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To get the realtime benefits I'd have to run everything that requires realtime scheduling as an eBPF program, right?

I mean, suppose I want to make an eBPF program to pipe audio input into my very fancy DSP algo. Awesome. But now I've limited the remainder of my signal chain to remain in eBPF land, haven't I? The moment I throw the signal to Supercollider, Pd, some Jack client, etc., I'm back to being a mere userland mortal and taking the performance hit that entails. (Unless I sample that data coming into userland at a lower rate, but that severely limits the use cases for such a design.)

Judging by some of the screenshots I've seen Linux audio users seem to want to mix their samples through every single audio program they can get their hands on. So at least in this domain I have a hard time seeing how eBPF could practically help realtime scheduling.

Edit: clarification



> So at least in this domain I have a hard time seeing how eBPF could practically help realtime scheduling.

Right, because jack is a user space collection of programs. It's obvious that any real-time or low latency DSP audio processing pipeline would have to be moved to BPF in its entirety.

So yes, BPF would work just fine if a new DAW pipeline was designed to work with in it. User space programs would only function as GUI front ends to load and configure BPF audio modules. This way audio can be sourced or sinked via DAC/ADC devices, disk, and network devices without the user-kernel space latencies.


You're never going to fit Ardour's engine into 1M instructions, nor is it going to work without unbounded loops. So that's not going to happen.

More interesting would be using BPF for the driver and being able to directly invoke JACK (or some other user-space endpoint) in a way that essentially bypasses the scheduler.

Put differently: rather than rely on the normal scheduler pathway to waking up JACK (or some other user-space endpoint), have BPF do it as soon as the device driver is ready.

But I don't think BPF has any way to do this at present - it seems to rely on accumulation of data in kernel space and a user space poll-driven process to pick it up and push it at the user.


Does the eBPF program need to communicate with userspace?

For example, can there be an eBPF "hello world" Dalek modulator that just sits between ALSA's audio input and output?


sure there could, but that's utterly unrelated in practical terms to what i think we're discussing, which is ways in which eBPF could impact full-scale audio software rather than limited inline DSP.


Hm, good point. But couldn't you say the same thing about CUDA? There is an overhead involved in shunting data to and from the cores ... but the nature of the work being done, and the massive parallelism, mean the overall end-to-end volume of throughput increases dramatically.

I don't know enough about audio processing to answer your specific question ... but processing network traffic for instance, you have the opportunity to "do things" with traffic. For instance partially decode PDUs to see if they interest you, and then pass them on to user-land for more in-depth processing if so ...

EDIT I think you're thinking in terms exclusively of `mapping` operations, but this kind of thing can be great for `reducing` or filtering.


There's bandwidth and there's latency. CUDA has the bandwidth, but not the latency, for generalizable low latency audio processing.


It could still be considered low latency by adding the constraint that all use cases are in the category of call-and-response musical forms. :)


Great idea for audio but that sounds very challenging given BPF's lack of support for floating point types and instructions [0].

I suppose it's possible to do audio processing being limited to integer arithmetic. BPF does have multiply and divide instructions at least.

[0] Scanning the x86_64 JIT compiler for example yields no mention of SIMD instructions: https://github.com/torvalds/linux/blob/master/arch/x86/net/b...




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