I’ve had nasty old time trying to get some audio stuff going on the iPhone, no thanks to Apple’s lack of documentation. I have a feeling it’s all still quite hush-hush, so no details here, but if you’re an iPhone developer interested in getting RemoteIO/IO Remote/whatever it’s called working on the iPhone… Do I have good news for you.
Read on here, at the Developer forums
Drop me a line if you find this helpful.
Update: I’m told the new NDA is pretty much all-good with blog postings. So, read on for the goods.
Update 2: Thanks to Joel Reymont, we now have an explanation for the “CrashIfClientProvidedBogusAudioBufferList” iPhone simulator bug: The simulator doesn’t like mono audio. Thanks, Joel!
Update 3: Happily, Apple have now created some excellent documentation on Remote IO, with some good sample projects. I recommend using that as a resource, now that it’s there, as that will continue to be updated.
So, we need to obtain an instance of the RemoteIO audio unit, configure it, and hook it up to a recording callback, which is used to notify you that there is data ready to be grabbed, and where you pull the data from the audio unit.
Overview
- Identify the audio component (kAudioUnitType_Output/ kAudioUnitSubType_RemoteIO/ kAudioUnitManufacturerApple)
- Use AudioComponentFindNext(NULL, &descriptionOfAudioComponent) to obtain the AudioComponent, which is like the factory with which you obtain the audio unit
- Use AudioComponentInstanceNew(ourComponent, &audioUnit) to make an instance of the audio unit
- Enable IO for recording and possibly playback with AudioUnitSetProperty
- Describe the audio format in an AudioStreamBasicDescription structure, and apply the format using AudioUnitSetProperty
- Provide a callback for recording, and possibly playback, again using AudioUnitSetProperty
- Allocate some buffers
- Initialise the audio unit
- Start the audio unit
- Rejoice
Here’s my code: I’m using both recording and playback. Use what applies to you!
Initialisation
Initialisation looks like this. We have a member variable of type AudioComponentInstance which will contain our audio unit.
The audio format described below uses SInt16 for samples (i.e. signed, 16 bits per sample)
#define kOutputBus 0
#define kInputBus 1
// ...
OSStatus status;
AudioComponentInstance audioUnit;
// Describe audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_RemoteIO;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Get component
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// Get audio units
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
checkStatus(status);
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
kInputBus,
&flag,
sizeof(flag));
checkStatus(status);
// Enable IO for playback
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
kOutputBus,
&flag,
sizeof(flag));
checkStatus(status);
// Describe format
audioFormat.mSampleRate = 44100.00;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
// Apply format
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
checkStatus(status);
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
kOutputBus,
&audioFormat,
sizeof(audioFormat));
checkStatus(status);
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
kInputBus,
&callbackStruct,
sizeof(callbackStruct));
checkStatus(status);
// Set output callback
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = self;
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
checkStatus(status);
// 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_Output,
kInputBus,
&flag,
sizeof(flag));
// TODO: Allocate our own buffers if we want
// Initialise
status = AudioUnitInitialize(audioUnit);
checkStatus(status);
Then, when you’re ready to start:
OSStatus status = AudioOutputUnitStart(audioUnit);
checkStatus(status);
And to stop:
OSStatus status = AudioOutputUnitStop(audioUnit);
checkStatus(status);
Then, when we’re finished:
AudioUnitUninitialize(audioUnit);
And now for our callbacks.
RECORDING
static OSStatus 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.
AudioBufferList *bufferList; // <- Fill this up with buffers (you will want to malloc it, as it's a dynamic-length list)
// Then:
// Obtain recorded samples
OSStatus status;
status = AudioUnitRender([audioInterface audioUnit],
ioActionFlags,
inTimeStamp,
inBusNumber,
inNumberFrames,
bufferList);
checkStatus(status);
// Now, we have the samples we just read sitting in buffers in bufferList
DoStuffWithTheRecordedAudio(bufferList);
return noErr;
}
PLAYBACK
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
// Notes: ioData contains buffers (may be more than one!)
// Fill them up as much as you can. Remember to set the size value in each buffer to match how
// much data is in the buffer.
return noErr;
}
Finally, rejoice with me in this discovery ;)
Resources that helped
- http://pastie.org/pastes/219616
- http://developer.apple.com/samplecode/CAPlayThrough/listing8.html
- http://listas.apesol.org/pipermail/svn-libsdl.org/2008-July/000797.html
No thanks at all to Apple for their lack of accessible documentation on this topic – They really have a long way to go here! Also boo to them with their lack of search engine, and refusal to open up their docs to Google. It’s a jungle out there!
Update: You can adjust the latency of RemoteIO (and, in fact, any other audio framework) by setting the kAudioSessionProperty_PreferredHardwareIOBufferDuration
property:
float aBufferLength = 0.005; // In seconds
AudioSessionSetProperty(kAudioSessionProperty_PreferredHardwareIOBufferDuration,
sizeof(aBufferLength), &aBufferLength);
This adjusts the length of buffers that’re passed to you – if buffer length was originally, say, 1024 samples, then halving the number of samples halves the amount of time taken to process them.