AudioConverterNew returned -50 - c++

Question

I have a little issue regarding the use of the AudioQueue services.
I have followed the guide that is available on Apple's webiste, but when I got to start and run the Audio Queue, I get the message telling me that "AudioConverterNew returned -50".
Now, I know that the -50 error code means that there is a bad parameter. However, what I don't know is which parameter is the bad one (thank you so much Apple...) !
So, here's my code.
Here are the parameters of my class, named cPlayerCocoa
AudioQueueRef mQueue;
AudioQueueBufferRef mBuffers[NUMBER_BUFFERS]; // NUMBER_BUFFERS = 3
uint32 mBufferByteSize;
AudioStreamBasicDescription mDataFormat;
Here's the first function :
static void
BuildBuffer( void* iAQData, AudioQueueRef iAQ, AudioQueueBufferRef iBuffer )
{
cPlayerCocoa* player = (cPlayerCocoa*) iAQData;
player->HandleOutputBuffer( iAQ, iBuffer );
}
It creates a cPlayerCocoa from the structure containing the AudioQueue and calls the HandleOutputBuffer function, which allocates the audio buffers :
void
cPlayerCocoa::HandleOutputBuffer( AudioQueueRef iAQ, AudioQueueBufferRef iBuffer )
{
if( mContinue )
{
xassert( iBuffer->mAudioDataByteSize == 32768 );
int startSample = mPlaySampleCurrent;
int result = 0;
int samplecount = 32768 / ( mSoundData->BytesPerSample() ); // BytesPerSample, in my case, returns 4
tErrorCode error = mSoundData->ReadData( (int16*)(iBuffer->mAudioData), samplecount, &result, startSample );
AudioQueueEnqueueBuffer( mQueue, iBuffer, 0, 0 ); // I'm using CBR data (PCM), hence the 0 passed into the AudioQueueEnqueueBuffer.
if( result != samplecount )
mContinue = false;
startSample += result;
}
else
{
AudioQueueStop( mQueue, false );
}
}
In this next function, the AudioQueue is created then started.
I begin to initialise the parameters of the Data format. Then I create the AudioQueue, and I allocate the 3 buffers.
When the buffers are allocated, I start the AudioQueue and then I run the loop.
void
cPlayerCocoa::ThreadEntry()
{
int samplecount = 32768 / ( mSoundData->BytesPerSample() );
mDataFormat.mSampleRate = mSoundData->SamplingRate(); // Returns 44100
mDataFormat.mFormatID = kAudioFormatLinearPCM;
mDataFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
mDataFormat.mBytesPerPacket = 32768;
mDataFormat.mFramesPerPacket = samplecount;
mDataFormat.mBytesPerFrame = mSoundData->BytesPerSample(); // BytesPerSample returns 4.
mDataFormat.mChannelsPerFrame = 2;
mDataFormat.mBitsPerChannel = uint32(mSoundData->BitsPerChannel());
mDataFormat.mReserved = 0;
AudioQueueNewOutput( &mDataFormat, BuildBuffer, this, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &mQueue );
for( int i = 0; i < NUMBER_BUFFERS; ++i )
{
AudioQueueAllocateBuffer( mQueue, mBufferByteSize, &mBuffers[i] );
HandleOutputBuffer( mQueue, mBuffers[i] );
}
AudioQueueStart( mQueue, NULL ); // I want the queue to start playing immediately, so I pass NULL
do {
CFRunLoopRunInMode( kCFRunLoopDefaultMode, 0.25, false );
} while ( !NeedStopASAP() );
AudioQueueDispose( mQueue, true );
}
The call to AudioQueueStart returns -50 (bad parameter) and I can't figure what's wrong...
I would really appreciate some help, thanks in advance :-)

Answer 1

I think your ASBD is suspect. PCM formats have predictable values for mBytesPerPacket, mBytesPerFrame, and mFramesPerPacket. For normal 16-bit interleaved signed 44.1 stereo audio the ASBD would look like
AudioStreamBasicDescription asbd = {
.mFormatID = kAudioFormatLinearPCM,
.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked,
.mSampleRate = 44100,
.mChannelsPerFrame = 2,
.mBitsPerChannel = 16,
.mBytesPerPacket = 4,
.mFramesPerPacket = 1,
.mBytesPerFrame = 4,
.mReserved = 0
};
AudioConverterNew returns -50 when one of the ASBDs is unsupported. There is no PCM format where mBytesPerPacket should be 32768, which is why you're getting the error.