I'm working on a project that requires sound output via the Windows audio subsystem, and so I've recently found myself using the LabVIEW "Sound" pallet vi's for the first time in.....well...ever!
The sound output ("Play") vi's allow you to setup and configure an output "task". As part of that configuration, you define a buffer size. Once that sound output task is configured, you effectively write your sound samples into this buffer over time, periodically refreshing this buffer with new audio sample data.
That's all fine and good, but unfortunately it seems like there is no mechanism to query the buffer status and find out if the buffer is about to overflow or underflow. While this might not seem to matter if you're playing a sound file for a few minutes on a machine with lots of RAM, it definitely does matter if you're streaming live audio continuously through the system for several days or weeks.
If you refresh the buffer with new audio sample data at rate that is just slightly faster than the audio card's configured sample rate, then it seems like the buffer will eventually use up all available PC RAM. if you are refreshing that buffer just slightly slower than the audio card's configured sample rate, then it seems like the buffer will eventually underflow and create a glitch in the audio output. And since there is no way to monitor whether the buffer is trending toward overflow or underflow, there is no way to figure out how to adjust the rate at which you feed new audio sample data into the buffers.
Am I missing something here? Is the audio subsystem doing some sample rate conversion that I don't know about? What is the proper way to ensure that the sound playback buffers do not overflow or underflow over an extended period of time?
