Cherry pick audio selection from m97 release (#35)

* [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

* handle guid.

Co-authored-by: Hiroshi Horie <[email protected]>

Author: cloudwebrtc

modules/audio_device/android/audio_device_template.h

@@ -282,11 +282,6 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
     RTC_CHECK_NOTREACHED();
   }
 
-  int32_t SetAudioDeviceSink(AudioDeviceSink* sink) override {
-    RTC_CHECK_NOTREACHED();
-    return -1;
-  }
-
   int32_t SetSpeakerMute(bool enable) override { RTC_CHECK_NOTREACHED(); }
 
   int32_t SpeakerMute(bool& enabled) const override { RTC_CHECK_NOTREACHED(); }

modules/audio_device/audio_device_data_observer.cc

@@ -307,6 +307,10 @@ 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.h

@@ -135,7 +135,7 @@ class AudioDeviceGeneric {
   virtual int GetRecordAudioParameters(AudioParameters* params) const;
 #endif  // WEBRTC_IOS
 
-  virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) = 0;
+  virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) { return -1; }
 
   virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
 

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/dummy/audio_device_dummy.cc

@@ -222,9 +222,5 @@ int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const {
   return -1;
 }
 
-int32_t AudioDeviceDummy::SetAudioDeviceSink(AudioDeviceSink* sink) {
-  return -1;
-}
-
 void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
 }  // namespace webrtc

modules/audio_device/dummy/audio_device_dummy.h

@@ -109,8 +109,6 @@ class AudioDeviceDummy : public AudioDeviceGeneric {
   // Delay information and control
   int32_t PlayoutDelay(uint16_t& delayMS) const override;
 
-  int32_t SetAudioDeviceSink(AudioDeviceSink* sink) override;
-
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
 };

modules/audio_device/dummy/file_audio_device.cc

@@ -427,10 +427,6 @@ int32_t FileAudioDevice::PlayoutDelay(uint16_t& delayMS) const {
   return 0;
 }
 
-int32_t FileAudioDevice::SetAudioDeviceSink(AudioDeviceSink* sink) {
-  return -1;
-}
-
 void FileAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   MutexLock lock(&mutex_);
 

modules/audio_device/dummy/file_audio_device.h

@@ -123,8 +123,6 @@ class FileAudioDevice : public AudioDeviceGeneric {
   // Delay information and control
   int32_t PlayoutDelay(uint16_t& delayMS) const override;
 
-  int32_t SetAudioDeviceSink(AudioDeviceSink* sink) override;
-
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
  private:

modules/audio_device/include/audio_device.h

@@ -65,12 +65,14 @@ 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;
@@ -180,6 +182,8 @@ class AudioDeviceModule : public rtc::RefCountInterface {
   virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
 #endif  // WEBRTC_IOS
 
+  virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const { return -1; }
+
  protected:
   ~AudioDeviceModule() override {}
 };