I reworked it so the convolution shader processes the audio in tandem with playback, so I'm *very* close to getting this working with live audio streams.

But more importantly, I used this to convolve my song "strange birds" with a choir-ish fanfare sound effect from a game I used to play as a kid and the result is like the grand cosmos opened up before me and I'm awash in the radiant light of the universe. Absolutely incredible.

ok the problem I'm having with latency now is that the audio latency in the system grows over time and I'm not sure why. like it starts snappy and after running for a short while it gets super laggy :/

I'm guessing it's because SDL3 can and will resize buffers as it wants to, whereas I'd rather it just go crazy if it under runs.

What I want to do is have a fixed size buffer for input and output, enough that I can have the output double or tripple buffered to smooth over hitches caused by linux. if my program can't keep up I don't want it to quietly allocate more runway I want it to scream at me LOUDLY and HORRIBLY, but it wont do that because I'll rejigger my program until it is perfect.

What actually happens is (sdl? poopwire?) just infinitybuffers so it never hitches and I get a second of latency after a little bit

I'm like 30% sure SDL3 is not the problem or at least not the only problem because I tried resetting the streams every frame with SDL_ClearAudioStream and it still accumulates latency (in addition to also now sounding atrocious due to missing samples).

I've also seen this happen with pipewire before in other situations, and it was resolved by bypassing pipewire.

ok I did it. I've got a program that writes a pipewire stream of F64 audio samples where each sample is the total elapsed time since the first frame, expressed in mintues.

I've got a second program that reads that pipewire stream, and checks the offset against it's own elapsed time since the first sample processed. This program prints out the calculated drift ever second.

The results are interesting.

In the first version of this, both programs just measured the time using std::chrono::steady_clock::time_point. This resulted in an oscillating drift that was well under a millisecond at its peak and nothing to be concerned about.

This is good! That means there's no place what so ever within pipewire on my computer for this specific audio setup where any intermediary buffers might be growing and adding more latency as the programs run.

This is not the interesting case.

In the second version, I changed the first program to instead calculate elapsed time as the frame number * the sampling interval, and left the second program alone.

In this version, the calculated drift is essentially the difference between the progress through the stream vs the amount of time that actually passed from the perspective of the observer. In this version, the amount of drift rises gradually. It seems the stream is advancing just a touch faster than it should.

The samples in the stream are reporting that more time has elapsed in the "recording" than actually has transpired according to the clock. The amount of drift accumulated seems to be a millisecond every few minutes.

I'm honestly not sure what to make of that.

I think my conclusions from this are

1. the latency drift I observed with my experiments with pipewire today is probably inconsequential.

2. there is probably nothing sinister about pipewire.

3. if you have a chain of nodes that are a mix of push or pull driven and have different buffering strategies, you are in the Cool Zone

4. my program is probably going to have to handle "leap samples" in some situations. I admit I wasn't expecting that, but it feels obvious in retrospect.

that or I'm just good at creating wizard problems for myself. either way I'm in a good mood.

https://github.com/Aeva/convolver/blob/c5d1ca8ec8a4aafd640def16d68e1c84bbc6b240/src/convolver.cpp#L509

some small problems with this system:

1. I've had to turn down the sampling rate so I can convolve longer samples. 22050 hz works out ok though for what I've been messing with so far, so maybe it's not that big a deal. longer samples kinda make things muddy anyway

2. now I want to do multiple convolutions at once and layer things and that's probably not happening on this hardware XD

I figure I should probably start recording my convolution experiments for reference, and this thread seems as good a place as any to post them.

Tonight's first experiment: An excerpt from a The King In Yellow audio book convolved with a short clip from the Chrono Cross OST (Chronopolis)

Tonight's second convolution experiment: The same audio book excerpt, but convolved with a frog instead.

Recordings of speech seem to convolve really well with music and weird samples like this, but it really depends on the voice and what you pick as a kernel.