Recording and processing audio on iOS

initAudioSession.c

// Yeah, global variables suck, but it's kind of a necessary evil here
AudioUnit *audioUnit = NULL;
float *convertedSampleBuffer = NULL;

int initAudioSession() {
  audioUnit = (AudioUnit*)malloc(sizeof(AudioUnit));

  if(AudioSessionInitialize(NULL, NULL, NULL, NULL) != noErr) {
    return 1;
  }

  if(AudioSessionSetActive(true) != noErr) {
    return 1;
  }

  UInt32 sessionCategory = kAudioSessionCategory_PlayAndRecord;
  if(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(UInt32), &sessionCategory) != noErr) {
    return 1;
  }

  Float32 bufferSizeInSec = 0.02f;
  if(AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof(Float32), &bufferSizeInSec) != noErr) {
    return 1;
  }

  UInt32 overrideCategory = 1;
  if(AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof(UInt32), &overrideCategory) != noErr) {
    return 1;
  }

  // There are many properties which you might want to provide callback functions for, including:
  // kAudioSessionProperty_AudioRouteChange
  // kAudioSessionProperty_OverrideCategoryEnableBluetoothInput
  // etc.

  return 0;
}

initAudioStreams.c

int initAudioStreams(AudioUnit *audioUnit) {
  UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
  if(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(UInt32), &audioCategory) != noErr) {
    return 1;
  }
  
  UInt32 overrideCategory = 1;
  if(AudioSessionSetProperty(kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof(UInt32), &overrideCategory) != noErr) {
    // Less serious error, but you may want to handle it and bail here
  }
  
  AudioComponentDescription componentDescription;
  componentDescription.componentType = kAudioUnitType_Output;
  componentDescription.componentSubType = kAudioUnitSubType_RemoteIO;
  componentDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
  componentDescription.componentFlags = 0;
  componentDescription.componentFlagsMask = 0;
  AudioComponent component = AudioComponentFindNext(NULL, &componentDescription);
  if(AudioComponentInstanceNew(component, audioUnit) != noErr) {
    return 1;
  }

  UInt32 enable = 1;
  if(AudioUnitSetProperty(*audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enable, sizeof(UInt32)) != noErr) {
    return 1;
  }

  AURenderCallbackStruct callbackStruct;
  callbackStruct.inputProc = renderCallback; // Render function
  callbackStruct.inputProcRefCon = NULL;
  if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &callbackStruct, sizeof(AURenderCallbackStruct)) != noErr) {
    return 1;
  }

  AudioStreamBasicDescription streamDescription;
  // You might want to replace this with a different value, but keep in mind that the
  // iPhone does not support all sample rates. 8kHz, 22kHz, and 44.1kHz should all work.
  streamDescription.mSampleRate = 44100;
  // Yes, I know you probably want floating point samples, but the iPhone isn't going
  // to give you floating point data. You'll need to make the conversion by hand from
  // linear PCM <-> float.
  streamDescription.mFormatID = kAudioFormatLinearPCM;
  // This part is important!
  streamDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked;
  // Not sure if the iPhone supports recording >16-bit audio, but I doubt it.
  streamDescription.mBitsPerChannel = 16;
  // 1 sample per frame, will always be 2 as long as 16-bit samples are being used
  streamDescription.mBytesPerFrame = 2;
  // Record in mono. Use 2 for stereo, though I don't think the iPhone does true stereo recording
  streamDescription.mChannelsPerFrame = 1;
  streamDescription.mBytesPerPacket = streamDescription.mBytesPerFrame * streamDescription.mChannelsPerFrame;
  // Always should be set to 1
  streamDescription.mFramesPerPacket = 1;
  // Always set to 0, just to be sure
  streamDescription.mReserved = 0;

  // Set up input stream with above properties
  if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamDescription, sizeof(streamDescription)) != noErr) {
    return 1;
  }

  // Ditto for the output stream, which we will be sending the processed audio to
  if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &streamDescription, sizeof(streamDescription)) != noErr) {
    return 1;
  }

  return 0;
}

renderCallback.c

OSStatus renderCallback(void *userData, AudioUnitRenderActionFlags *actionFlags, const AudioTimeStamp *audioTimeStamp,
 UInt32 busNumber, UInt32 numFrames, AudioBufferList *buffers) {
  OSStatus status = AudioUnitRender(*audioUnit, actionFlags, audioTimeStamp, 1, numFrames, buffers);
  if(status != noErr) {
    return status;
  }

  if(convertedSampleBuffer == NULL) {
    // Lazy initialization of this buffer is necessary because we don't know the frame count until the first callback
    convertedSampleBuffer = (float*)malloc(sizeof(float) * numFrames);
  }

  SInt16 *inputFrames = (SInt16*)(buffers->mBuffers->mData);

  // If your DSP code can use integers, then don't bother converting to floats here, as it just wastes CPU. However,
  // most DSP algorithms rely on floating point, and this is especially true if you are porting a VST/AU to iOS.
  for(int i = 0; i < numFrames; i++) {
    convertedSampleBuffer[i] = (float)inputFrames[i] / 32768f;
  }

  // Now we have floating point sample data from the render callback! We can send it along for further processing
  // plugin->processReplacing(convertedSampleBuffer, NULL, sampleFrames);

  // Assuming that you have processed in place, we can now write the floating point data back to the input buffer
  for(int i = 0; i < numFrames; i++) {
    // Note that we multiply by 32767 here, NOT 32768. This is to avoid overflow (and thus clipping).
    inputFrames[i] = (SInt16)(convertedSampleBuffer[i] * 32767f);
  }

  return noErr;
}

startAudioUnit.c

int startAudioUnit(AudioUnit *audioUnit) {
  if(AudioUnitInitialize(*audioUnit) != noErr) {
    return 1;
  }

  if(AudioOutputUnitStart(*audioUnit) != noErr) {
    return 1;
  }

  return 0;
}

stopProcessingAudio.c

int stopProcessingAudio(AudioUnit *audioUnit) {
  if(AudioOutputUnitStop(*audioUnit) != noErr) {
    return 1;
  }

  if(AudioUnitUninitialize(*audioUnit) != noErr) {
    return 1;
  }

  *audioUnit = NULL;
  return 0;
}

How to use this for internal audio??