Hopefully the title + picture are self explanatory. I built a 4ch TPA3255 board amp to run some DML panels that need heavy EQ and a powered sub to cover the full spectrum. So I have a 4ch ADAU1701 with 2 channels going to the panels and 2 to the subs. I can't find the parametric EQ in the SigmaStudio tools. I looked through other folders but everything I see in documentation and other discussions says to look in the "Filters" folder. Any ideas?
Or if you like to work just with biquad blocks:
Filters -> Second order -> double precision (or single precision) -> 1 Ch -> General (2nd order)
Click on the little graph button on the filter block and select Parametric (or any other filter type you want)
Filters -> Second order -> double precision (or single precision) -> 1 Ch -> General (2nd order)
Click on the little graph button on the filter block and select Parametric (or any other filter type you want)
Thank you! Is it possible to send multiple channels through 1 PEQ?
General 2nd order won't work as I have a ton of filters to put in. I will give this a try.
General 2nd order won't work as I have a ton of filters to put in. I will give this a try.
There is no difference between "general 2nd order" and multichannel in sense of memory.
It's just visually different.
For those General 2nd order filter blocks, you just use more channels, up to 6 channel.
Which will get all the same filter. This can be tested with the "simulation probe" and "Simulation Stimuli" (on the top left, under schematic design")
For the "Multichannel" option, click on the filter block -> right mouse button -> grow algorithm and select amount of channels.
I think, I personally never use this multichannel filter block.
For a general filter block, the same action can be used to add more filters as well (add algorithm) and/or make more channels.
Basically doing the same thing as the multichannel option.
My personal preference why I like to use those general filter blocks, is because I can see immateriality the amount of filters when opening a new filter preset/project in Sigmastudio.
Otherwise you first need to open a new window.
The other advantage, is that in some cases it can be handy to have filters in parallel as well, not only in series.
btw, depending on your filter setup, and how the channels are being split, but often it's better to add common filters to the "main channel" (so to speak).
That way there are less issues with phase issues and such.
Linkwitz has a page on his website explaining this very well.
In case of a ADAU1701/1401 this is also nice, because it will save you some memory.
That way there are less issues with phase issues and such.
Linkwitz has a page on his website explaining this very well.
In case of a ADAU1701/1401 this is also nice, because it will save you some memory.
@b_force you're a life saver. I will just use the multichannel PEQ for now but if you have good resources on how to set up filters I'd be happy to read them.
I'm also curious if there's any easy way to dump a set of known filters into a SigmaStudio PEQ. Though for whatever reason previous filters I've tried don't seem to sound the same. I will prob have to retest and start over with the filter designs.
I'm also curious if there's any easy way to dump a set of known filters into a SigmaStudio PEQ. Though for whatever reason previous filters I've tried don't seem to sound the same. I will prob have to retest and start over with the filter designs.
I am not really sure what you mean with that last bit?
"known filters"?
There are sometimes some rounding differences, especially in single precision mode.
But otherwise the output isn't any different?
To save the PEQ filter settings: Right click on the PEQ and select "Cell Settings" followed by "SaveAs". This will save the parameters as a .bin file and let you change the file name and/or folder if you want. You can then re-load the .bin file (Right-click, Cell Settings, Open) into this or another PEQ.
@ernperkins got it. I calculated some PEQ filters from measurements in REW. Was looking for an easy way to port them to SigmaStudio.
Oh right, yeah.
There are IIR coefficient blocks as well.
I stopped directly importing parameters from 3rd party programs.
Had way to many issues with it to be consistent.
So I usually copy the parameters by hand, which isn't that much work either.
But maybe it works totally fine these days. 🙂
Sorry about reviving an old thread, but this is why I joined.
I just ordered the Dayton Audio ADAU1701 board for a project so don't have any first hand experience with it yet. I do although have plenty of experience with pro audio DSPs. What I can tell you is that it's a bad idea to dump settings from one DSP into a different make/model of DSP. For the same make/model this is a safe practice and often fine for different model, same make. It's fairly annoying, but this is the reality. If you want the best possible response from a different DSP, always remeasure.
I will illustrate the problem with a PEQ. Say you have a 4dB bump at 440Hz that you're trying to tame as measured in REQ wizard. So you introduce a -4dB PEQ at 440Hz with Q=10. After implementation and remeasuring, you will likely find that you have brought the output closer to flat, but off just a little. To make the region around 440Hz flat, you end up changing it to a -5dB, 452Hz, Q=10.4 and it's near perfection. Now, you enter these specs into a different DSP and are confused and frustrated that it doesn't work nearly as well. After measuring and adjusting, you get that near perfection again, but the settings are -3.5dB, 433Hz, Q=9.7. This is essentially a real world example. "Essentially" only beause I can't remember the actual numbers.
With multiple PEQs that likely overlap at least a little, this problem gets compounded and will make you pull out your hair trying to figure it out. Best bet is to start over.
The lesson to be learned here is that you can't trust the numbers on your DSP. Maybe there is a $10k DSP out there that has razor sharp accuracy, but my experience is that they are all slightly off. Don't get me wrong, most are pretty close to what they say but off just enough to cause translation problems and make you question if your DSP is broken. Think about the situation like a volume knob or fader marked with -40, -30, -20, -10, U, +10dB where these are ball park figures and taking measurements with an accurate meter will uncover that it's off by 5%. If it's decent gear, it will be consistent and predictable but always off just a little.
My question is always? Why would you want that?
I always either focus on the just the tool I am using.
In all those years, I never had to pull out my hair for these kind of issues?
Yeah, they might be off by like 0.25dB or so, or like 1Hz?
Both are not really important for the end results, plus I will always tweak the end result for the tools and components it's using in practice.
Or personally think that some people focus on the wrong details.
oh, fyi, a $10k DSP is not gonna fix these issues either.
In fact, I don't see any reason why someone would spend that much money for just a DSP.
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The $10k DSP is a sarcastic nod to the esoteric audiophile people who will spend all gods money on a sound system and put it in an untreated concrete bunker, then claim that they can hear subtle differences that you would need a $1k mic and an $8k o-scope to measure.
The DSP differences I'm speaking of are most definitely audible. The example I used has a difference between the two of 1.5dB, 19Hz, and Q of 0.7.
At 440Hz, a change of 19Hz is enough to pass the next note on the standard scale. There are usually multiple PEQs, crossovers, delays, etc. that all have to work together to flatten response and when you start adding all these together with overlaps and they are all off just a little, it adds up to something that sounds WAY off the flat response that was previously achieved. One time in particular, I started questioning everything about the system until I took a plot without DSP that exactly matched what I expected and with DSP was nowhere near flat. After an hour of tuning it, it sounded just as good as the previous DSP but comparing to the previous DSP settings, everything was off just a little. Until this point, I had always chalked up the difference in sound to different environments which is a significant factor but now I know that most of it is DSP differences.
Also, it may be worth mentioning that my use of DSP is mostly to tune horn loaded speakers that have some ripple in response. For example, a mid-horn will usually have delay, high pass and low pass crossover, and at least six PEQs to achieve a relatively flat reasponse.
The DSP differences I'm speaking of are most definitely audible. The example I used has a difference between the two of 1.5dB, 19Hz, and Q of 0.7.
At 440Hz, a change of 19Hz is enough to pass the next note on the standard scale. There are usually multiple PEQs, crossovers, delays, etc. that all have to work together to flatten response and when you start adding all these together with overlaps and they are all off just a little, it adds up to something that sounds WAY off the flat response that was previously achieved. One time in particular, I started questioning everything about the system until I took a plot without DSP that exactly matched what I expected and with DSP was nowhere near flat. After an hour of tuning it, it sounded just as good as the previous DSP but comparing to the previous DSP settings, everything was off just a little. Until this point, I had always chalked up the difference in sound to different environments which is a significant factor but now I know that most of it is DSP differences.
Also, it may be worth mentioning that my use of DSP is mostly to tune horn loaded speakers that have some ripple in response. For example, a mid-horn will usually have delay, high pass and low pass crossover, and at least six PEQs to achieve a relatively flat reasponse.
I am using the WONDOM device to be a preamp that sends high to my old two-channel amp and lows to my subs. I want one PEQ block to do room/speaker compensation, and one to correct my hearing loss. I have a measurement mic and REW, but I am not sure if I should mess with the PEQ manually or if I should find a way to auto-generate a list of corrections. What do people recommend?
Sigma Studio has an auto EQ that can calculate equalizer settings based on the measured frequency response, you can set the final frequency response to any shape. After automatic calculation of the equalizer parameters, you can fine-tune manually.
You can see how to do this in this topic in post number 18.
https://www.diyaudio.com/community/threads/3-way-dsp-amp.415065/