Question

I am using Mat Gallager’s Audio Streamer in order to play an audio stream. At another point in the program (a different controller) I am trying to record something from the device’s microphone.

This is my setup:

void SFIdentificator::startRecord()

{
int i, bufferByteSize;
UInt32 size;

try {
    numberOfPackets = 0;

    // specify the recording format
    SetupAudioFormat(kAudioFormatLinearPCM);

    AudioQueueNewInput(  &mRecordFormat,
                       MyInputBufferHandler,
                       this /* userData */,
                       CFRunLoopGetMain() /* run loop */, kCFRunLoopCommonModes /* run loop mode */,
                       0 /* flags */, &mQueue);
    mRecordPacket = 0;

    size = sizeof(mRecordFormat);
    AudioQueueGetProperty(mQueue, kAudioQueueProperty_StreamDescription, &mRecordFormat, &size);

    bufferByteSize = ComputeRecordBufferSize(&mRecordFormat, kBufferDurationSeconds);   // enough bytes for half a second


    size = sizeof(mRecordFormat);
    XThrowIfError(AudioQueueGetProperty(mQueue, kAudioQueueProperty_StreamDescription,
                                        &mRecordFormat, &size), "couldn't get queue's format");

    for (i = 0; i < kNumberRecordBuffers; ++i) {
        XThrowIfError(AudioQueueAllocateBuffer(mQueue, bufferByteSize, &mBuffers[i]), "AudioQueueAllocateBuffer failed");
        XThrowIfError(AudioQueueEnqueueBuffer(mQueue, mBuffers[i], 0, NULL), "AudioQueueEnqueueBuffer failed");
    }
    mIsRunning = true;

    XThrowIfError(AudioQueueStart(mQueue, NULL), "AudioQueueStart failed");


} catch (CAXException e) {
    char buf[256];
    fprintf(stderr, "Error: %s (%s)\n", e.mOperation, e.FormatError(buf));
}catch (...) {
    fprintf(stderr, "An unknown error occurred\n");;
}

}

void SFIdentificator::SetupAudioFormat(UInt32 inFormatID)

{
memset(&mRecordFormat, 0, sizeof(mRecordFormat));

UInt32 size = sizeof(mRecordFormat.mSampleRate);
XThrowIfError(AudioSessionGetProperty(  kAudioSessionProperty_CurrentHardwareSampleRate, &size, &mRecordFormat.mSampleRate), "couldn't get hardware sample rate");

size = sizeof(mRecordFormat.mChannelsPerFrame);
XThrowIfError(AudioSessionGetProperty(  kAudioSessionProperty_CurrentHardwareInputNumberChannels, &size, &mRecordFormat.mChannelsPerFrame), "couldn't get input channel count");

mRecordFormat.mFormatID = inFormatID;
if (inFormatID == kAudioFormatLinearPCM){
    // if we want pcm, default to signed 16-bit little-endian

    mRecordFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
    //      mRecordFormat.mBitsPerChannel = 16;


    mRecordFormat.mBytesPerPacket = mRecordFormat.mBytesPerFrame = (mRecordFormat.mBitsPerChannel / 8) * mRecordFormat.mChannelsPerFrame;
    mRecordFormat.mFramesPerPacket = 1;

    mRecordFormat.mFormatID         = kAudioFormatLinearPCM;
    mRecordFormat.mSampleRate       = 32000.0;
    mRecordFormat.mChannelsPerFrame = 1;
    mRecordFormat.mBitsPerChannel   = 16;
    mRecordFormat.mBytesPerPacket   =  mRecordFormat.mBytesPerFrame = mRecordFormat.mChannelsPerFrame * sizeof (SInt16);
    mRecordFormat.mFramesPerPacket  = 1;
}

}

UInt32 SFIdentificator::ComputeRecordBufferSize(const AudioStreamBasicDescription *format, float seconds){
static const int maxBufferSize = 0x50000;

int maxPacketSize = format->mBytesPerPacket;
if (maxPacketSize == 0) {
    UInt32 maxVBRPacketSize = sizeof(maxPacketSize);
    AudioQueueGetProperty (mQueue, kAudioQueueProperty_MaximumOutputPacketSize, &maxPacketSize, &maxVBRPacketSize);
}

Float64 numBytesForTime = DataFormat().mSampleRate * maxPacketSize * seconds;
//    *outBufferSize = (UInt32)(numBytesForTime < maxBufferSize ? numBytesForTime : maxBufferSize);
return (UInt32)(numBytesForTime < maxBufferSize ? numBytesForTime : maxBufferSize);

}

It seems that if I use the AudioStreamer class first and try to record something later, not even the Callback is called. But if I don’t first use the AudioStreamer, everything is fine.

Can anyone point me in the right direction?

Answer 1

Answering my own question, it was a problem with setting up correctly the init() of the AudioSession object. it should be done only once per session, and doing it once more will result in an error, which I had identified a inability to record a stream. However, I could record the stream, even if the second init had failed (the first one had succeeded of course).