Realtime audio processing without output - ios

Question

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);

Answer 1

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

Answer 2

Thanks for #Mar0ux 's answer. Whoever got here looking for complete sample code doing this can take a look here:
https://code.google.com/p/ios-coreaudio-example/

Answer 3

I am doing a similar app working with the same code and I found that you can end playback by changing the enumeration kAudioSessionCategory_PlayAndRecord to RecordAudio
int initAudioStreams(AudioUnit *audioUnit) {
UInt32 audioCategory = kAudioSessionCategory_RecordAudio;
if(AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(UInt32), &audioCategory) != noErr) {
return 1;
}
This stopped the feedback between mic and speaker on my hardware.