Enable audio testability through RTCPeerConnectionFactory

sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h

@@ -9,6 +9,7 @@
  */
 
 #import "RTCPeerConnectionFactory.h"
+#import <AVFoundation/AVFoundation.h>
 
 #include "api/peer_connection_interface.h"
 #include "api/scoped_refptr.h"
@@ -18,13 +19,18 @@ NS_ASSUME_NONNULL_BEGIN
 @interface RTC_OBJC_TYPE (RTCPeerConnectionFactory)
 ()
 
-    /**
-     * PeerConnectionFactoryInterface created and held by this
-     * RTCPeerConnectionFactory object. This is needed to pass to the underlying
-     * C++ APIs.
-     */
-    @property(nonatomic,
-              readonly) rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+
+
+/**
+ * PeerConnectionFactoryInterface created and held by this
+ * RTCPeerConnectionFactory object. This is needed to pass to the underlying
+ * C++ APIs.
+ */
+@property(nonatomic,
+        readonly) rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+
+@property(nonatomic,
+        readonly) rtc::scoped_refptr<webrtc::AudioDeviceModule> internalAudioDeviceModule;
 
 @end
 

sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h

@@ -9,6 +9,7 @@
  */
 
 #import <Foundation/Foundation.h>
+#import <AVFoundation/AVFoundation.h>
 
 #import "RTCMacros.h"
 
@@ -91,6 +92,9 @@ RTC_OBJC_EXPORT
 /* Stop an active AecDump recording */
 - (void)stopAecDump;
 
+- (void)initializeAudioTesting;
+- (void)deliverRecordedDataWithNumFrames:(uint32_t)numFrames
+                            ioData:(AudioBufferList*)ioData;
 @end
 
-NS_ASSUME_NONNULL_END
+NS_ASSUME_NONNULL_END

sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm

@@ -66,6 +66,21 @@ @implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
 }
 
 @synthesize nativeFactory = _nativeFactory;
+@synthesize internalAudioDeviceModule = _audioDeviceModule;
+
+- (void)initializeAudioTesting {
+  #if defined(WEBRTC_IOS)
+  auto adm = _audioDeviceModule.get();
+  static_cast<webrtc::ios_adm::AudioDeviceModuleIOS *>(adm)->InitializeAudioTesting();
+  #endif
+}
+- (void)deliverRecordedDataWithNumFrames:(uint32_t)numFrames
+                            ioData:(AudioBufferList*)ioData {
+  #if defined(WEBRTC_IOS)
+  auto adm = _audioDeviceModule.get();
+  static_cast<webrtc::ios_adm::AudioDeviceModuleIOS *>(adm)->DeliverRecordedData(numFrames, ioData);
+  #endif
+}
 
 - (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
 #if defined(WEBRTC_IOS)
@@ -175,6 +190,7 @@ - (instancetype)initWithNoMedia {
     dependencies.worker_thread = _workerThread.get();
     dependencies.signaling_thread = _signalingThread.get();
     _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+    _audioDeviceModule = nil;
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
   }
   return self;
@@ -265,6 +281,7 @@ - (instancetype)initWithNativeAudioEncoderFactory:
     dependencies.media_transport_factory = std::move(mediaTransportFactory);
 #endif
     _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+    _audioDeviceModule = audioDeviceModule;
     NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
   }
   return self;

sdk/objc/native/src/audio/audio_device_ios.h

@@ -166,6 +166,10 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
 
   bool IsInterrupted();
 
+  void InitializeAudioTesting();
+
+  void DeliverRecordedData(UInt32 num_frames,
+                           AudioBufferList* io_data) const;
  private:
   // Called by the relevant AudioSessionObserver methods on |thread_|.
   void HandleInterruptionBegin();

sdk/objc/native/src/audio/audio_device_ios.mm

@@ -1138,5 +1138,15 @@ static void LogDeviceInfo() {
   audioSink_ = audioSink;
 }
 
+void AudioDeviceIOS::InitializeAudioTesting() {
+  audio_unit_->InitializeAudioTesting();
+}
+
+void AudioDeviceIOS::DeliverRecordedData(UInt32 num_frames,
+                                            AudioBufferList* io_data) const {
+    rtc::ArrayView<int16_t> recorded_audio(reinterpret_cast<int16_t*>(io_data->mBuffers[0].mData), num_frames);
+    fine_audio_buffer_->DeliverRecordedData(recorded_audio, kFixedRecordDelayEstimate);
+}
+
 }  // namespace ios_adm
 }  // namespace webrtc

sdk/objc/native/src/audio/audio_device_module_ios.h

@@ -128,6 +128,9 @@ class AudioDeviceModuleIOS : public AudioDeviceModule {
   int32_t EnableBuiltInNS(bool enable) override;
 
   int32_t GetPlayoutUnderrunCount() const override;
+  void InitializeAudioTesting() const;
+  void DeliverRecordedData(UInt32 num_frames,
+                           AudioBufferList* io_data) const;
 #if defined(WEBRTC_IOS)
   int GetPlayoutAudioParameters(AudioParameters* params) const override;
   int GetRecordAudioParameters(AudioParameters* params) const override;

sdk/objc/native/src/audio/audio_device_module_ios.mm

@@ -659,6 +659,14 @@
     return ok;
   }
 
+  void AudioDeviceModuleIOS::InitializeAudioTesting() const {
+    audio_device_->InitializeAudioTesting();
+  }
+
+  void AudioDeviceModuleIOS::DeliverRecordedData(UInt32 num_frames,
+                                                AudioBufferList* io_data) const {
+    audio_device_->DeliverRecordedData(num_frames, io_data);
+  }
 #if defined(WEBRTC_IOS)
   int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
       AudioParameters* params) const {

sdk/objc/native/src/audio/voice_processing_audio_unit.h

@@ -92,6 +92,10 @@ class VoiceProcessingAudioUnit {
                   UInt32 num_frames,
                   AudioBufferList* io_data);
 
+  void InitializeAudioTesting();
+
+  static bool vpio_enabled;
+
  private:
   // The C API used to set callbacks requires static functions. When these are
   // called, they will invoke the relevant instance method by casting

sdk/objc/native/src/audio/voice_processing_audio_unit.mm

@@ -82,6 +82,7 @@ static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
   DisposeAudioUnit();
 }
 
+bool VoiceProcessingAudioUnit::vpio_enabled = true;
 const UInt32 VoiceProcessingAudioUnit::kBytesPerSample = 2;
 
 bool VoiceProcessingAudioUnit::Init() {
@@ -395,8 +396,13 @@ static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
     AudioBufferList* io_data) {
   VoiceProcessingAudioUnit* audio_unit =
       static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
-  return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
+
+  if (VoiceProcessingAudioUnit::vpio_enabled) {
+    return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
                                                num_frames, io_data);
+  } else { 
+    return noErr;
+  }
 }
 
 OSStatus VoiceProcessingAudioUnit::NotifyGetPlayoutData(
@@ -466,5 +472,10 @@ static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
   }
 }
 
+void VoiceProcessingAudioUnit::InitializeAudioTesting() {
+  VoiceProcessingAudioUnit::vpio_enabled = false;
+  RTCLog(@"Initializing Audio Testing...Removing Audio Unit Input Callback.");
+}
+
 }  // namespace ios_adm
 }  // namespace webrtc

tools_webrtc/ios/build_ios_libs.py

@@ -59,7 +59,7 @@ def _ParseArgs():
            'If specified together with -c, deletes the dir.')
   parser.add_argument('-r', '--revision', type=int, default=0,
       help='Specifies a revision number to embed if building the framework.')
-  parser.add_argument('-e', '--bitcode', action='store_true', default=True,
+  parser.add_argument('-e', '--bitcode', action='store_true', default=False,
       help='Compile with bitcode.')
   parser.add_argument('--verbose', action='store_true', default=False,
       help='Debug logging.')