Not what you are asking because this is not a modular platform but AK5558VN is available on Digikey as well as the 6-channel AK4456VN DAC which is pin compatible with AK4458VN (that 8ch DAC will be available in the end of February next year).
You could pick any S/PDIF DAC or ADC from Aliexpress, Ebay or elsewhere and tap into SPDIF TX and RX pins on the headers, but those are just 2 channels in / out.
Wouldn't it be possible to use an external summing and filtering stage on separate PCB if you just left the IC's out and replaced the appropriate resistors with 0 ohms ones (maybe leave just the first passive RC on the circuit) and bridged the pins 1 and 3 (and 2 in between) of the filtering stage IC's to get passive outputs (differential only)?
Regarding the distortion wouldn't it help if you used lower gain setting resistors in the filter stage replacing the 3k40 ones with something like 2.78 kohms?
The problem, aside from the parts shortage, is the non existing user interface for the ADAU DSP. The use of Sigma Studio is not an interface most expect, but a bare tool to program some function blocks together. That makes it uncomfortable, but extreme flexible and mighty. The ADAU was not made as an end user configurable DSP, but to work inside commercial products like sound bars, active speaker and the like. The users options are usually reduced to the potentiometers you can find at a Sure/ Wondom DSP for example. Which is a sub 20 US$ value part.
For the majority of users, some ADAU solution, combined with an USB connection to a software to adjust the parameter needed, is the way to go. The first one being the Mini DSP, which today is the worst one to buy and really overpriced. Not even woth half the price IMO. It still had the option to import different profiles.
The fixed function solutions offered by Dayton Audio are a step better.
Today some newer products do all you want: Comfortable change of parameter, predictable functioning and low price, starting at around 100$.
So the very good and mighty "freedsp", which is much better than the commercial versions, will stay a minority product.
For the majority of users, some ADAU solution, combined with an USB connection to a software to adjust the parameter needed, is the way to go. The first one being the Mini DSP, which today is the worst one to buy and really overpriced. Not even woth half the price IMO. It still had the option to import different profiles.
The fixed function solutions offered by Dayton Audio are a step better.
Today some newer products do all you want: Comfortable change of parameter, predictable functioning and low price, starting at around 100$.
So the very good and mighty "freedsp", which is much better than the commercial versions, will stay a minority product.
Depends on your location. In Europe the https://www.thomann.de/de/the_t.racks_dsp_4x4_mini.htm is a very good offer. Steel/ aluminum case, balanced in and out, power supply included. Good software. And ADAU 1701 can not get much better than that if you want to avoid Sigma Studio and complicated interfaces.
Then there are many others around, from Dayton for example. Just google 4 channel DSP for a start.
In the end it boils down to the software. If you just want to do a single project, avoid Sigma Studio if you do not like to spend a whole week learning something new.
For some set-up that has to be flexible, a simple user interface is a must. You will not change your DSP's parameters on the fly with Sigma Studio, while an audience is listening or just if movie needs more bass. Such needs a simple USB based, real time user interface.
If you want to build something like a bunch of active speakers, that are adjusted once and never re-programmed again, a simple ADAU like the Sure/ Wondom thing is the cheap way to go, at around 20$ a piece. You do not spent much money, even if you dedicate one DSB to each cabinet. Sonic performance will be just the same. After installatin you are limited to the 4 pot's for some simple adjustments, if you program them to do so. Any deeper influence need connection to the programing interface and Sigma Studio (again). So something for the development lab, not end user.
I got both and the time getting the Thomann DSP to a first sound output needs a fraction of the time you need to even install the software and make a first connection with the ICP programmer. Maybe the Tiny Sine is faster to set up than mine.
To make it short, if you feel waisting time with Sigma Studio, get a Dayton or Thomann DSP. Dayton has understood this limitation and developed a version which needs a programmer, but has a simple user interface. You get it in some of the Dayton D-amp boards. Look for KPX Bluetooth programmer.
Then there are many others around, from Dayton for example. Just google 4 channel DSP for a start.
In the end it boils down to the software. If you just want to do a single project, avoid Sigma Studio if you do not like to spend a whole week learning something new.
For some set-up that has to be flexible, a simple user interface is a must. You will not change your DSP's parameters on the fly with Sigma Studio, while an audience is listening or just if movie needs more bass. Such needs a simple USB based, real time user interface.
If you want to build something like a bunch of active speakers, that are adjusted once and never re-programmed again, a simple ADAU like the Sure/ Wondom thing is the cheap way to go, at around 20$ a piece. You do not spent much money, even if you dedicate one DSB to each cabinet. Sonic performance will be just the same. After installatin you are limited to the 4 pot's for some simple adjustments, if you program them to do so. Any deeper influence need connection to the programing interface and Sigma Studio (again). So something for the development lab, not end user.
I got both and the time getting the Thomann DSP to a first sound output needs a fraction of the time you need to even install the software and make a first connection with the ICP programmer. Maybe the Tiny Sine is faster to set up than mine.
To make it short, if you feel waisting time with Sigma Studio, get a Dayton or Thomann DSP. Dayton has understood this limitation and developed a version which needs a programmer, but has a simple user interface. You get it in some of the Dayton D-amp boards. Look for KPX Bluetooth programmer.
Yes, but you can't get an Aurora. This is a slightly limiting point of it's performance.
See, Aurora is the better thing if you look at it as a technician, but for the end user to hard to get up and running.
It is a minority project, with a very positive educational background.
You don't learn a fub if you buy a Dayton or T.Racks.
I have just meet to many people that search a solution for a DSP and do not understand that Sigma Studio is not the user interface they expect.
Most don't want to dive that deep into componet search, programming interfaces and stuff. Some don't have the time, too.
See, Aurora is the better thing if you look at it as a technician, but for the end user to hard to get up and running.
It is a minority project, with a very positive educational background.
You don't learn a fub if you buy a Dayton or T.Racks.
I have just meet to many people that search a solution for a DSP and do not understand that Sigma Studio is not the user interface they expect.
Most don't want to dive that deep into componet search, programming interfaces and stuff. Some don't have the time, too.
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That does look excellent actually! The omnitronic 2x4 dsp also looks interesting. I'm in Australia so postage from Europe is a frequent problem.
I'm rather hopeful that the Octavia freedsp project from CyberPit can take us forward, though I do accept what you're saying about freedsp perhaps not reaching the mainstream. That's OK with me to be honest, as long there's a small core community I don't mind investing in learning a new interface (we shall see how I go with the tinysine however 🤣)
If you know what you need and have some experience with basics of such tools like Sigma Studio, you should have no problem. You can compare it to a programmable injection and ignition system for a gasoline engine. Many options that, used right, in the end give a working system.
Install SS (need a Win 64 for it) and load some simple application, like a 2.1 x-over to see how such a chain looks. Learn how to use the different function blocks and what you can do with them.
You change values in SS and load it up to the DSP, then test the result. This is much more complicated, time consuming and error prone than a simple graphic software, but works just the same.
The 4 pots are quite universal, but a little complicated to change in function. You can move them from the DSP to the outside of an amp or active speaker and even change filters with them. A switch with different resistor values should work too. You see, a lot of options to explore. Very mighty.
So a lot to screw up. There are lots of information, tutorials and a wiki out there. Take care to use the right version of the software, this can get complicated with the interface driver. If you have contact to the DSP and can read and write to it, it is best is to draw a plan what you want to do to the signal. Then pick your functions and arrange them. Last, keep in mind that you have to take care with levels, if you don't you may loose a lot of resolution from analog to digital and back.
explains it
Install SS (need a Win 64 for it) and load some simple application, like a 2.1 x-over to see how such a chain looks. Learn how to use the different function blocks and what you can do with them.
You change values in SS and load it up to the DSP, then test the result. This is much more complicated, time consuming and error prone than a simple graphic software, but works just the same.
The 4 pots are quite universal, but a little complicated to change in function. You can move them from the DSP to the outside of an amp or active speaker and even change filters with them. A switch with different resistor values should work too. You see, a lot of options to explore. Very mighty.
So a lot to screw up. There are lots of information, tutorials and a wiki out there. Take care to use the right version of the software, this can get complicated with the interface driver. If you have contact to the DSP and can read and write to it, it is best is to draw a plan what you want to do to the signal. Then pick your functions and arrange them. Last, keep in mind that you have to take care with levels, if you don't you may loose a lot of resolution from analog to digital and back.
do you plan to share the mods done to Aurora to get this specs, perhaps? I'm working on freeDSP-Infinitas (almost the same architecture) and I'm wondering if I could integrate those mods there too
thanks!
Hi tomik,
I got the exact same error message. I havent't worked with XMOS before.
Could you please give me a hint where to delete the described declarations (in which file(s))?
To be honest I cannot remember anymore 
Haven't used the tool since. But you could search your project folder for the 'quad_spi' so maybe you can find files with those declarations. I tried to do that, but it seems I have really deleted those instead of commenting them out, so no luck...
Edit: found it on my other computer. It is in the xk-audio-216-mc.xn file, under ExternalDevices tag
Haven't used the tool since. But you could search your project folder for the 'quad_spi' so maybe you can find files with those declarations. I tried to do that, but it seems I have really deleted those instead of commenting them out, so no luck...Edit: found it on my other computer. It is in the xk-audio-216-mc.xn file, under ExternalDevices tag
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I am building a box with an Aurora board in it, with a low-voltage SMPS feeding it, and the box also has several amplifier boards in it, which are fed from a higher-voltage (single rail) SMPS. The higher-voltage SMPS is also in the same box.
I am using a USB cable for the Aurora board that is wired permanently in/out of the chassis (i.e. feed-through style, with no connectors at the chassis entry). The source end is factory-terminated, with shielding properly connected to cable shell and ground wires (probably). The device end is terminated to a 5-pin plug that matches the USB header on the Aurora, with the shield intact as close as possible to the individual wire terminations. This means that the ground wire(s) and the shield are directly connected to the Aurora. The Aurora's 0V power input (and also the 0V output from the LV SMPS) is then connected to the box's safety earth through a ground loop breaker. The AC power input to the LV SMPS is ungrounded (only L/N available) while the AC power input to the HV SMPS is grounded to safety earth (L/N/G) without GLB. The 0V output from the HV SMPS is also tied to the box's safety earth through the GLB.
I believe that the above installation would always result in a ground loop flowing from the PC USB end through the USB ground/shield into the Aurora, and then to the following:
1) to box safety earth through the ground loop breaker (so, probably insignificant) to PC safety earth
2) to all signal grounds leaving the Aurora, since it looks like USB ground, power "ground", and audio "ground" are tied together on the board, unless the downstream devices are fully isolated from any ground and not capacitively-coupled or whatever.
If we ignore any EMI/RFI effects (haha) from the USB cable entering through the chassis toward the Aurora board, I believe - and could be wrong - that I will not suffer from hum/buzz/noise as long as I only use the analog balanced outputs. Does this make sense? Because if I use analog single-ended, I would have a ground loop from the USB source end to a single-ended amp via RCA shield.
Taking that further, does it not mean that when using a USB connection (that does not have a proper galvanic isolator on it), that the only way to get a loop-agnostic setup is to use balanced analog connections, or single-ended if and only if the downstream side is fully isolated from ground? Even if the shield were to be separated from the USB wires, the USB ground wire(s) is inevitably connected at both ends... unless even the USB grounds can be lifted and only data wires connected at device end? This does not seem like common practice as I have heard that the exposed wires can radiate quite a lot of RFI. Which might be fine in a metallic enclosure, but not so much if there are devices inside that same enclosure that are sensitive to interference from USB.
I am using a USB cable for the Aurora board that is wired permanently in/out of the chassis (i.e. feed-through style, with no connectors at the chassis entry). The source end is factory-terminated, with shielding properly connected to cable shell and ground wires (probably). The device end is terminated to a 5-pin plug that matches the USB header on the Aurora, with the shield intact as close as possible to the individual wire terminations. This means that the ground wire(s) and the shield are directly connected to the Aurora. The Aurora's 0V power input (and also the 0V output from the LV SMPS) is then connected to the box's safety earth through a ground loop breaker. The AC power input to the LV SMPS is ungrounded (only L/N available) while the AC power input to the HV SMPS is grounded to safety earth (L/N/G) without GLB. The 0V output from the HV SMPS is also tied to the box's safety earth through the GLB.
I believe that the above installation would always result in a ground loop flowing from the PC USB end through the USB ground/shield into the Aurora, and then to the following:
1) to box safety earth through the ground loop breaker (so, probably insignificant) to PC safety earth
2) to all signal grounds leaving the Aurora, since it looks like USB ground, power "ground", and audio "ground" are tied together on the board, unless the downstream devices are fully isolated from any ground and not capacitively-coupled or whatever.
If we ignore any EMI/RFI effects (haha) from the USB cable entering through the chassis toward the Aurora board, I believe - and could be wrong - that I will not suffer from hum/buzz/noise as long as I only use the analog balanced outputs. Does this make sense? Because if I use analog single-ended, I would have a ground loop from the USB source end to a single-ended amp via RCA shield.
Taking that further, does it not mean that when using a USB connection (that does not have a proper galvanic isolator on it), that the only way to get a loop-agnostic setup is to use balanced analog connections, or single-ended if and only if the downstream side is fully isolated from ground? Even if the shield were to be separated from the USB wires, the USB ground wire(s) is inevitably connected at both ends... unless even the USB grounds can be lifted and only data wires connected at device end? This does not seem like common practice as I have heard that the exposed wires can radiate quite a lot of RFI. Which might be fine in a metallic enclosure, but not so much if there are devices inside that same enclosure that are sensitive to interference from USB.
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It worked. Thank you a lot.To be honest I cannot remember anymore
Edit: found it on my other computer. It is in the xk-audio-216-mc.xn file, under ExternalDevices tag
