| Hi all,
I want to do a very basic CoreAudio setup using the AUHal AudioUnit component. The output should be 16kHz, 16-bit signed samples, 1 channel. See the code below. In the callback, AudioUnitRender is called, but that one fails with "Cannot do in current context". What is going wrong?
AudioStreamBasicDescription inDesiredStreamFormat;
AudioConverterRef captureConverter;
inDesiredStreamFormat.mChannelsPerFrame = 1;
inDesiredStreamFormat.mSampleRate = 16000;
inDesiredStreamFormat.mFramesPerPacket = 1;
inDesiredStreamFormat.mBitsPerChannel = 16;
inDesiredStreamFormat.mBytesPerFrame = (inDesiredStreamFormat.mBitsPerChannel >> 3) * (inDesiredStreamFormat.mChannelsPerFrame);
inDesiredStreamFormat.mBytesPerPacket = inDesiredStreamFormat.mBytesPerFrame;
inDesiredStreamFormat.mFormatID = kAudioFormatLinearPCM;
inDesiredStreamFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian;
AudioComponentDescription cd;
cd.componentType = kAudioUnitType_Output;
cd.componentSubType = kAudioUnitSubType_HALOutput;
cd.componentManufacturer = kAudioUnitManufacturer_Apple;
cd.componentFlags = 0;
cd.componentFlagsMask = 0;
AudioComponent cp = AudioComponentFindNext(0, &cd);
AudioComponentInstance auHal;
err = AudioComponentInstanceNew(cp, &auHal);
UInt32 enable = 1;
AudioUnitSetProperty(auHal,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
1, //input element; 0 = output element
&enable,
sizeof(enable));
enable = 0;
AudioUnitSetProperty(auHal,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
0, //output element
&enable,
sizeof(enable));
AURenderCallbackStruct cb;
cb.inputProc = inputCallback;
cb.inputProcRefCon = auHal;
AudioUnitSetProperty(auHal,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
0,
&cb,
sizeof(cb));
//retrieve default input device
//and set it on the audio unit
AudioUnitSetProperty(auHal,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global,
0,
&defaultInputId,
sizeof(AudioDeviceID));
AudioUnitSetProperty(auHal,
kAudioUnitProperty_StreamFormat,
AudioUnitScope_Input,
0,
&inDesiredStreamFormat,
sizeof(inDesiredStreamFormat));
AudioUnitSetProperty(auHal,
kAudioUnitProperty_StreamFormat,
AudioUnitScope_Output,
1,
&inDesiredStreamFormat,
sizeof(inDesiredStreamFormat));
AudioUnitInitialize(auHal);
AudioUnitStart(auHal);
The audio callback looks as follows: OSStatus inputCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { OSStatus err = noErr; AudioComponentInstance auHal = static_cast<AudioComponentInstance>(inRefCon); err = AudioUnitRender(auHal, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, gAudioBufferList); if (err != noErr) { qDebug() << "Error rendering:" << getAudioUnitError(err); } }
I also tried to use an AudioConverter, leaving the input as is, but then I get "cracks" in the converted audio.
Can someone provide me a simple example of how I could record 16kHz audio, signed 16-bit, 1 channel?
Thanks. |