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That is as expected! It's just the sinc reconstruction filter. Look around the zero crossing, there you should have nice steep flanks.

Sine waves are not great for measuring the delay accurately. At 100 Hz the period is 10ms, and 30 us is only 1/300 of a period. Can't do much better that that! Try with a slow square wave instead, like 10 Hz. The sharp flanks are much easier to compare.

The RPi2 and 3 have issues with usb. There are lots of reports of sound glitches when using them with usb dacs. The RPi4 and 5 have superior usb implementations and are much better choices.

You mean the worst you have seen was 210 clipped after a full day? I would not worry about that. When the clipping is rare, then it's usually just one or a few consecutive samples that clip on each event, and that's really difficult to hear even if you really try and listen for it.

There are several filters with quite high q-values. These can give significant ringing. Your clipping probably happens when unfortunate music signals excite this ringing strongly. How many clipped samples do you get? If it's only a handful then it's likely inaudible and nothing to worry about.

This can be done just fine in Windows too, as long as the application uses Wasapi exclusive mode. But that's not at all what the question was about. It's about receiving different sample rates on a split spdif (thanks autocorrect!) input, and there is simply no good solution for it. Doesn't...

That's what I was referring to! Specifically this post: https://www.diyaudio.com/community/threads/camilladsp-cross-platform-iir-and-fir-engine-for-crossovers-room-correction-etc.349818/page-87#post-6579441

The new version behaves the same.

It looks like that card doesn't support interleaved access, which is the mode camilladsp uses. Try adding a plug to convert. It should be ok to just change "hw" to "plughw", like this: device: "plughw:HDSPMx000000"

@MarcosCh this project may be interesting: https://github.com/raptorlightning/I2S-Hat

I don't know about the S2 digi, but in general soundcards and interfaces with spdif input are very poor at handling changes in the spdif input sample rate. You select the rate when opening the device from the application, and this needs to match the incoming rate. A mismatch often leads to...

Something like this? https://github.com/joakimfors/PyMasVis

I haven't tried myself but there seems to be some trouble using camilladsp and peppymeter together. See https://github.com/HEnquist/camilladsp/issues/218

My service file was made for systems running with a graphical desktop. I didn't consider headless systems etc and to be honest I didn't think too much about the setting of WantedBy, I just picked one that worked. For a system without a graphical desktop it definitely needs to change.

The new "Uses Sourceforge config?" row has several errors. Should be no for at least camilladsp, brutefir and EqualizerAPO. The row name is also weird, what does soureforge have to do with it? Call it "uses Convolver .cfg config format" or something like that.

In my opinion a good config file is both human- and machine-readable. That .cfg format fails miserably on the human-readable part. I also dislike that it is a very fragile format where only the line number and position of a number determines what it is. Any structured format (yaml, json, xml...

I still think filter banks should be unlimited for camilladsp. It can easily implement an equalizer as well. And it has a loudness-compensated volume control that works the same way as the one in RME ADI-2 FS. Not exactly the iso226 curve but subjectively it gets quite close. For special...

There is something weird with VB-Audio Cable. I had to implement a workaround to get it stable with CamillaDSP, see here: https://github.com/HEnquist/camilladsp/blob/master/src/wasapidevice.rs#L494 Without that it works most of the time, but when it decides to not work it stutters horribly. I...

We are not talking about the same thing. I mean the concept of presenting a virtual playback device, so that sound from normal desktop apps can be captured and processed. I'm not sure what you are talking about, low-latency advantages of using asio maybe?

This is system-wide processing, that it is able to process the sound produced by any app.That in itself doesn't have anything to do with asio. But the processed audio may be fed to an asio device if the convolver app supports that.