Sure DSP ADAU1701 board with amplifier JAB3-50

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Hello everyone
I already asked in the profile branch, but I did not receive any answers, then within 2 weeks I looked for information on the Internet, but I did not find the answers to my questions and therefore I created a topic in the hope that someone will see and answer


Within my project I have to create amplified subwoofer (40-200 Hz and max volume 3 L). I know that this is a challenge to reach this parameters, but I would like to discuss just electronics part.

I found JAB3-50 Audio Amplifer with DSP. I am really interesting in this solution, but I need to clarify some practicals aspects of this solution. Since I have limited budget it is very iportant for me to know all aspects, because Sure web site does not provide full info. Also I have contacted to Sure via mail, But I recieved answer from manager (...you know)

So, could you, please, answer and clarify some information:

1) To get 2.1 (I need to connect mu sub to the other passive 2 channel sustem) I need to use JAB2 & JAB3, but in this case does it work like whole system, I mean if I want to use Bluetooth connection will I have output sound from both JAB2 & JAB3 (Bluetooth will work for these two boards). In general, bluetooth connection is good?

2) In case of using input audio via cables (not Bluetooth) where I have to connect my input signal to the JAB2 board? or JAB3? I did not find this information

3) In case of using JAB2 & JAB3, if I understood correctly, DSP on JAB3 provide 4 output channels? So in Sigma Studio can I use 2 channels from JAB2 and 2 from JAB3?

4) Can I use many filters in Sigma studio and what are the limitations for this DSP ( I mean how many filters or types of filters or memory and so on...) ?
How many instructions I can use with ADAU1701?

Because I have ADAU1772 evaluation board (for a test) and inside Sigma studio there are just 32 instructions and limited toolbox for the dsp

5) If someone had the problem, when this two board are in idling regime (without input signal) there are some noise? Because I have seen some video on YouTube, when this problem was take a place ( YouTube )

6) If I want JAB3 only for subwoofer channel I have to connect two output channels in one channel, in this case all parameters for this 2 channels in sigma studio should be identical? is not it?

7) To provide DSP without Sigma studio, In-Circuit Programmer- ICP1 should be connected all the time to JAB2 & JAB3?

8) If I choose bluetooth regime on ICP1, all DSP settings are disabled?

Because it would be great just to control the volume from the Sure app from smartphone and using self-boot EEPROM. Or this is not possible?

9) What about the problem of 12 MHz crystal. Is the problem fixed? now all boards are 12.228 MHz crystal?
But what is critically to use the board with 12 MHz crystal?

10) Are there any other bugs and is this solution generally good?

What about amplifiers of this board? Are they implemented well? will they work with 2 Ohm load?

Sorry for the big questions.But I really hope for answers and I will be very grateful

Thank you so much!
 
Did you ever find the answers to you questions ?
I'm especially interested in you findings regarding effectiv channels (are ther 4 in the jab2-jab3 combo) ?
And do you need the jab2 ? (I have no need for Bluetooth, is there some way to use another amp for the last 2 channels)

Br
 
Wow, that's a lot questions! I’m not surprised no one has replied. I did some testing on these boards last fall, but nothing since as speaker building is a higher priority for me. I bought a JAB2-50 and a JAB3-50. Since then the JAB3-50 has been replaced with the JAB3-250 and Sure has added two single channel subwoofer versions of the JAB3. I'll answer to the best of my ability and if I'm wrong somebody should respond. I attached a couple of files, one from Sure Electronics (Steps 1 -5.pdf) that shows how to connect the devices and one from Walter Fehrmann (Generic DSP 2-way crossover.pdf ) that has a lot useful information on using SigmaStudio. Neil Davis also has an excellent site, Audiodevelopers Reborn – A collaborative website for active speaker design. Check out his “Articles” section. So onto your questions:

1) To get 2.1 (I need to connect my sub to the other passive 2 channel system) I need to use JAB2 & JAB3, but in this case does it work like whole system, I mean if I want to use Bluetooth connection will I have output sound from both JAB2 & JAB3 (Bluetooth will work for these two boards). In general, bluetooth connection is good?

Yes, they will work as a whole system. The SigmaStudio program determines what gets sent to JAB2. See my response to question 3 below for more information. I never tried the Bluetooth connection, but I believe it should work.

2) In case of using input audio via cables (not Bluetooth) where I have to connect my input signal to the JAB2 board? or JAB3? I did not find this information

See the Sure Electronics file.

3) In case of using JAB2 & JAB3, if I understood correctly, DSP on JAB3 provide 4 output channels? So in Sigma Studio can I use 2 channels from JAB2 and 2 from JAB3?

Yes, you can get up to four separate channels. DACs 2 (left channel) and 3 (right channel) are wired to the JAB3 amplifiers while DACs 0 (left channel) and 1 (right channel) are routed to the J5 connector, the audio extension connector. When you connect JAB3 (J5) and JAB2 (J11) with Sure’s “Communication Cable” the outputs of DACs 0 and 1 are fed into the inputs of JAB2. The SigmaStudio program determines if JAB3 is fed a mono or stereo signal. According to Sure’s SigmaStudio program (HERE) the left and right channels are first summed (“M Mixer1” in the program), filtered and passed through a volume control. This mono signal is then fed to both DACs 2 and 3, so you get a (loosely defined) 2.1 system. I say “loosely defined 2.1” because JAB2 is simply outputting two channels of mono. You’ll need two subwoofers if you want to use both channels.

Important note: There is volume control (potentiometer and cable) option that can be used with the JAB2 when used in stand-alone mode. It can NOT be used when the JAB2 is paired with the JAB3. The reason is the volume control cable uses the same connector (J11) as the cable that interconnects the JAB2 and JAB3 for paired operation.

4) Can I use many filters in Sigma studio and what are the limitations for this DSP ( I mean how many filters or types of filters or memory and so on...) ? How many instructions I can use with ADAU1701?

About the only limitation with the ADAU1701 is if you’re interested in FIR filters. Otherwise I think it has more capability than you’ll need using SigmaStudio IIR filters. At 48 KHz you have 1024 instructions per sample. At 96 KHz you get 512 instructions per sample.

5) If someone had the problem, when this two board are in idling regime (without input signal) there are some noise? Because I have seen some video on YouTube, when this problem was take a place ( YouTube )

Watched the video, seems the noise only occurred if Bluetooth and the Aux cable were connected and he found a solution.

6) If I want JAB3 only for subwoofer channel I have to connect two output channels in one channel, in this case all parameters for this 2 channels in sigma studio should be identical? is not it?

Yes, just like the Sure SigmaStudio example. As stated above, you actually have two channels of identical mono since the JAB3 is a stereo amp. If you want JAB3 to be a true subwoofer amp get the JAB3-160 (1X60) or JAB3-1100 (1 X 100), although I don’t know how the DACs are setup on those boards.

7) To provide DSP without Sigma studio, In-Circuit Programmer- ICP1 should be connected all the time to JAB2 & JAB3?

No, not if you write your program to EEPROM. See this YouTube video: YouTube.

8) If I choose bluetooth regime on ICP1, all DSP settings are disabled?

Did you mean Bluetooth on the JAB2? ICP1 is just the programmer. The answer is no; Bluetooth is just another input, like the Aux in cable. All DSP settings should work for Bluetooth.

Because it would be great just to control the volume from the Sure app from smartphone and using self-boot EEPROM. Or this is not possible?

Not without adding a microprocessor to send commands. Neil Davis has done this though, see his website. He’s done a lot of great work.

9) What about the problem of 12 MHz crystal. Is the problem fixed? now all boards are 12.228 MHz crystal? But what is critically to use the board with 12 MHz crystal?

The 12 MHz problem may have been only for the Sure ADAU1701 board and not for the JAB3 which came later. AFAIK all the newer ADAU1701 boards have the correct (12.288 MHz) crystal.

10) Are there any other bugs and is this solution generally good?

These are OEM boards and documentation is sparse for DIYers. We’ve pieced together a fair amount of documentation in the original thread, but nothing’s guaranteed in DIY. And there are errors in Sure’s documentation too. For example, in “Step 4” of the Sure document there’s this note: “When JAB3 is used to work with JAB2, make sure J9 on JAB2 and J12 on JAB3 is connected by a 3Pos cable, then J5 on JAB2 will be used will be used to be connected to switch cable.” There are a couple of problems with this statement:

1. I think J9 on the JAB2 (2 pins; Vcc and Gnd) is the wrong connector. I think it should be J5 (2 pins; EN and Gnd).

2. J12 on JAB3 is a 3-pin connector (Stby, Gnd and Mute). I believe a custom cable is needed that connects J12-Mute to J9-En and the two Gnd pins. I haven’t done this, just looking at the pin definitions. I’m assuming the GPIO_11 pin (see the Sure SigmaStudio program) is connected to the J12-Mute pin.

So expect to run into some problems, just like the guy in the YouTube video that fixed the feedback issue. A lot of problems could be quickly corrected if Sure would release the schematics to DIYers. I believe you have to purchase several hundred boards to get the schematics.

What about amplifiers of this board? Are they implemented well? will they work with 2 Ohm load?

I did some load testing and measured distortion in the 0.1 to 1% range at 1 watt. These boards were designed for applications like kiosks so low distortion wasn’t a high priority. My opinion is the 2 channel boards (BTL) should be limited to a minimum of 4 ohms while the single channel subwoofer boards (PBTL) should handle 2 ohm loads. Other more knowledgeable people can give their opinions.
 

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  • Generic DSP 2-way crossover.pdf
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I'll answer to the best of my ability and if I'm wrong somebody should respond.
wow thank you! Some info is useful for me!

I asked sure and the gave me some answers but it seems like I was mailing with manager)

So I ordered Jab3 2*50 and Jab2 2*50 (waiting for the delivery). I want to use this like: 1 channel for subwoofer (Tang Band W5-1138SMF) and 3 other channels for small sound bar of 6 speakers (one is 10 W). But I am still worry about power. Its mention that 50 W RMS 4 ohms by channel. But is this real rms or ''Chinese" Watt?

Since my small speakers is 4 ohm and I want to connect them 2 in parallel to one channel, I hope this amplifier will work with 2 ohm otherwise I will use resistor to compensate resistance (to get 4 ohms). What do you think about this?

Actually I asked about Sure programmer...It contains also EEPROM, is not it?. can I save my settings from sigma in EEPROM and use this like standalone system? in this case sure programmer should be connected every time?

...Or EEPROM is on JAB3 board?

So if I understood I can use limiters in Sigma studio, but how to control volume for both boards? I did not understand is it possible?

Or only way is to change the volume of the input aux signal? (or Bluetooth source?)

Because sure told me that I can use that potentiometer in cable kit for both boards...


What do you mean that JAB2 is not stereo? or I did not understand... If I send stereo signal to the input I get amplified stereo signal as well, is not it?


thank you!
 
I don't have any experience with the JAB boards but do have 3 of the stand alone DSP boards, one I bought last November and the other two in Feburary this year, all have the 12.288MHz crystals.
To program the eeprom I leave the switches on the ICP and DSP in position '1', power on the DSP first load Sigmastudio, connect the ICP to the USB port and check it shows 'green' in the hardware config tab in Sigmastudio and lastly connect the cable to the DSP. this allows me to control and or write new configs to the DSP for selfboot.
For the new config to be able to boot from the DSP I will write the config to the DSP and then remove the cable connection between the ICP and the DSP, then swith power off and back on again to the DSP, it will now boot with the new config.
With the DSP powered back up, reconnect the ICP and you will have control over the parameters.
My Dsp's are built into 19" rack cases in the amp rack and the ICP built into a box using a 5 pin DIN for connection, no 5V required on this link. Access to switches on the DSP would be a pain to get at so this is a solution I found that works for me.
The DSP's are FIR capable but you will have under 500 to play with, in practice with crossovers, limiters EQ etc I use about 150 - 200 on my fullrange 12" to EQ above 1KHz in stereo.
Connecting my USB soundcard and the ICP at the same time causes a lot of noise, using the headphone out and the ICP there is still a small amount of noise, I haven't tried investigating further yet.
 
I want to use this like: 1 channel for subwoofer (Tang Band W5-1138SMF) and 3 other channels for small sound bar of 6 speakers (one is 10 W).

I don’t quite understand how you intend to distribute four channels of amplification to your subwoofer and three channels. For your 3 channels: is 1 channel left, 1 channel right and the third channel is a center channel? If so, I imagine you’d want to combine the left and right signals for the center channel.

But I am still worry about power. Its mention that 50 W RMS 4 ohms by channel. But is this real rms or ''Chinese" Watt?

50 watts is at 10% distortion using a 24 vdc power supply. That’s best case and the most you can expect. I personally don’t think you need to worry about power given the speakers involved.

Since my small speakers is 4 ohm and I want to connect them 2 in parallel to one channel, I hope this amplifier will work with 2 ohm otherwise I will use resistor to compensate resistance (to get 4 ohms). What do you think about this?

Connect them in series for an 8 ohm load. The amp will put out 25W max instead of 50W, but your resistor would consume 25 of the 50W too. I believe this will still be more than enough for your sound bar speakers. And you won’t have a resistor burning off half of the amp’s output.

Actually I asked about Sure programmer...It contains also EEPROM, is not it?. can I save my settings from sigma in EEPROM and use this like standalone system? in this case sure programmer should be connected every time? ...Or EEPROM is on JAB3 board?
The code in the EEPROM on the ICP1 is what makes it function as a programmer. You write the SigmaStudio program to EEPROM on the amplifier board.

So if I understood I can use limiters in Sigma studio, but how to control volume for both boards? I did not understand is it possible? Or only way is to change the volume of the input aux signal? (or Bluetooth source?) Because sure told me that I can use that potentiometer in cable kit for both boards... ?

Look at Sure’s SigmaStudio program I mentioned before. It uses the four on-board potentiometers to control both volume and filter settings. For example, the potentiometer connected to AUX_ADC_1 controls the volume for two channels using SW Vol 1_3. This is a SigmaStudio “Single SW slew volume (adjustable)” volume control. The basic control is for one channel. Right click on it, select “Grow Algorithm” and you can select how many channels it will control. So one volume control (and one potentiometer) can control all 4 channels.

What do you mean that JAB2 is not stereo? ?
Yes, JAB2 is stereo. I just wanted to point out that if you connect a subwoofer to JAB2 you can’t use it as a single channel amplifier.
 
And do you need the jab2 ? (I have no need for Bluetooth, is there some way to use another amp for the last 2 channels)

I used the J5 connector on the JAB3 to connect to a second amplifier. I attached the pin out for the connector. You need to connect the LOUT (pin 2), ROUT (pin 5) and one or both of the GND (pins 3 or 4) to the second amplifier. I did it with an external TPA3116 based amplifier. You'll get volume differences between the two amps if the gain setting on the second amp is different from the JAB3 amp. Not sure what the gain is on the JAB3 amp.

You could use Sure's "Extension Board" for your signal input. You could also use the J5 LIN (pin 1), RIN (pin 6) and GND pins for input instead if you don't want to use the Extension Board. If you use this method you could add an inexpensive preamp for volume control. I used one like THIS because it can use the amp's power supply.
 

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Wow, that's a lot questions! I’m not surprised no one has replied...
Finally all the boards came to me and I can start the project.

But today I am faced with the following problems:

- First of all could you please tell me is it possible to use adaptive filters inside 1701. Or can it be implemented by combining some tools of 1701?

I need to use different filters (different shape) depending on the level of signal. For example at low lewel loudspeaker does not have distortion at 40 Hz ( linear behavior) but when increasing the level of signal, at some moment, at 40 hz speaker reaches maximum excursion and gets distortions. So I need to create a filter with signal level dependance. Is it possible in Sigma studio?


- Secondly, Sure provides WONDOM SigmaStudio Programmer Board, but I can not find the drivers for this, So maybe you know where i can find the drivers for this. Because know I have ''unknown device'' (w7 64, Sigma studio 4).

on another computer it was possible to somehow install the driver and write the settings into the EEPROM of this programmer (but I can not find out how). Sometimes during self-loading, the subwoofer channel is cut off after 2-3 seconds (there may be a pulse in the subwoofer), but the second high-frequency channel is not turned off. What could be the problem?

- I also wanted to know, when selecting a self boot mode (switch on the board) is it possible to use sigma in real time (or it is not possible?)


- Is it possible to use potentiometer to adjust volume of jab2 and jab 3 and where I need to connect it?

Thank you so much!
 
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.. First of all could you please tell me is it possible to use adaptive filters inside 1701. Or can it be implemented by combining some tools of 1701?

This probably won't work, for several reasons.

First, the ADAU1701 is a stream-oriented processor rather than block-oriented. You can execute over 1000 instructions for each sample, which is fine for basic DSP operations such as filters, scaling, signal routing, etc, but it's hard to do operations that require evaluating blocks of data. The adaptive filter would need to look across many audio samples to adjust the filter parameters, and there isn't the storage in the ADAU1701 to do that efficiently.

Second, adaptive filters are tricky and best done with FIR filters. IIR filters can have settling issues if the coefficients are changed too "rapidly", so figuring out how to implement an adaptive filter with an IIR is nontrivial. But FIR filters take up a lot of resources--and the ADAU1701 is relatively "small" with only 1K of Program and Parameter RAM.

Third, the ADAU1701 could calculate a running RMS level and use that value to select the filter parameters, but it would require a lot of look-up tables to implement that capability, to translate RMS calculations to filter coefficients. That could probably be done, but tables are implemented in Parameter RAM, and there just isn't a whole lot of Parameter RAM to work with in this chip.

And finally, the ADAU1701 has very limited branching capability and implementing conditional processing is clumsy. It's not designed to be used for procedural coding, and there will probably be a need for some "if-then-else" type of logic in a well implemented adaptive filter.

A better approach would be to implement the block-level processing in a general-purpose microprocessor, and have that processor load the parameters for the filters in the ADAU1701. But an even better approach would be to implement all of the adaptive filter in the general-purpose processor. Processor boards like the Pi and many others provide plenty of speed and floating point support, and would have no trouble implementing all of the adaptive filter functionality. I wouldn't try to use the ADAU1701 for this application--it's the wrong tool for the job.
 
This probably won't work, for several reasons.
Thank you for explanation!

Have you already used JAB3 and Sure's programmer for it?

I don't understand.
EEPROM memory is inside JAB3 or programmer board?

So if I want to use self boot with predefined filters, programmer should be connected to JAB3?

Potentiometers on JAB3 (filters LP, HP...) connected directly to DSP chip? or its basic configuration which written in EEPROM?

Because I confused, I wrote configuration to EEPROM from Sigma, everything was ok, then I cleared EEPROM and now JAB3 does not work like amplifier without DSP and Potentiometers don't change the situation
 
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I've used the JAB3 but not the programmer. I use a microprocessor to control the DSP. The code waits for the ADAU1701 to self-boot with whatever is in the EEPROM, and then the micro overwrites the Program RAM and Parameter RAM with my SigmaStudio code. That flow is documented in the articles at Audiodevelopers Reborn – A collaborative website for active speaker design.

Once the SigmaStudio is written to the EEPROM, you do not need to connect a programmer, unless you want to make changes in real-time. The ADAU1701 will boot up from the code in the JAB30 EEPROM (U4) and run your DSP code without the programmer.

if you clear the EEPROM, the ADAU1701 will execute 1024 "no-op" instructions for each sample, so you won't have any DSP functionality.
 
if you clear the EEPROM, the ADAU1701 will execute 1024 "no-op" instructions for each sample, so you won't have any DSP functionality.

Thank you! I thought differently and was confused

So firstly ADAU1701 checks what in the EEPROM and self boots from it, but then if you connect programmer and PC with Sigma, ADAU1701 will be operated by PC, right?

As you know JAB3 has potentiometers on board (implementation of changing filter's shapes) and I really do not understand if ADAU1701 loaded from EEPROM with your Sigma settings what this potentiometers will change or how they will affect?

Or from the factory the manufacturer records his program on EEPROM to make its possible to use these potentiometers? It turns out when EEPROM is rewritten, then potentiometers will never work again, right? I am confused with this. Because I do not see the point if potentiometers work always and simultaneously with self boot from EEPROM.

Thank you!
 
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The ADAU1701 has 4 "Auxiliary ADC" inputs that can be used to read voltage levels from potentiometers, and some of the SigmaStudio processing blocks can be controlled by these inputs. Someone a while back had posted the code that Sure uses for these amps. Find that code and open it up with SigmaStudio and you will see how the potentiometers are used to control the volume and filters. Then program that code into the EEPROM on the JAB30 and your potentiometers will work once again.

Here is the link: http://www.sure-electronics.net/DSP1701_AMP2_Project Source File.rar

The self-boot process is documented in the ADAU1701 data sheet. You should probably keep re-reading that document until it makes sense.
 
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I know that the I2C lines on the JAB3 are protected with diodes, because the diode pack (U01) failed on one of my boards. It probably failed due to a voltage spike on either SDA or SCL. Also, I know that the ADAU1701 can fail if you apply voltages greater than 3.3V on the SDA or SCL pins, because I've done that before, trying to program it from a 5V Arduino CPU (different board).

I've never used the programmer, and I don't know whether there is a 5V/3.3V switch or jumper, but check for that first. Also, I know you need to be careful about connecting or disconnecting I2C devices with power on, because you can get voltage spikes that can cause damage. I think that's what caused the diodes on mine to fail.
 
Hello,
I am currently working with JAB3, and it is a great board, but it is very hard to find any information...
So, in order to program the JAB3 with ICP1 you must follow this procedure :
1 - Plug ICP1 to your computer, the programmer should turn green in Sigma Studio.
2 - Plug JAB3 to DC, and give it a entry signal (some music from your phone...)
3 - Plug JAB3 to ICP1
4 - short circuit SW1 with a jumper cable



It should work...

I managed to do it one time, but since then I was unable to do it again...

It woks with arduino
 
Ok, I played a bit with Sure Wondom JAB3, and it was quite difficult to find any information on those boards, so I'll try to share here what I finally found. Most of those informations are already existing, I'm just trying to compile them in one post.
This thread was the most informative I found, thanks to ernperkins answers, so I choose to post it here.

I'm French, so please excuse my English.


I Bought a JAB3 1100 witch is a mono version.
My goal was to create a PA speaker working on battery, using the JAB3 for high frequency, and a Sure T-Amp (1x500W) for low frequency (and i finally managed it to work).
The goal was to use the ADAU1701 on the JAB3 to make an active filter, programming it with the Wondom ICP1. My first ICP1 was fried during first use, so I had to exchange it.



How to use ICP1 with Simgma studio and JAB3 :
You need to plug ICP1 to your computer before plugging it to JAB3 (it turns green in sigma studio).
You need to power JAB3, and give it an audio source before plugging it to ICP1.




Debugging :

If you are just testing your program, you can use "link compile and download" in sigma studio. You don't have to short SW1. The program will be loaded to JAB3 and you will be able to test it directly. The board program will remain unchanged when you reset it, so it's an "idiot-proof" way to test your design.
You can use level detectors and see the signal levels directly in sigma studio, and use volume control to tune it.
You will find lot of informations on how to program with Sigma Studio here.
You can inspire from the example from Sure : HERE (TY again ernperkins).


Long term programming :
The steps are detailed in the video given by ernperkins.

If you need to long term program your board, you have to short SW1 and write the EEPROM using the method given in the video (right click on adau1701, write last compilation to eeprom).

So to resume :
1 - Plug ICP1 to your computer, the programmer should turn green in Sigma Studio.
2 - Plug JAB3 to DC, and give it a entry signal (some music from your phone...)
3 - Plug JAB3 to ICP1
4 - short circuit SW1 with a jumper cable

- keep the jab3 SW1 shorted while programming the EEPROM
- when finished writing the EEPROM: remove SW1 short and reset the JAB3


I don't know why, but if you don't unplug ICP1 from JAB3 before reseting it, the ICP1 will not work any-more, refusing to upload again to eeprom. When this happen, I turn my computer in "stanby mode" for a few seconds, I restart Sigma Studio, I reset JAB3, re-plug it to ICP1 and it works back.


External volume control :
I needed an external volume control for my active speaker...
I first used the potentiometers that are on the board.
You can inspire from the Sure example.
The important things to know :
- the 4 pots are plugged to auxiliary ADC ADC0, ADC1, ADC2 and ADC3.
ADC0 : MP9
ADC3 : MP8
ADC1 : MP2

ADC2 : MP3


- you need to configure them in the "hardware configuration / register control"

turn them to ADC
Don't forget to check the box on the right (I can't see the right column it in my Sigma Studio, I have to slide...)
An externally hosted image should be here but it was not working when we last tested it.



You will find lot of informations on ADAU1701 inputs (and lot more) on this document.


Since the potentiometers are soldered to the four auxiliary ADC input, you have to de-solder them from the board if you want to use an external pot. Since I didn't want to make some soldering on the board, I finally chose to use a rotary encoder.



Rotary encoder :

I used a keyes rotary encoder module. It works OK, even if it don't 100% comply to analog devices application note.
Don't forget to check the pins used in the input section of hardware configuration.


That all for today, I'll try to complete this post when I could...
 
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