[Mac] Allow audio device selection (#21)

* first attempt * remove unused dep * init playout / recording * use AudioDeviceID as guid * switch device method * equality * default device * `isDefault` property * dont format default device name * type param * bypass * refactor * fix * append Audio to thread labels * ref * lk headers * low level apis * fix thread checks Some methods of ADM needs to be run on worker thread, otherwise RTC's thread check will fail. * switch to default device when removed * close mixerManager if didn't switch to default device * default audio device switched * expose devices update handler * fix ios compile * fix bug: don't always recreate RTCAudioDeviceModule

Author: hiroshihorie

modules/audio_device/audio_device_data_observer.cc

@@ -288,6 +288,10 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
  }
 #endif // WEBRTC_IOS
 
+ int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const override {
+ return impl_->SetAudioDeviceSink(sink);
+ }
+
  protected:
  rtc::scoped_refptr<AudioDeviceModule> impl_;
  AudioDeviceDataObserver* legacy_observer_ = nullptr;

modules/audio_device/audio_device_generic.cc

@@ -63,4 +63,10 @@ int AudioDeviceGeneric::GetRecordAudioParameters(
 }
 #endif // WEBRTC_IOS
 
+int32_t AudioDeviceGeneric::SetAudioDeviceSink(AudioDeviceSink* sink) {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << sink << ")";
+ audio_device_module_sink_ = sink;
+ return 0;
+}
+
 } // namespace webrtc

modules/audio_device/audio_device_generic.h

@@ -135,9 +135,14 @@ class AudioDeviceGeneric {
  virtual int GetRecordAudioParameters(AudioParameters* params) const;
 #endif // WEBRTC_IOS
 
+ int32_t SetAudioDeviceSink(AudioDeviceSink* sink);
+
  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
 
  virtual ~AudioDeviceGeneric() {}
+
+ protected:
+ AudioDeviceSink* audio_device_module_sink_ = nullptr;
 };
 
 } // namespace webrtc

modules/audio_device/audio_device_impl.cc

@@ -72,15 +72,17 @@ namespace webrtc {
 
 rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
  AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory) {
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
- return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory);
+ return AudioDeviceModule::CreateForTest(audio_layer, task_queue_factory, bypass_voice_processing);
 }
 
 // static
 rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
  AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory) {
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
 
  // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
@@ -93,7 +95,7 @@ rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
 
  // Create the generic reference counted (platform independent) implementation.
  auto audio_device = rtc::make_ref_counted<AudioDeviceModuleImpl>(
- audio_layer, task_queue_factory);
+ audio_layer, task_queue_factory, bypass_voice_processing);
 
  // Ensure that the current platform is supported.
  if (audio_device->CheckPlatform() == -1) {
@@ -116,8 +118,13 @@ rtc::scoped_refptr<AudioDeviceModuleForTest> AudioDeviceModule::CreateForTest(
 
 AudioDeviceModuleImpl::AudioDeviceModuleImpl(
  AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory)
- : audio_layer_(audio_layer), audio_device_buffer_(task_queue_factory) {
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing)
+ : audio_layer_(audio_layer),
+ #if defined(WEBRTC_IOS)
+ bypass_voice_processing_(bypass_voice_processing),
+ #endif
+ audio_device_buffer_(task_queue_factory) {
  RTC_DLOG(LS_INFO) << __FUNCTION__;
 }
 
@@ -280,7 +287,7 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
 #if defined(WEBRTC_IOS)
  if (audio_layer == kPlatformDefaultAudio) {
  audio_device_.reset(
- new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
+ new ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/bypass_voice_processing_));
  RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
  }
 // END #if defined(WEBRTC_IOS)
@@ -937,6 +944,13 @@ int AudioDeviceModuleImpl::GetRecordAudioParameters(
 }
 #endif // WEBRTC_IOS
 
+int32_t AudioDeviceModuleImpl::SetAudioDeviceSink(AudioDeviceSink* sink) const {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "(" << sink << ")";
+ int32_t ok = audio_device_->SetAudioDeviceSink(sink);
+ RTC_LOG(LS_INFO) << "output: " << ok;
+ return ok;
+}
+
 AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const {
  RTC_LOG(LS_INFO) << __FUNCTION__;
  return platform_type_;

modules/audio_device/audio_device_impl.h

@@ -43,7 +43,8 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
  int32_t AttachAudioBuffer();
 
  AudioDeviceModuleImpl(AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory);
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing = false);
  ~AudioDeviceModuleImpl() override;
 
  // Retrieve the currently utilized audio layer
@@ -145,6 +146,8 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
  int GetRecordAudioParameters(AudioParameters* params) const override;
 #endif // WEBRTC_IOS
 
+ int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const override;
+
 #if defined(WEBRTC_ANDROID)
  // Only use this acccessor for test purposes on Android.
  AudioManager* GetAndroidAudioManagerForTest() {
@@ -165,7 +168,9 @@ class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
  AudioLayer audio_layer_;
  PlatformType platform_type_ = kPlatformNotSupported;
  bool initialized_ = false;
-#if defined(WEBRTC_ANDROID)
+#if defined(WEBRTC_IOS)
+ bool bypass_voice_processing_;
+#elif defined(WEBRTC_ANDROID)
  // Should be declared first to ensure that it outlives other resources.
  std::unique_ptr<AudioManager> audio_manager_android_;
 #endif

modules/audio_device/include/audio_device.h

@@ -20,6 +20,15 @@ namespace webrtc {
 
 class AudioDeviceModuleForTest;
 
+// Sink for callbacks related to a audio device.
+class AudioDeviceSink {
+ public:
+ virtual ~AudioDeviceSink() = default;
+
+ // input/output devices updated or default device changed
+ virtual void OnDevicesUpdated() {}
+};
+
 class AudioDeviceModule : public rtc::RefCountInterface {
  public:
  enum AudioLayer {
@@ -45,12 +54,14 @@ class AudioDeviceModule : public rtc::RefCountInterface {
  // Creates a default ADM for usage in production code.
  static rtc::scoped_refptr<AudioDeviceModule> Create(
  AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory);
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing = false);
  // Creates an ADM with support for extra test methods. Don't use this factory
  // in production code.
  static rtc::scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
  AudioLayer audio_layer,
- TaskQueueFactory* task_queue_factory);
+ TaskQueueFactory* task_queue_factory,
+ bool bypass_voice_processing = false);
 
  // Retrieve the currently utilized audio layer
  virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
@@ -156,6 +167,8 @@ class AudioDeviceModule : public rtc::RefCountInterface {
  virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
 #endif // WEBRTC_IOS
 
+ virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const = 0;
+
  protected:
  ~AudioDeviceModule() override {}
 };

modules/audio_device/include/test_audio_device.cc

@@ -141,6 +141,11 @@ class TestAudioDeviceModuleImpl
  return capturing_;
  }
 
+ int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const override {
+ // no-op
+ return 0;
+ }
+
  // Blocks until the Renderer refuses to receive data.
  // Returns false if `timeout_ms` passes before that happens.
  bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever) override {