I'm slowly working through the vulkan spec writing a compute-only vulkan program from scratch that doesn't render anything, and it's going pretty well because the spec is really well written and I already know more or less exactly what I want to do anyway, but I just want to say just how silly (fun) it feels to write a program like this because you get to just skip over large swaths of the API.
Like, I'm working from the spec because the tutorials all make it more complicated.
also the tutorials I reviewed all did the annoying thing where the tutorial squirrels away the stuff you're trying to learn or reference into abstractions that only serve the needs of the tutorial writer, which given my goal is very specifically to *not draw anything*, there's really not much of a point to any of them lol. I'm really not the intended audience here though :3
I think it's cute that practically every vulkan command has one or more optional args to let you enter Hard Mode
(sorry for the double post, I added this to the wrong thread)
I wonder how many people have actually managed to knuckle down and write a complete, useful vulkan program from scratch (no copy pasting from tutorials and stack overflow, no offloading significant parts to 3rd party libraries like VMA)
To think if I power through and get this thing working I could potentially be like the 20th person to bother
oh, update on my little vulkan compute project, last night I got as far as repeatedly dispatching an empty compute shader and allocating some memory 😎 I'm in the home stretch! I think I just need to figure out the resources / resource binding stuff and then I'll be able to start on my DSP experiment :3
which mostly means the next things are figuring out the least effort way of getting audio data into C++ (probably stb_vorbis?) and writing even more boilerplate for alsa...
Success! I got the vulkan compute shader cranking out the fibonacci series and reading it back to the CPU through a 8 byte persistently mapped buffer. Should be smooth sailing from here.
ok *whew* I finally did it! I implemented convolution reverb as a vulkan compute shader, and the results seem to be correct. I have it convolving the audio up front at the moment, but it seems to be reasonably fast and the results seem more or less correct. I'm using SDL3 to verify the output. It doesn't look like it'll be too crazy to rework it such that the stream is generated live.
it turns out the main difficulty working with vulkan is accidentally breaking your laptop in half
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.
I want to power through and get this into a state where I can use it with live instruments, but I am completely exhausted 😴
I reworked some things and now my audio convolving compute shader can convolve ~11 milliseconds worth of audio samples with an average processing time of ~7 milliseconds. That's with one channel with a bit rate of 22050. When the bit rate is 44100, the average processing time is a paltry ~8 milliseconds.
also sometime in the last week I made it so it can operate entirely on a live input stream from SDL3 rather than a wave file, so in theory I can incorporate this into a modular setup now, but the results are higher latency than I'd like, and SDL3 doesn't give you much control over audio latency.
Apparently my best frame time can get as low as 3 ms. I think vulkan should let me VK_QUEUE_GLOBAL_PRIORITY_REALTIME this program, but sadly vulkan is being a coward about it.
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 data 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 it (sdl? poopwire?) just infinitybuffers so it never hitches and I get a second of latency after a little bit
I like that pipewire has an option to not be terrible ("pro audio" mode) and it doesn't work
99% of audio problems on linux these days are just programmers refusing to just fucking use alsa. I'm part of the problem, because I'm using SDL3 instead because the API is simple. SDL3 is part of the problem because when I tell it to just fucking use alsa it uses pipewire instead! and pipewire is part of the problem because it's just completely terrible. like, wayland terrible.
want to have low latency audio on linux? we have a tool for it, it's called STOP PILING LAYERS OF BOILERPLATE ON TOP OF ALSA YOU IDIOTS YOU ABSOLUTE FOOLS
@dos sounds like what I need is to throw it in the garbage and just use ALSA