OSStatus error -50 (paramErr) on AudioUnitRender call on device - ios

Question

I'm writing an iOS app that captures audio from the microphone, filters it with a high-pass filter, and plays it back through the speakers.
I'm getting a -50 OSStatus error when I call AudioUnitRender on the render callback function when I run it on an iPhone 4S, but it runs fine on the simulator.
I'm using an AUGraph, which has a RemoteIO unit, a HighPassFilter effect unit, and an AUConverter unit to make the ASBDs between the HPF and the output match. The converter AudioUnit instance is called converterUnit.
Here's the code.
static OSStatus renderInput(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
{
AudioController *THIS = (AudioController*)inRefCon;
AudioBuffer buffer;
AudioStreamBasicDescription converterOutputASBD;
UInt32 converterOutputASBDSize = sizeof(converterOutputASBD);
AudioUnitGetProperty([THIS converterUnit], kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &converterOutputASBD, &converterOutputASBDSize);
buffer.mDataByteSize = inNumberFrames * converterOutputASBD.mBytesPerFrame;
buffer.mNumberChannels = converterOutputASBD.mChannelsPerFrame;
buffer.mData = malloc(buffer.mDataByteSize);
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = buffer;
OSStatus result = AudioUnitRender([THIS converterUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList);
...
}
I think -50 error means one of the parameters is wrong. The only parameters that can be wrong are [THIS converterUnit] and &bufferList, given that all the rest are handed to me as arguments. I've checked the converterUnit instance and it is correctly allocated and initialized (what's more, if that was the problem, it wouldn't run on the simulator either). The only parameter left to check is the bufferList. What I could make out so far from debugging is that both the RemoteIO's output element's input ASBD, and the inNumberFrames are different in the phone and on the simulator. But still, I think that to me that doesn't change things, given that I create and allocate memory for the AudioBuffer buffer based on an ASBD resulting from a AudioUnitGetProperty([THIS ioUnit], kAudioUnitProperty_StreamFormat, ...) call.
Any help will be much appreciated, I'm kind of running desperate here..
You guys rock.
Cheers.
UPDATE:
Here's the audio controller class' definition:
#interface AudioController : NSObject
{
AUGraph mGraph;
AudioUnit mEffects;
AudioUnit ioUnit;
AudioUnit converterUnit;
}
#property (readonly, nonatomic) AudioUnit mEffects;
#property (readonly, nonatomic) AudioUnit ioUnit;
#property (readonly, nonatomic) AudioUnit converterUnit;
#property (nonatomic) float* volumenPromedio;
-(void)initializeAUGraph;
-(void)startAUGraph;
-(void)stopAUGraph;
#end
, and here's the initialization code for the AUGraph (defined in AudioController.mm):
- (void)initializeAUGraph
{
NSError *audioSessionError = nil;
AVAudioSession *mySession = [AVAudioSession sharedInstance];
[mySession setPreferredHardwareSampleRate: kGraphSampleRate
error: &audioSessionError];
[mySession setCategory: AVAudioSessionCategoryPlayAndRecord
error: &audioSessionError];
[mySession setActive: YES error: &audioSessionError];
OSStatus result = noErr;
// create a new AUGraph
result = NewAUGraph(&mGraph);
AUNode outputNode;
AUNode effectsNode;
AUNode converterNode;
// effects component
AudioComponentDescription effects_desc;
effects_desc.componentType = kAudioUnitType_Effect;
effects_desc.componentSubType = kAudioUnitSubType_HighPassFilter;
effects_desc.componentFlags = 0;
effects_desc.componentFlagsMask = 0;
effects_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// output component
AudioComponentDescription output_desc;
output_desc.componentType = kAudioUnitType_Output;
output_desc.componentSubType = kAudioUnitSubType_RemoteIO;
output_desc.componentFlags = 0;
output_desc.componentFlagsMask = 0;
output_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// stream format converter component
AudioComponentDescription converter_desc;
converter_desc.componentType = kAudioUnitType_FormatConverter;
converter_desc.componentSubType = kAudioUnitSubType_AUConverter;
converter_desc.componentFlags = 0;
converter_desc.componentFlagsMask = 0;
converter_desc.componentManufacturer = kAudioUnitManufacturer_Apple;
// Add nodes to the graph
result = AUGraphAddNode(mGraph, &output_desc, &outputNode);
[self hasError:result:__FILE__:__LINE__];
result = AUGraphAddNode(mGraph, &effects_desc, &effectsNode);
[self hasError:result:__FILE__:__LINE__];
result = AUGraphAddNode(mGraph, &converter_desc, &converterNode);
// manage connections in the graph
// Connect the io unit node's input element's output to the effectsNode input
result = AUGraphConnectNodeInput(mGraph, outputNode, 1, effectsNode, 0);
// Connect the effects node's output to the converter node's input
result = AUGraphConnectNodeInput(mGraph, effectsNode, 0, converterNode, 0);
// open the graph
result = AUGraphOpen(mGraph);
// Get references to the audio units
result = AUGraphNodeInfo(mGraph, effectsNode, NULL, &mEffects);
result = AUGraphNodeInfo(mGraph, outputNode, NULL, &ioUnit);
result = AUGraphNodeInfo(mGraph, converterNode, NULL, &converterUnit);
// Enable input on remote io unit
UInt32 flag = 1;
result = AudioUnitSetProperty(ioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &flag, sizeof(flag));
// Setup render callback struct
AURenderCallbackStruct renderCallbackStruct;
renderCallbackStruct.inputProc = &renderInput;
renderCallbackStruct.inputProcRefCon = self;
result = AUGraphSetNodeInputCallback(mGraph, outputNode, 0, &renderCallbackStruct);
// Get fx unit's input current stream format...
AudioStreamBasicDescription fxInputASBD;
UInt32 sizeOfASBD = sizeof(AudioStreamBasicDescription);
result = AudioUnitGetProperty(mEffects, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &fxInputASBD, &sizeOfASBD);
// ...and set it on the io unit's input scope's output
result = AudioUnitSetProperty(ioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
1,
&fxInputASBD,
sizeof(fxInputASBD));
// Set fx unit's output sample rate, just in case
Float64 sampleRate = 44100.0;
result = AudioUnitSetProperty(mEffects,
kAudioUnitProperty_SampleRate,
kAudioUnitScope_Output,
0,
&sampleRate,
sizeof(sampleRate));
AudioStreamBasicDescription fxOutputASBD;
// get fx audio unit's output ASBD...
result = AudioUnitGetProperty(mEffects, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &fxOutputASBD, &sizeOfASBD);
// ...and set it to the converter audio unit's input
result = AudioUnitSetProperty(converterUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &fxOutputASBD, sizeof(fxOutputASBD));
AudioStreamBasicDescription ioUnitsOutputElementInputASBD;
// now get io audio unit's output element's input ASBD...
result = AudioUnitGetProperty(ioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &ioUnitsOutputElementInputASBD, &sizeOfASBD);
// ...set the sample rate...
ioUnitsOutputElementInputASBD.mSampleRate = 44100.0;
// ...and set it to the converter audio unit's output
result = AudioUnitSetProperty(converterUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &ioUnitsOutputElementInputASBD, sizeof(ioUnitsOutputElementInputASBD));
// initialize graph
result = AUGraphInitialize(mGraph);
}
The reason I make the connection between the converter's output and the remote io unit's output element's input with a render callback function (rather than with the AUGraphConnectNodeInput method) is because I need to make some calculations on the samples right after they've been processed by the high-pass filter. The render callback gives me the opportunity to look into the samples buffer right after the AudioUnitRender call, and do said calculations there.
UPDATE 2:
By debugging, I found differences in the Remote IO output bus' input ASBD on the device and on the simulator. It shouldn't make a difference (I allocate and initialize the AudioBufferList based on data coming from a previous AudioUnitGetProperty([THIS ioUnit], kAudioUnitProperty_StreamFormat, ...) call), but it's the only thing I can see different in the device and the simulator.
Here's the Remote IO output bus' input ASBD on the device:
Float64 mSampleRate 44100
UInt32 mFormatID 1819304813
UInt32 mFormatFlags 41
UInt32 mBytesPerPacket 4
UInt32 mFramesPerPacket 1
UInt32 mBytesPerFrame 4
UInt32 mChannelsPerFrame 2
UInt32 mBitsPerChannel 32
UInt32 mReserved 0
, and here it is on the simulator:
Float64 mSampleRate 44100
UInt32 mFormatID 1819304813
UInt32 mFormatFlags 12
UInt32 mBytesPerPacket 4
UInt32 mFramesPerPacket 1
UInt32 mBytesPerFrame 4
UInt32 mChannelsPerFrame 2
UInt32 mBitsPerChannel 16
UInt32 mReserved 0