Windows Audio (Low latency mode)


I’m trying the new “Low Latency Mode” of Windows Audio with juce 6.0.3. What surprises me (and I guess will cause some questions from my users) is that the low latency mode only suggests a buffer size of 480 on my realtek audio device, while both the shared and exclusive flavours of Windows Audio allow me to select 192 samples (and it works fine with that setting, although I have not checked if the actual latency is really lower than the “Low Latency Mode”).

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WASAPI low latency mode will directly query the driver for its supported buffer sizes so it sounds like it could be a driver issue, are your realtek drivers up to date? Do you see lower buffer sizes when selecting a different device?

The drivers are up-to-date. I have also tested with a yamaha audio interface, and it is also showing 441 (at 44100kHz) as the sole buffer size availablein “low latency” mode. I did not feel that the latency was lower than with the ‘basic’ Windows Audio. On the realtek, the Latency tester of the juce DemoRunner measures the same latency on “Windows Audio” and “Windows Audio (low latency)”. Also the latency reported when using a buffer size of 192 instead of 480 with “Windows Audio” does not change.

Anyway, it is not a big deal, I was just wondering if my setup is a exception or if “low latency” on windows is 480 samples at 48kHz for most people.