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I am developing an audio capture application. How to capture audio on Mac OS X using 8 kHz sampling rate with single chanel using Audio Unit API ?

This is the code I have tried.

    Component                   component;
ComponentDescription        description;
OSStatus    err = noErr;
UInt32  param;
AURenderCallbackStruct  callback;

description.componentType = kAudioUnitType_Output;
description.componentSubType = kAudioUnitSubType_HALOutput;
description.componentManufacturer = kAudioUnitManufacturer_Apple;
description.componentFlags = 0;
description.componentFlagsMask = 0;
if(component = FindNextComponent(NULL, &description))
{
    err = OpenAComponent(component, &fAudioUnit);
    if(err != noErr)
    {
        fAudioUnit = NULL;
        return err;
    }
}
param = 1;
err = AudioUnitSetProperty(fAudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof(UInt32));
if(err == noErr)
{
    param = 0;
    err = AudioUnitSetProperty(fAudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(UInt32));
}
param = sizeof(AudioDeviceID);
err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &param, &fInputDeviceID);
if(err != noErr)
{
    fprintf(stderr, "failed to get default input device\n");
    return err;
}
err = AudioUnitSetProperty(fAudioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &fInputDeviceID, sizeof(AudioDeviceID));
if(err != noErr)
{
    fprintf(stderr, "failed to set AU input device\n");
    return err;
}
callback.inputProc = AudioInputProc; 
callback.inputProcRefCon = NULL;
err = AudioUnitSetProperty(fAudioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct));
param = sizeof(AudioStreamBasicDescription);
err = AudioUnitGetProperty(fAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &fDeviceFormat, &param);
if(err != noErr)
{
    printf("failed to get input device ASBD\n");
    return err;
}
fDeviceFormat.mSampleRate = 8000.0;
fDeviceFormat.mChannelsPerFrame = 1;
fDeviceFormat.mBitsPerChannel = 16;
fDeviceFormat.mFormatID = kAudioFormatLinearPCM;
fDeviceFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
err = AudioUnitSetProperty(fAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &fDeviceFormat, sizeof(AudioStreamBasicDescription));
if(err != noErr)
{
    printf( "failed to set input device ASBD= %4.4s\n",(char *)&err);
    if(err == kAudioUnitErr_FormatNotSupported)
    {
        printf("kAudioUnitErr_FormatNotSupported\n");
    }
    return err;
}
err = AudioUnitGetProperty(fAudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &fDeviceFormat, &param);
if(err != noErr)
{
    printf( "failed to get input device ASBD\n");
    return err;
}   
param = sizeof(UInt32);
err = AudioUnitGetProperty(fAudioUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &fAudioSamples, &param);
if(err != noErr)
{
    fprintf(stderr, "failed to get audio sample size\n");
    return err;
}
err = AudioUnitInitialize(fAudioUnit);
if(err != noErr)
{
    fprintf(stderr, "failed to initialize AU\n");
    return err;
}

Here I am unable to change the sample rate and bit per samples from 32 to 16. Please any one help me to do this.

Thanks & Regards.

Vimal Alex
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3 Answers3

2

Capture with supported format and convert it to 8000 kHz,

With the above code AudioUnitSetProperty is going to be failed.

You need to create audio converter using AudioConvertNew and convert your buffer to desired format using AudioConverterConvertBuffer.

The iOSDev
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0

I had a similar problem, I had to modify sand box permissions. Check if your application is sand boxed, you have access to the microphone.

the Reverend
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0

You should check out the source code for SoundFlower. That might help to point you in the right direction.

Nik Reiman
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