Fully Featured Audio DSP Firmware for the Raspberry Pi Pico
github.com
github.com
Dirac seemed to have a fairly heavy-handed correction. In my case, I only had fairly narrow frequency ranges that needed correcting, but Dirac seemed to move much wider ranges at a time. It's also nearly impossible to tweak; you basically can only increase/decrease "the lows" or "the highs". But maybe I'm missing something.
In contrast, the suggestions produced by REW were loaded in EasyEffects on Linux, and I could tweak everything to my heart's content. But I actually just left it alone, since it was good enough.
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Maybe for more involved situations Dirac does a better job, but, in my case, it didn't really solve anything. Also, I see they now have this newer "bass control" thing, and it's not clear if my version had it when I last tested it (around November 2025).
Ideally you want to be going into it intenting to correct a specific aspect of the room or the speakers, after already ensuring that you've placed the speakers correctly for the room and listening position. If you did not use a tape measure and the full dimensions of the speakers, start over. One of the most useful things REW/Dirac can do for you is confirm that you've placed everything correctly. It is not a magic "make it sound better" utility.
Hate to sound like an ad but the most impressive thing I've purchased wrt audio in the past 20 years has been some isolators from https://isoacoustics.com/. It's legit engineering magic, you will spend the first hour thinking something is wrong with your body because you can no longer feel the sound.
Indeed, but I'd bet many people are in the "can't" category. Especially for low frequencies, you need pretty hefty treatment to make a difference, which is oftentimes impractical to install in a room which wasn't designed for that. And I seriously doubt any sizable number of rooms in apartments are designed for that. Combine this with the ungodly amount of snake oil peddled, and I can easily understand why many people look at Dirac and similar solutions.
And while they are band-aids, in many cases that's enough. I used to live in a studio apartment where room correction made a night and day difference to my listening position. Elsewhere the sound wasn't that great, but I didn't really care since I never listened from theme. I was renting, and the space was rather small, so there was no way to install any useful treatment.
In my current apartment it works much worse, it's actually close to useless. But it's rather bigger, so I could put in some treatment. But I've spent a lot of time researching this, and it's still not clear how to go about doing this. People can't even seem to agree on what kind of material to look at. And while I love listening to music, I'm not keen on investing thousands, plus time living under construction for weeks just to throw multiple solutions at the walls and see what sticks.
I have one and personally didn't bother, did the usual UMIK-1 + REW to create the room correction.
> https://www.minidsp.com/products/dirac-series/index.php?opti...
Dirac is the most user friendly of the bunch, but honestly once you limit the correction to below Schroeder frequency I cannot tell them apart. So for my systems I just stick to a few PEQs targeting the main peaks under 300hz.
The loudspeaker would have used one; a driver is both cheaper and of higher quality.
Also, for those watching for it: https://www.audiosciencereview.com/forum/index.php?threads/i...
> I can't take all of the credit. My little robot intern (Opus 4.5) has been very helpful with the busy work, leaving me free to handle the trickier planning and implementation. ;)
mov r3, r10
adds r5, r5, r3
mov r9, r5
mov r9, r10
add r9, r5
Since a Raspberry Pi Pico doesn’t have built-in audio output ports, I think the main thing blocking ordinary people from using it is figuring out the hardware? A link to a tutorial for how to add audio output would be useful.
There will also be an official plug-and-play custom board that includes all of the relevant IO, connectors and codecs.
I had a project in mind that was waiting for something like this! :)
For 2.1 configurations in a pinch, the firmware includes a software DAC that's more than adequate to drive a subwoofer, so only one external DAC is needed.
I mean, I know what you meant, but that's pretty misleading phrasing for many people.
It's not much of a stretch to think that most people interpret a "USB sound card" as a thing with analog audio on one side and USB on the other side. But other than the subwoofer output, we don't have any analog IO on a Pico running this firmware.
It is not 100% plug and play as you can choose your own software.
A custom board sounds great, too.
I wonder if 264/520 kB RAM is also enough for a high quality parametric stereo reverb/echo effect? Should fit about 3/6 seconds of uncompressed 16-bit 44.1/48 kHz audio.
Also: Raspberry Pi Ltd - please keep increasing the RAM size in future iterations to unlock even more use cases.
You can easily find dev boards with 8MB of PSRAM online if you need it. Or you can buy the PSRAM and hook it up yourself. If you still need more memory than that then you're looking at the wrong chip for the job.
Although, going back to the start of the thread where the suggestion was adding more RAM to future chips perhaps the request could be for support for multiple channels in the future.
It;s the age old question of parallel Vs serial Vs multi channel serial.
I’m sometimes annoyed that the home audio/audiophile world is so separate from the live/professional world.
For playing recordings with fancy effects, you can throw massive overkill CPUs at it with small batches, brutefir style, or you can do high-latency FFT filters, and you can get essentially perfect FIR reverb effects with a latency vs complexity tradeoff.
But the algorithm in the middle exists and is not that exotic. You divide your impulse response into a very short piece at the beginning, then a longer piece after that, then a longer piece after that, in exponentially increasing pieces. And then you add up the results, with straight addition and multiplication for the short one, and (carefully scheduled to avoid stalls) FFT convolution for the long ones, and you get basically arbitrary long FIR filters with logarithmic amortized complexity per sample and as low as zero sample latency if you are so inclined.
I think this is called “non-uniform partitioning” or something to the effect. I’m not aware of any serious, public implementation for audio use.
What are the odds a Raspberry Pi could keep up with BTrack?
If it's not doing anything else and the sample rates aren't outrageous it might be doable but I'd have to dig into the code more to see how much work they're doing per sample.
There are other projects for the Pico which implement S/PDIF in.
In either case, since it is digital, the quality (or lack of) of the internal ADCs should not matter.
The cheapest option is probably some Behringer mixer with enough inputs and multitrack interface over USB, like XR18.
For the $450 you get a lot of stuff. Preamps for mic and guitar pickups. Powerful headphone amp. It's clearly worth it if you make use of some of it, and potentially even just for the inputs alone. $450/8 = $56 per ludicrously clean input is good.
I bought an E1x2 kind of as a joke. Just to see how bad it was. It's actually really, really good.
And also:
It's actually possible to gang together multiple disparate audio interfaces. Let the audio stack keep them in sync with ASRC. Aggregate Device on macOS can do this. People say you can't but you can. Linux is good for this too. If you find a cheaper per channel input, this can actually be done; Piecemeal it.
For more inputs, a Behringer ADA8200 can be connected with a garden-variety TOSLINK cable, bringing the total of 16.
Or: Two UMC1820s, clocked together using that same TOSLINK cable. That provides 16 inputs that are all identical and also operating in lock-step.
In terms of cost: A smart way to play with this stuff is to buy used gear, and treat eBay as a long-term rental program. Just buy it, use it, and when you want to try something different: Sell it. It works because the depreciation on stuff like this is basically a straight line once the initial hit of turning "new" into "used" gear is over with.
The long-term rental cost then is mostly a combination of time, shipping expense, and seller fees. Keep it as long as you want. :)
edit: alright. so the UMC1820 is apparently having production issues right now, which constrains supply, so prices are higher than normal. On a normal day, they sell for $229 new. I've bought them for ~$100 used. Things will go back to normal soon enough.
The end-to-end delay is about 10ms, according to this comment:
https://www.audiosciencereview.com/forum/index.php?threads/i...
People use audio system at home to play electronic instruments. People also play video games. People do all kinds of stuff.
Latency is an important factor in these things.
Even videoconferencing and podcasting: With a microphone pointed at your face and a set of headphones used for monitoring that microphone, latency matters.
(It matters more to some people than others -- some people can tolerate hearing themselves later and continue to speak just fine, while some others increasingly sound like they're having a stroke as monitoring latency goes up and eventually become unable to produce coherent strings of phonemes.)
Latency is percievable by most people down to about 8-12ms .. lower than that and its harder to perceive, higher than that and you will get some people feeling like there are glitches in the audio ..
This is also important for musicians such as keyboard players, whose perception of their instrument is radically altered by that instruments latencies. Most modern synthesizers work very hard to get audio latency in their internal engines below 20ms ..
https://github.com/WeebLabs/DSPi/blob/main/Documentation/Roa...
But there seems to be new features being planned all the time, so who knows what it might do in the future.
With a Nyquist frequency of ~96KHz, all of the arguments about whether a person can hear up to eg 22.05KHz, 24KHz, or if there's something meaningful all the way up at 48KHz, become completely and totally ameliorated.
Those arguments were always such tiresome ordeals.
The cost of dissolving those arguments is just some some bandwidth and CPU cycles -- which is to say, it costs approximately nothing.
Please let the man cook. :)
And that's perfectly OK, too: The neat part about having too much data is that other end-users (like you and me) are free to throw it away as expeditiously as we choose to.
To that end: I, for one, welcome our 192kHz overlords. (And then I'll shove it through my hardware DSP that operates at 24-bit 48kHz and fuhgettaboutit.)
Because of stairsteps he can hear or something.
But even here in these comments, they're not arguing that 192kHz is insufficient. I haven't seen anyone ever make that argument, actually, so from my perspective this sampling rate represents a useful amount of overkill.
The extra data is a very small price to pay for silence.
Your DAW (or whatever) may be able to show you the stairsteps of individual samples on a screen, but with a functional playback system it is never that way at all by the time things become analog again. Instead, it's always smoothed out by an anti-aliasing filter.
It works this way regardless of sampling rate. The stairsteps don't make it outside of number-land. You can run your DAW at 48KHz, 96KHz, or 192KHz, and signals below the least-common-denominator cutoff frequency will be identical on an oscilloscope -- and free of stairsteps. (Try it sometime. It's fun.)
Aliasing is a solved problem that has been solved for a long time. Your analogy about scaling and diagonal lines is actually a decent visual representation of how this stuff works, except it has already been working that way without being deliberately clever with overkill sampling rates.
Meanwhile: This Pi Pico DSP stack is structured very heavily towards being the last digital stage of a listening system. As-constructed, it's quite clearly evident that it is really not meant to be anything else. A person can certainly bend it to be other things (yay open source!), but you've probably already got a set of filters well-integrated into your existing toolchain that work superbly.
But if that's what you want, then by all means: Use it. Integer sampling rate conversions are trivial operations to get correct. To get the 96KHz that this project works with from your your 192KHz workflow, it's just a matter of throwing away half of the samples and playing back whatever remains. Any aliasing is out-of-band, and is removed by the anti-aliasing filter that is part of the digital-to-analog stage.
You can't hear it if it isn't there. :)