consuming audio data from circular buffer in a render callback attached to the input scope of a remoteio audio unit

Question

The title pretty much sums up what I'm trying to achieve. I am trying to use Michael Tyson's TPCircularBuffer inside of a render callback while the circular buffer is getting filled with incoming audio data. I want to send the audio from the render callback to the output element of the RemoteIO audio unit so I can hear it through the device speakers.

The audio is interleaved stereo 16 bit coming in as packets of 2048 frames. Here's how I've set up my audio session:

#define kInputBus 1
#define kOutputBus 0
NSError *err = nil;
NSTimeInterval ioBufferDuration = 46;
AVAudioSession *session = [AVAudioSession sharedInstance];
[session setCategory:AVAudioSessionCategoryPlayback withOptions:AVAudioSessionCategoryOptionMixWithOthers error:&err];
[session setPreferredIOBufferDuration:ioBufferDuration error:&err];
[session setActive:YES error:&err];
AudioComponentDescription defaultOutputDescription;
defaultOutputDescription.componentType = kAudioUnitType_Output;
defaultOutputDescription.componentSubType = kAudioUnitSubType_RemoteIO;
defaultOutputDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
defaultOutputDescription.componentFlags = 0;
defaultOutputDescription.componentFlagsMask = 0;

AudioComponent defaultOutput = AudioComponentFindNext(NULL, &defaultOutputDescription);
NSAssert(defaultOutput, @"Can't find default output.");

AudioComponentInstanceNew(defaultOutput, &remoteIOUnit);
UInt32 flag = 0;

OSStatus status = AudioUnitSetProperty(remoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, kOutputBus, &flag, sizeof(flag));
size_t bytesPerSample = sizeof(AudioUnitSampleType);
AudioStreamBasicDescription streamFormat = {0};
streamFormat.mSampleRate = 44100.00;
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags = kAudioFormatFlagsCanonical;
streamFormat.mBytesPerPacket = bytesPerSample;
streamFormat.mFramesPerPacket = 1;
streamFormat.mBytesPerFrame = bytesPerSample;
streamFormat.mChannelsPerFrame = 2;
streamFormat.mBitsPerChannel = bytesPerSample * 8;
streamFormat.mReserved = 0;

status = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &streamFormat, sizeof(streamFormat));

AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = render;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, kOutputBus, &callbackStruct, sizeof(callbackStruct));

And here's where the audio data gets loaded into the circular buffer and used in the render callback:

#define kBufferLength 2048
-(void)loadBytes:(Byte *)byteArrPtr{
TPCircularBufferProduceBytes(&buffer, byteArrPtr, kBufferLength);
}

OSStatus render(
                void *inRefCon,
                AudioUnitRenderActionFlags *ioActionFlags,
                const AudioTimeStamp *inTimeStamp,
                UInt32 inBusNumber,
                UInt32 inNumberFrames,
                AudioBufferList *ioData)
{
AUDIOIO *audio = (__bridge AUDIOIO *)inRefCon;
AudioSampleType *outSample = (AudioSampleType *)ioData->mBuffers[0].mData;
//Zero outSample
memset(outSample, 0, kBufferLength);
int bytesToCopy = ioData->mBuffers[0].mDataByteSize;
SInt16 *targetBuffer = (SInt16 *)ioData->mBuffers[0].mData;
//Pull audio
int32_t availableBytes;
SInt16 *buffer = TPCircularBufferTail(&audio->buffer, &availableBytes);
memcpy(targetBuffer, buffer, MIN(bytesToCopy, availableBytes));
TPCircularBufferConsume(&audio->buffer, MIN(bytesToCopy, availableBytes));
return noErr;
}

There is something wrong with this setup because I am not getting any audio through the speakers, but I'm also not getting any errors when I test on my device. As far as I can tell the TPCircularBuffer is being filled and read from correctly. I've followed the Apple documentation for setting up the audio session. I am considering trying to set up an AUGraph next but I want to see if anyone could suggest a solution for what I'm trying to do here. Thanks!

Answer 1

For stereo (2 channels per frame), your bytes per frame and bytes per packet have to be twice your sample size in bytes. Same with bits per channel in terms of bits.

Added: If availableBytes/yourFrameSize isn't almost always as large or larger than inNumberFrames, you won't get much continuous sound.

Answer 2

At a glance, it looks like you've got everything set up correctly. You're missing a call to AudioOutputUnitStart() though:

...
// returns an OSStatus indicating success / fail
AudioOutputUnitStart(remoteIOUnit);

// now your callback should be being called
...

Answer 3

I believe one your problem is with using streamFormat.mBitsPerChannel = bytesPerSample * 8;

You assign bytesPerSample to be sizeof(AudioUnitSampleType) which is essentially 4 bytes.

So streamFormat.mBytesPerPacket = bytesPerSample; is ok. But the assignment streamFormat.mBitsPerChannel = bytesPerSample * 8; is saying that you want 32 bits per sample instead of 16 bits per sample.

I would not create your audio format based on AudioUnitSampleType because this has nothing to do with your personal format that you want to utilize. I would create defines and do something like this:

#define BITS_PER_CHANNEL 16
#define SAMPLE_RATE 44100.0
#define CHANNELS_PER_FRAME 2
#define BYTES_PER_FRAME CHANNELS_PER_FRAME * (BITS_PER_CHANNEL / 8)  //ie 4
#define FRAMES_PER_PACKET 1
#define BYTES_PER_PACKET FRAMES_PER_PACKET * BYTES_PER_FRAME




    streamFormat.mSampleRate = SAMPLE_RATE;  // 44100.0
    streamFormat.mBitsPerChannel = BITS_PER_CHANNEL; //16
    streamFormat.mChannelsPerFrame = CHANNELS_PER_FRAME; // 2
    streamFormat.mFramesPerPacket = FRAMES_PER_PACKET; //1
    streamFormat.mBytesPerFrame = BYTES_PER_FRAME; // 4 total,  2 for left ch,  2 for right ch

    streamFormat.mBytesPerPacket = BYTES_PER_PACKET;

    streamFormat.mReserved = 0;
    streamFormat.mFormatID = kAudioFormatLinearPCM;  // double check this also
    streamFormat.mFormatFlags = kAudioFormatFlagsCanonical;`

You also need to look at the return values set to err and status immediately after each are run. You still need to add error checking at some of the calls as well such as

checkMyReturnValueToo = AudioComponentInstanceNew(defaultOutput, &remoteIOUnit);

You also have an extremely high value for your buffer duration. You have 46 and I am not sure where that came from. That means you want 46 seconds worth of audio during each audio callback. Usually you want something less than one second depending on your latency requirements. Most likely iOS will not use anything that high but you should try setting it to say 0.025 or so (25ms). You can try to lower it if you need faster latency.