freeDSP-aurora - DSP with 8 I/Os, USB Audio, S/P-DIF, ADAT, Bluetooth and Wifi contro

[dspverden]

Hello,

I would like to introduce my newest design of the freeDSP family: freeDSP-aurora. It was already introduced in the freeDSP main thread, but there the idea came up to open a new thread only for freeDSP-aurora.

freeDSP-aurora is a one board DSP solution. It all started with the development of a DSP powered subwoofer amplifier. Once the prototype of this amplifier was presented on the German forum diy-hifi.forum.de a fruitful discussion about additional features and ideas started. After two years of development freeDSP-aurora is the distillation of the best ideas.
So here are the features of the board:

• Analog Devices ADAU1452, 294.912 MHz, 32-bit SigmaDSP, 6144 SIMD instructions per sample @ 48kHz fs, 40kWords of data RAM, 800ms digital audio delay pool @ 48kHz fs, eight stereo ASRCs with 139dB DNR
• XMOS XE216-512-TQ128 for multichannel bidirectional audio streaming
• ESP32 for WiFi or Bluetooth control
• AKM AK4458 32bit-DAC
• AKM AK5558 32bit-ADC
• Supporting sample rates between 44.1kHz and 192kHz
• 8 analog balanced input channels, +6dBu
• 8 analog balanced output channels, +6dBu
• S/P-DIF input and output
• ADAT input and output
• Wordclock input and output
• Support for display, rotary encoder, volume potentiometer, temperature sensor, PWM controlled fan, IR sensor
• One freeDSP expansion header
• USB Audio Class 2 Bidirectional streaming with 8 channels in and 8 channels out, full-duplex. Works with ASIO driver under Windows 10 and driverless under macOS and Linux.
• Board dimensions: 100mm x 100mm

To make using as easy as possible I wrote my own app that controls the DSP via Bluetooth. The app runs on macOS, Windows or iOS. Thanks to the ESP32 framework Wifi instead of Bluetooth is an option.
The features of the DSP depend on the DSP firmware. Therefore, I have made a first DSP firmware with the following features per channel.

• Input select
• High pass up to fourth order (Bessel, Butterworth, Linkwitz-Riley)
• Low shelving
• 10 parametric EQs
• High shelving
• Low pass up to fourth order (Bessel, Butterworth, Linkwitz-Riley)
• Frequency depending phase shift (allpass)
• Phase inversion
• Delay 100ms
• Gain

It supports 8 channels in and out.
This feature list is only a first step. It can be extended in the future, e.g. FIR-Filters are already on the ToDo list.

Like any other freeDSP the project is licensed under an open source license. Thus, everybody is invited to modify or contribute to this project. You can even download everything and build as many freeDSPs as you want. You can find everything on freeDSP | An Open-Source Low-Budget Audio DSP and in the git repository GitHub - freeDSP/freeDSP-aurora: freeDSP ADAU1452 with 8 analog input, 8 analog outputs, S/P-DIF I/O, ADAT I/O, USB Audio Class2, WiFi, Bluetooth

Well, for freeDSP-aurora a new idea came up: A kickstarter campaign has been launched. This campaign shall help to make boards available for those who do not want to order empty PCBs and solder the SMD stuff and do all the firmware flashing. The kickstarter campaign can be found here:
freeDSP-aurora DSP by auverdion — Kickstarter

I am sure you will have a lot of questions because the board has so many features. Please feel free, to ask me!

Raphael

[CharlieLaub]

Wow, that seems pretty awesome!

Looking at your list of supported filter types, one thing I strongly suggest is making something like the MiniDSP "advanced biquiad" functionality - the user can input the five biquad coefficients as real numbers. This makes it possible to make other kinds of filters, like higpass-notch, and lowpass-notch, or anything that can be realized with a biquad.

The other thing to consider, based on my experience with the miniDSP interface, was that there were too few biquad stages available on the inputs before stages that were dedicated to crossover filters. The input filters are used for global EQ and you want them applied to all outputs without having to duplicate each filter in the output filter chain.

With the ADAU1452 you have a lot more processing power on tap compared to the ADAU1701 and the specs of the DSP are better. That should deliver a big improvement all around. Glad to see you are using the separate AK4458 DAC. Lots of promise there.

Have you built and measured a prototype for S/N, etc? Apart from the main PCB, what other PCBs will be available (e.g. power supply) and will there be a case/chassis available?

[Windforce85]

Great project! Would you like to add inverted all-pass for crossover phase shift correction? How does analog output stage, clock distribution and DAC Vref power supply looks like?

[Markw4]

Don't want to be a downer, but at one point I looked into possibly using ADAU1452 or even SHARC processors for some prospective applications. Decided they have more distortion than I would prefer for some things such as ASRC, when compared some other possible chip choices. If anyone else is like me in terms of listening preferences, I would suggest maybe taking a look at the specs of whatever you might consider using and see if you think it will satisfy all your requirements.

[googlyone]

Mark,
I am interested to hear more. Where in the docs are these levels of distortion you are referring to?

[Markw4]

Page 4 of the ADAU1452 data sheet says ASRC THD+N = -120dB. By way of comparison SRC4392 is hovers around -140dB. AK4137 claims somewhere between -133dBt and -153dB, depending, and with typical DNR around 184dB.

I fully understand that some people will be skeptical that anyone can hear any distortion at even -120dBFS under any conditions. I don't want to start up another debate on that subject, so i will leave it with an acknowledgement that it remains a controversial subject in the minds of many people who happen to be interested in audio.

[JayShoe]

I'm sure the quality if this chip is plenty fine for my digital mixer project. This chip is used in many commercial live audio products. It uses industry best practices to get the job done nicely. I'm looking forward to building my project with it.

I wanted to mention that it might be possible to swap an ADAU1466/ADAU1462 to get more memory. "1600 ms digital audio delay pool at 48 kHz".

[dspverden]

Originally Posted by CharlieLaub

Have you built and measured a prototype for S/N, etc? Apart from the main PCB, what other PCBs will be available (e.g. power supply) and will there be a case/chassis available?

Some final measurements of the prototype will be made next week.
For power supply you just need a +12V DC. You can use a wall mount AC/DC power supply or one of these SMPS modules. But of course you can build your own power supply. I have no chassis available yet. The discussion on the German forum has shown there is no clear favorite for a connector configuration. My idea is to design a few add-on boards for some use cases (e.g. loudspeaker management, active loudspeaker box). Then it would be possible to make a case design for e.g. a loudspeaker management solution.
I am still hoping that soon other DIY'ers will start to contribute add-ons (either hardware or software) to this project.

[dspverden]

Originally Posted by Markw4

Page 4 of the ADAU1452 data sheet says ASRC THD+N = -120dB. By way of comparison SRC4392 is hovers around -140dB. AK4137 claims somewhere between -133dBt and -153dB, depending, and with typical DNR around 184dB.

I fully understand your point. Well, you only need to think about this, if you use the internal ASRCs of the DSP. So far, I never used them in my DSP programs on SigmaDSPs. By turning off the ASRCs are not in the signal path.

[JayShoe]

My idea is to design a few add-on boards for some use cases (e.g. loudspeaker management, active loudspeaker box). Then it would be possible to make a case design for e.g. a loudspeaker management solution.
I am still hoping that soon other DIY'ers will start to contribute add-ons (either hardware or software) to this project.

Does this circuit require a mic preamp? Or can the input stage be modified to add additional gain?

How about a preamp like this, but instead of xlr, it would have the analog i/o pinout as seen on the Aurora... A circuit like this would be between the port/jack. We would create chasis mount pcb->Aurora io separately.



Chasis mount xlr> Aurora io > mic preamp > Aurora io > Aurora.

I can help getting some parts into the kicad library, and I can help create some simple conversion boards for different io. What's your thoughts?