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New Wondom ADAU1701 DSP Unit

.... A common problem with low cost Chinese imports is bizarre wording caused by the writer's poor English language skills.
I wonder how well you can communicate in any Chinese dialect? English first speakers, of which I am one, are often very judgemental expecting a lot from non-english speakers. Frankly, given how typically awful we are at communicating in other languages I think people in glass houses should not throw stones.
 
How is this unit powered?
I use an HTPC, can i connect an USB to USB-C cable to power the unit?
(i want to use this unit to equalize my 2 DIY subwoofers)
Something like this?
- https://www.amazon.es/dp/B01GGKYO1I/ref=sr_1_1_ffob_sspa?crid=35HP6W2V6OJTP
It depends. The back panel says 5V 1A which is more than the standards specify for Type A ports. USB 2 must supply 0.5A and USB 3 0.9A. Type C may supply considerably more, depending on which standards it supports. Having said that, many USB Type A ports will happily supply more current than the spec calls for, and the DSP may draw less than its rated maximum, so it could work fine.
https://www.asme.org/topics-resources/content/explore-the-basics-of-usb-c-and-usb-power-delivery
 
I wonder how well you can communicate in any Chinese dialect? English first speakers, of which I am one, are often very judgemental expecting a lot from non-english speakers. Frankly, given how typically awful we are at communicating in other languages I think people in glass houses should not throw stones.
It's more that we expect sellers targeting a market with a different language to have made the investment in a proper translation, or make clear that they've used a machine translation that may have unintended consequences. I would expect the same of an English speaking company selling to countries with any other language, or get called out for it like these Welsh examples. Chinese manufacturers are more frequently called out because they manufacture a large proportion of products at the cheap and price sensitive end of the market, so more likely to have cut corners to keep costs down, including on translation. When it's a local brand rebranding a Chinese OEM product it's the local brand that deserves the grief for poor documentation.
 
Just some tips from me about subwoofer equalization,best way to equalize subwoofer is decreasing resonant frequencys and leaving dips as they are,lets say you have a -5db dip on 72hz,if you add +5 db peq gain, this area will be fixed,but if you move your microfon just a little bit, there will be dip on other frequency,and 72hz will have 5db boost,so best way to equalize your mlp is selecting maximum 3-4 db boost in rew for best results,you can set gain range in rew so it won't suggest you gain more than 3-4db in same menu I've posted picture before,as you can see on red curve no eq I had 12db error with huge resonance on 46hz, after eq there is 3db error which is really good,one thing that I like about this dsp that you can add up to 35 peq settings,so for example if you adjusted your curve and later you gonna move your furniture and after remeasuring your mlp you get errors,you can add few more peq settings to fix it,here is video measuring my subwoofer with wondom dsp:
Edit: this dsp should have 24db internal gain,so no need to reduce output volume.
 
What do you mean by this? If I add gain to the output will it clip the RCA output?
First I fought that you couldn't add gain,otherwise your output signal will be clipped,later I found out that adau1701 has 24db headroom,so if you add gain or peq with frequency boost,nothing bad will happen.
 
First I fought that you couldn't add gain,otherwise your output signal will be clipped,later I found out that adau1701 has 24db headroom,so if you add gain or peq with frequency boost,nothing bad will happen.
Will it not clip the outputs?
 
I am slightly confused, is the ICP5/USBi board only required if you want to save the settings directly to the unit?

Without one of those will it work in the same way as a minidsp for example? (i.e you just load the config saved on Windows with each reboot)
 
I am slightly confused, is the ICP5/USBi board only required if you want to save the settings directly to the unit?

Without one of those will it work in the same way as a minidsp for example? (i.e you just load the config saved on Windows with each reboot)
It is a lot more effort to update the DSP than minidsp I will say that. If you unplug the ICP5 it does take a while to get it to reconnect everything. Even updating sigma studio with filters is a lot more tedious than a simple dsp/EQ interface.

I would not recommend this dsp if you plan to constantly tweak the filters and update it. If you plan to set it once and leave it then it's pretty good.

One thing to note, It cannot do more than 2ms of delay, if you plan to use it for subwoofer alignment.
 
It seems to be limited to 5ms of delay
And by the way,why would you need to delay your sub for 5ms?Usually you need to delay your speakers,and if you would need to time allign 2nd subwoofer,probably you better leave delay as low as possible and play with subs polarity.
 
Will it not clip the outputs?
From the datasheet of ADAU1701, page 29:
"The serial port accepts up to 24 bits on the input and is sign extended to the full 28 bits of the DSP core.
This allows internal gains of up to 24 dB without internal clipping.
A digital clipper circuit is used between the output of the DSPcore and the DACs or serial port outputs (see Figure 29).
This clips the top four bits of the signal to produce a 24-bit output with a range of 1.0 (minus 1 LSB) to −1.0.
Figure 29 shows the maximum signal levels at each point in the data flow in both binary and decibel levels
."

ADAU1701 Clipping Structure.jpg


That means that the core has an additional 24dB headroom gain internally but if you don't attenuate it down to 0dB or lower at the Output to your DACs it will be clipped.
If you apply a 0dB signal in ADC Input, anything in your signal flow with a positive gain should be attenuated down to 0dB or lower before it leaves the Core.
You can use a full range Compressor or Limiter in your program to keep on the safe side.
 
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You'll lose the front panel potentiometer and switch functionality if you add additional audio I/O to the AA-AP23123, the GPIO pins you need are already used:

View attachment 329774
There should be another MP# used for the front panel "signal" LED.

From the Manual, page 4:

2. BUSY LED. BUSY LED is controlled by MP7. When MP7 is used as GPIO pin, set athigh level, the BUSY LED will be OFF; set at low level, the BUSY LED will be ON. MP7 is not defined now. You can customize its function in SigmaStudio.When MP7 is set as I2S1 data input,the LED will be ON when there is signal, OFF when there is no signal.

I guess that means the function of the led can be customized in the program, as a clip indicator for example.


But what if I want to use as I2S Input? How can that be configured? Don't we need more MPs available to make it work as such?
 
From the datasheet of ADAU1701, page 29:
"The serial port accepts up to 24 bits on the input and is sign extended to the full 28 bits of the DSP core.
This allows internal gains of up to 24 dB without internal clipping.
A digital clipper circuit is used between the output of the DSPcore and the DACs or serial port outputs (see Figure 29).
This clips the top four bits of the signal to produce a 24-bit output with a range of 1.0 (minus 1 LSB) to −1.0.
Figure 29 shows the maximum signal levels at each point in the data flow in both binary and decibel levels
."

View attachment 421364

That means that the core has an additional 24dB headroom gain internally but if you don't attenuate it down to 0dB or lower at the Output to your DACs it will be clipped.
If you apply a 0dB signal in ADC Input, anything in your signal flow with a positive gain should be attenuated down to 0dB or lower before it leaves the Core.
You can use a full range Compressor or Limiter in your program to keep on the safe side.
There is a level detector module,any gain applied to the input signal shouldn't clip output,as long as it stays under 0,so if my input signal is -60db and +20db additional gain applied it is still -40.
 

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