Realtime audio processing without output

core-audio, ios, objective-c

Solution

If you want only recording, no playback, simply comment out the line that sets renderCallback:

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

Update after seeing code:

As I suspected, you're missing input callback. Add these lines:

// at top:
#define kInputBus 1

AURenderCallbackStruct callbackStruct;
/**/
callbackStruct.inputProc = &ALAudioUnit::recordingCallback;
callbackStruct.inputProcRefCon = this;
status = AudioUnitSetProperty(audioUnit,
                              kAudioOutputUnitProperty_SetInputCallback,
                              kAudioUnitScope_Global,
                              kInputBus,
                              &callbackStruct,
                              sizeof(callbackStruct));

Now in your recordingCallback:

OSStatus ALAudioUnit::recordingCallback(void *inRefCon,
                                        AudioUnitRenderActionFlags *ioActionFlags,
                                        const AudioTimeStamp *inTimeStamp,
                                        UInt32 inBusNumber,
                                        UInt32 inNumberFrames,
                                        AudioBufferList *ioData)
{
    // TODO: Use inRefCon to access our interface object to do stuff
    // Then, use inNumberFrames to figure out how much data is available, and make
    // that much space available in buffers in an AudioBufferList.

    // Then:
    // Obtain recorded samples

    OSStatus status;

    ALAudioUnit *pThis = reinterpret_cast<ALAudioUnit*>(inRefCon);
    if (!pThis)
        return noErr;

    //assert (pThis->m_nMaxSliceFrames >= inNumberFrames);

    pThis->recorderBufferList->GetBufferList().mBuffers[0].mDataByteSize = inNumberFrames * pThis->m_recorderSBD.mBytesPerFrame;

    status = AudioUnitRender(pThis->audioUnit,
                             ioActionFlags,
                             inTimeStamp,
                             inBusNumber,
                             inNumberFrames,
                             &pThis->recorderBufferList->GetBufferList());
    THROW_EXCEPTION_IF_ERROR(status, "error rendering audio unit");

    // If we're not playing, I don't care about the data, simply discard it
    if (!pThis->playbackState || pThis->isSeeking) return noErr;

    // Now, we have the samples we just read sitting in buffers in bufferList
    pThis->DoStuffWithTheRecordedAudio(inNumberFrames, pThis->recorderBufferList, inTimeStamp);

    return noErr;
}

Btw, I'm allocating my own buffer instead of using the one provided by AudioUnit. You might want to change those parts if you want to use AudioUnit allocated buffer.

Update:

How to allocate own buffer:

recorderBufferList = new AUBufferList();
recorderBufferList->Allocate(m_recorderSBD, m_nMaxSliceFrames);
recorderBufferList->PrepareBuffer(m_recorderSBD, m_nMaxSliceFrames);

Also, if you're doing this, tell AudioUnit to not allocate buffers:

// Disable buffer allocation for the recorder (optional - do this if we want to pass in our own)
flag = 0;
status = AudioUnitSetProperty(audioUnit,
                              kAudioUnitProperty_ShouldAllocateBuffer,
                              kAudioUnitScope_Input,
                              kInputBus,
                              &flag,
                              sizeof(flag));

You'll need to include CoreAudio utility classes

Problem

I'm looking on this example http://teragonaudio.com/article/How-to-do-realtime-recording-with-effect-processing-on-iOS.html and i want to turn off my output. I try to change: `kAudioSessionCategory_PlayAndRecord` to `kAudioSessionCategory_RecordAudio` but this is not working. I also try to get rid off: ``` if(AudioUnitSetProperty(*audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &streamDescription, sizeof(streamDescription)) != noErr) { return 1; } ``` Becouse i want to get sound from microphone but not playing it. But not matter what i do when my sound get to renderCallback method there is a -50 error. When audio is automatically play on output everything works fine... Update with code: ``` using namespace std; 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 you might want to provide callback functions for: // kAudioSessionProperty_AudioRouteChange // kAudioSessionProperty_OverrideCategoryEnableBluetoothInput // etc. return 0; } OSStatus renderCallback(void *userData, AudioUnitRenderActionFlags *actionFlags, const AudioTimeStamp *audioTimeStamp, UInt32 busNumber, UInt32 numFrames, AudioBufferList *buffers) { OSStatus status = AudioUnitRender(*audioUnit, actionFlags, audioTimeStamp, 1, numFrames, buffers); int doOutput = 0; 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); baseTime = (float)QRealTimer::getUptimeInMilliseconds(); } 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. int i; for( i = numFrames; i < fftlength; i++ ) // Shifting buffer x_inbuf[i - numFrames] = x_inbuf[i]; for( i = 0; i < numFrames; i++) { x_inbuf[i + x_phase] = (float)inputFrames[i] / (float)32768; } if( x_phase + numFrames == fftlength ) { x_alignment.SigProc_frontend(x_inbuf); // Signal processing front-end (FFT!) doOutput = x_alignment.Align(); /// Output as text! In the real-time version, // this is where we update visualisation callbacks and launch other services if ((doOutput) & (x_netscore.isEvent(x_alignment.Position())) &(x_alignment.lastAction()<x_alignment.Position()) ) { // here i want to do something with my input! } } else x_phase += numFrames; return noErr; } 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; streamDescription.mBitsPerChannel = 16; // 1 sample per frame, will always be 2 as long as 16-bit samples are being used streamDescription.mBytesPerFrame = 2; 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; } int startAudioUnit(AudioUnit *audioUnit) { if(AudioUnitInitialize(*audioUnit) != noErr) { return 1; } if(AudioOutputUnitStart(*audioUnit) != noErr) { return 1; } return 0; } ``` And calling from my VC: ``` initAudioSession(); initAudioStreams( audioUnit); startAudioUnit( audioUnit); ```

Original source