VB-Audio's Forums

Hi,
I have a problem with clicks on all “soft synced” asio devices = all asio devices that are not master. Every 2 or 3 minutes I get a click or crackle or burst (depending on the audio material)

I would expect it to use a proper sample rate conversion algorithm to make an audio-connection possible between devices that are not in sync. Instead it behaves as if you jut connect two digital devices with independet clocks - it seems to work but from time to time you get clicks when it missed a sample…

[- it works fine after changing the asio device to master (but that’s inconvenient and makes impossible to use two devices at the same time - like for headphone and speakers)

- when using a different sample rate on that device the conversion itself works fine (no audible difference from 48 to 96 for example) but still you get those clicks from time to time - so something seems to be wrong with the handling of audio between non synced devices…]

Best
Sebastian

yes, you can get some more or less hearable artefact on soft synchronized devices (device running on their own clock) with different QOS : one per minute, one per hour... pending on clock differences and also pending on the latency buffer size in use (the first way to minimize them is to increase the buffer size and use the SECURE "latency performance mode").

For most of users it works pretty good because using device with same chipset or same clock speed... There is more problem when using old devices, built with different generation of chipset/clock from different manufacturers...

We are still working on minimizing it or make them not hearable, but the only true solution to avoid this problem, as you may know, is to link all devices with a common WordClock (only present on professional devices).

Everything is explained in the documentation page 26: https://vb-audio.com/Matrix/VBMatrix_UserManual.pdf

Hi Vincent,

thanks for the clarification! Since it can use a sample rate conversion for these indepentent ASIO devices I thought it would work like an AES-input with asynchronous sample rate conversion on a hardware device: You connect a device with independent clock and the sample rate conversion algorithm takes care of making a clean signal with internal clock out of it (doing internal oversampling, like this: https://www.analog.com/media/en/technic ... 268v01.pdf)

Do you think it is possible to implement this into the matrix software at some point? (with that it would be an incredibly powerful tool

Best
Sebastian

Thanks for the document reference but the problem we have under windows cannot be solved by a DSP solution (at least not directly).
Windows is not an audio system, there is no strict timestamped callback (no interrupt timer), no fixed buffer size, no precision clock.

But we are still working on...