Hi, I’ve stumbled across this: I’m building a simple test app based on the AudioDeviceManager tutorial, i.e. standalone audio application (under Windows), with practically nothing added but code that sums up buffersToFill.numSamples in getNextAudioBlock and then displays how many samples per second are collected this way. For a sample rate of 44100 and an audio buffer size of e.g. 1024 this pretty good fluctuates around 44100, for audio buffer size 512 it settles around 41000, for size 256 around 47000 and for 128 it goes down below 37000. CPU usage (which the tutorial displays) is below 1% in all cases.
Can my observation be true? Is this a problem of stand alone where e.g. cheap audio hardware/driver software (Realtek in a laptop) does not provide a proper reference clock and would it be gone if used in a DAW context?
Audio Device Type (top of the audio device manager gui component) is set to DirectSound in all cases. If I set it to Windows Audio numbers go completely out of order.