The loudspeaker would have used one; a driver is both cheaper and of higher quality.
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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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.
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...
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. ;)
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.
It is not 100% plug and play as you can choose your own software.
A custom board sounds great, too.
https://github.com/WeebLabs/DSPi/blob/main/Documentation/Roa...
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.
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.
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.
The cheapest option is probably some Behringer mixer with enough inputs and multitrack interface over USB, like XR18.
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.
mov r3, r10
adds r5, r5, r3
mov r9, r5
mov r9, r10
add r9, r5
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. :)
The end-to-end delay is about 10ms, according to this comment:
https://www.audiosciencereview.com/forum/index.php?threads/i...
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.
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.