolume(minVol) == -1) {
    return -1;
  }
  *minVolume = minVol;
  return 0;
}

int32_t AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetSpeakerMute(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  return audio_device_->SetSpeakerMute(enable);
}

int32_t AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool muted = false;
  if (audio_device_->SpeakerMute(muted) == -1) {
    return -1;
  }
  *enabled = muted;
  RTC_LOG(LS_INFO) << "output: " << muted;
  return 0;
}

int32_t AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetMicrophoneMute(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  return (audio_device_->SetMicrophoneMute(enable));
}

int32_t AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool muted = false;
  if (audio_device_->MicrophoneMute(muted) == -1) {
    return -1;
  }
  *enabled = muted;
  RTC_LOG(LS_INFO) << "output: " << muted;
  return 0;
}

int32_t AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetMicrophoneVolume(uint32_t volume) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
  CHECKinitialized_();
  return (audio_device_->SetMicrophoneVolume(volume));
}

int32_t AudioDeviceModuleImpl::MicrophoneVolume(uint32_t* volume) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  uint32_t level = 0;
  if (audio_device_->MicrophoneVolume(level) == -1) {
    return -1;
  }
  *volume = level;
  RTC_LOG(LS_INFO) << "output: " << *volume;
  return 0;
}

int32_t AudioDeviceModuleImpl::StereoRecordingIsAvailable(
    bool* available) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetStereoRecording(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  if (audio_device_->RecordingIsInitialized()) {
    RTC_LOG(LS_ERROR)
        << "unable to set stereo mode after recording is initialized";
    return -1;
  }
  if (audio_device_->SetStereoRecording(enable) == -1) {
    if (enable) {
      RTC_LOG(LS_WARNING) << "failed to enable stereo recording";
    }
    return -1;
  }
  int8_t nChannels(1);
  if (enable) {
    nChannels = 2;
  }
  audio_device_buffer_.SetRecordingChannels(nChannels);
  return 0;
}

int32_t AudioDeviceModuleImpl::StereoRecording(bool* enabled) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool stereo = false;
  if (audio_device_->StereoRecording(stereo) == -1) {
    return -1;
  }
  *enabled = stereo;
  RTC_LOG(LS_INFO) << "output: " << stereo;
  return 0;
}

int32_t AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::SetStereoPlayout(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  if (audio_device_->PlayoutIsInitialized()) {
    RTC_LOG(LS_ERROR)
        << "unable to set stereo mode while playing side is initialized";
    return -1;
  }
  if (audio_device_->SetStereoPlayout(enable)) {
    RTC_LOG(LS_WARNING) << "stereo playout is not supported";
    return -1;
  }
  int8_t nChannels(1);
  if (enable) {
    nChannels = 2;
  }
  audio_device_buffer_.SetPlayoutChannels(nChannels);
  return 0;
}

int32_t AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool stereo = false;
  if (audio_device_->StereoPlayout(stereo) == -1) {
    return -1;
  }
  *enabled = stereo;
  RTC_LOG(LS_INFO) << "output: " << stereo;
  return 0;
}

int32_t AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::RecordingIsAvailable(bool* available) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  bool isAvailable = false;
  if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
    return -1;
  }
  *available = isAvailable;
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return 0;
}

int32_t AudioDeviceModuleImpl::MaxMicrophoneVolume(uint32_t* maxVolume) const {
  CHECKinitialized_();
  uint32_t maxVol(0);
  if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
    return -1;
  }
  *maxVolume = maxVol;
  return 0;
}

int32_t AudioDeviceModuleImpl::MinMicrophoneVolume(uint32_t* minVolume) const {
  CHECKinitialized_();
  uint32_t minVol(0);
  if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
    return -1;
  }
  *minVolume = minVol;
  return 0;
}

int16_t AudioDeviceModuleImpl::PlayoutDevices() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
  RTC_LOG(LS_INFO) << "output: " << nPlayoutDevices;
  return (int16_t)(nPlayoutDevices);
}

int32_t AudioDeviceModuleImpl::SetPlayoutDevice(uint16_t index) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
  CHECKinitialized_();
  return audio_device_->SetPlayoutDevice(index);
}

int32_t AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  return audio_device_->SetPlayoutDevice(device);
}

int32_t AudioDeviceModuleImpl::PlayoutDeviceName(
    uint16_t index,
    char name[kAdmMaxDeviceNameSize],
    char guid[kAdmMaxGuidSize]) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
  CHECKinitialized_();
  if (name == NULL) {
    return -1;
  }
  if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
    return -1;
  }
  if (name != NULL) {
    RTC_LOG(LS_INFO) << "output: name = " << name;
  }
  if (guid != NULL) {
    RTC_LOG(LS_INFO) << "output: guid = " << guid;
  }
  return 0;
}

int32_t AudioDeviceModuleImpl::RecordingDeviceName(
    uint16_t index,
    char name[kAdmMaxDeviceNameSize],
    char guid[kAdmMaxGuidSize]) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
  CHECKinitialized_();
  if (name == NULL) {
    return -1;
  }
  if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
    return -1;
  }
  if (name != NULL) {
    RTC_LOG(LS_INFO) << "output: name = " << name;
  }
  if (guid != NULL) {
    RTC_LOG(LS_INFO) << "output: guid = " << guid;
  }
  return 0;
}

int16_t AudioDeviceModuleImpl::RecordingDevices() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  uint16_t nRecordingDevices = audio_device_->RecordingDevices();
  RTC_LOG(LS_INFO) << "output: " << nRecordingDevices;
  return (int16_t)nRecordingDevices;
}

int32_t AudioDeviceModuleImpl::SetRecordingDevice(uint16_t index) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
  CHECKinitialized_();
  return audio_device_->SetRecordingDevice(index);
}

int32_t AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  return audio_device_->SetRecordingDevice(device);
}

int32_t AudioDeviceModuleImpl::InitPlayout() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  if (PlayoutIsInitialized()) {
    return 0;
  }
  int32_t result = audio_device_->InitPlayout();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
                        static_cast<int>(result == 0));
  return result;
}

int32_t AudioDeviceModuleImpl::InitRecording() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  if (RecordingIsInitialized()) {
    return 0;
  }
  int32_t result = audio_device_->InitRecording();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
                        static_cast<int>(result == 0));
  return result;
}

bool AudioDeviceModuleImpl::PlayoutIsInitialized() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  return audio_device_->PlayoutIsInitialized();
}

bool AudioDeviceModuleImpl::RecordingIsInitialized() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  return audio_device_->RecordingIsInitialized();
}

int32_t AudioDeviceModuleImpl::StartPlayout() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  if (Playing()) {
    return 0;
  }
  audio_device_buffer_.StartPlayout();
  int32_t result = audio_device_->StartPlayout();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
                        static_cast<int>(result == 0));
  return result;
}

int32_t AudioDeviceModuleImpl::StopPlayout() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  int32_t result = audio_device_->StopPlayout();
  audio_device_buffer_.StopPlayout();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
                        static_cast<int>(result == 0));
  return result;
}

bool AudioDeviceModuleImpl::Playing() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  return audio_device_->Playing();
}

int32_t AudioDeviceModuleImpl::StartRecording() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  if (Recording()) {
    return 0;
  }
  audio_device_buffer_.StartRecording();
  int32_t result = audio_device_->StartRecording();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
                        static_cast<int>(result == 0));
  return result;
}

int32_t AudioDeviceModuleImpl::StopRecording() {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  int32_t result = audio_device_->StopRecording();
  audio_device_buffer_.StopRecording();
  RTC_LOG(LS_INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
                        static_cast<int>(result == 0));
  return result;
}

bool AudioDeviceModuleImpl::Recording() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  return audio_device_->Recording();
}

int32_t AudioDeviceModuleImpl::RegisterAudioCallback(
    AudioTransport* audioCallback) {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  return audio_device_buffer_.RegisterAudioCallback(audioCallback);
}

int32_t AudioDeviceModuleImpl::PlayoutDelay(uint16_t* delayMS) const {
  CHECKinitialized_();
  uint16_t delay = 0;
  if (audio_device_->PlayoutDelay(delay) == -1) {
    RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
    return -1;
  }
  *delayMS = delay;
  return 0;
}

bool AudioDeviceModuleImpl::BuiltInAECIsAvailable() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  bool isAvailable = audio_device_->BuiltInAECIsAvailable();
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return isAvailable;
}

int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  int32_t ok = audio_device_->EnableBuiltInAEC(enable);
  RTC_LOG(LS_INFO) << "output: " << ok;
  return ok;
}

bool AudioDeviceModuleImpl::BuiltInAGCIsAvailable() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return isAvailable;
}

int32_t AudioDeviceModuleImpl::EnableBuiltInAGC(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  int32_t ok = audio_device_->EnableBuiltInAGC(enable);
  RTC_LOG(LS_INFO) << "output: " << ok;
  return ok;
}

bool AudioDeviceModuleImpl::BuiltInNSIsAvailable() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized__BOOL();
  bool isAvailable = audio_device_->BuiltInNSIsAvailable();
  RTC_LOG(LS_INFO) << "output: " << isAvailable;
  return isAvailable;
}

int32_t AudioDeviceModuleImpl::EnableBuiltInNS(bool enable) {
  RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
  CHECKinitialized_();
  int32_t ok = audio_device_->EnableBuiltInNS(enable);
  RTC_LOG(LS_INFO) << "output: " << ok;
  return ok;
}

int32_t AudioDeviceModuleImpl::GetPlayoutUnderrunCount() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  CHECKinitialized_();
  int32_t underrunCount = audio_device_->GetPlayoutUnderrunCount();
  RTC_LOG(LS_INFO) << "output: " << underrunCount;
  return underrunCount;
}

#if defined(WEBRTC_IOS)
int AudioDeviceModuleImpl::GetPlayoutAudioParameters(
    AudioParameters* params) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  int r = audio_device_->GetPlayoutAudioParameters(params);
  RTC_LOG(LS_INFO) << "output: " << r;
  return r;
}

int AudioDeviceModuleImpl::GetRecordAudioParameters(
    AudioParameters* params) const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  int r = audio_device_->GetRecordAudioParameters(params);
  RTC_LOG(LS_INFO) << "output: " << r;
  return r;
}
#endif  // WEBRTC_IOS

AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  return platform_type_;
}

AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer()
    const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  return audio_layer_;
}

}  // namespace webrtc