Multichannel Audio DSP Field Mixer Recorder

bluetooth app controlled professional portable DSP mixer.balanced audio IO,phantom power,flexible routing,ISO recording. timcode. AVB audio.

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Multichannel 8 channel + Professional Field RecorderDSP mix routing and FX.Backpackable!100% app controlled. website for app and firmware updates ( eventually )

Android and IOS App used to monitor audio levels and mixer settings. Rig up a car , set up the mixer settings and hit record.....Maybe you never see your mixer again because your working on a documentary with suspect characters.. but , it didn't cost as much as you main mixer and life goes on ;)

record your bands live gig and set the mix levels on your phone in the sound check ?

use it with your DAW in your home studio

design your own custom analogue cards and plug them in !

small , low power , lightweight , customizable , low cost

Future bluetooth hardware knob controller.



8 channels AES3

AVB gigabit ethernet audio


USB C PC/MAC DAW connectivity

TXCO RTC for accurate time of day ( Timecode )

Individual Timecode IO , JamSync

Dual RF modules for digital audio link to camera ( or possible timecode to sync boxes )


isolated digi opto power switching for the preamps ( power saving )

8 balanced 10 - 65db preamp cards on 1.27mm headers ( phantom power and 0db line level ) 

4 balanced line drivers cards

2 x very high quality headphone cards.

preamp front end limiters

Voltage monitoring

started printing 3d cases to try and get a design worth making out of aluminium or carbon fibre.

experimenting with changeable connector plates.

It's Alive !!! here's the 1st test recording for historic reasons :)

here's an old realtime audio meter input demo. was another mile stone!

The bluetooth App does a lot more now also ! FX compressors LP HP filters etc.


The beginning , learning how to use SPI and I2C. I believe this ever worked :) 

  • 1 × ADAU1467 Semiconductors and Integrated Circuits / Misc. Semiconductors and Integrated Circuits
  • 1 × ARM STM32H7xxxx
  • 1 × Fan out Buffer
  • 1 × 12.288mhz oscilator
  • 1 × CS4385 Audio ICs / Audio Digital to Analog Converters (DACs)

View all 8 components

  • preamp / line driver and HP cards in with +/-9v opa1637 buffers biased from VQ on ADC and DAC

    ben biles10/06/2021 at 13:38 0 comments

    4 preamp cards and 4 balanced line drivers using OPA1637.

    The Vcom and higher +/-9v rails for OPA1637 buffer are supplied through the stackable headers on the preamp cards. They are 1mm to long so not sure I will be able to squeeze all 8 preamps in there using those. 

    The 6mm brass standoffs worked fine for the line drivers and I send the VQ down one line of brass standoffs and it works. 

  • more work on the chassis / box

    ben biles09/26/2021 at 12:18 0 comments

    I've started work on the case again and will print another in CF and will glue on some thin aluminium panels in recesses in the CF.

    I am making plates for connectors that bolt to the chassis. so far only 2 plates , 1 for the mini XLR inputs / outputs and 1 panel for the AES IO mini XLR and 1/4" headphone sockets.

    I made some slots in the digital and analogue main board mounts so I can slide in some EMI shielding between the 2 boards.

    this is a very rough 3d thing of the assembly. you can click to make a cutaway.

  • more work on the chassis / box

    ben biles09/26/2021 at 12:18 0 comments

    I've started work on the case again and will print another in CF and add some aluminium panels.

    I am making plates for connectors that bolt to the chassis. so far only 2 plates , 1 for the mini XLR inputs / outputs and 1 panel for the AES IO mini XLR and 1/4" headphone sockets.

    I have no idea if this 3d grab cad page will embed but its a rough idea!

  • buffer fail ! solution in video

    ben biles06/21/2021 at 03:31 0 comments

    hopefully final video update. The filter is for half mclk frequency of 12.288 ( 6.144mhz ) 

    some interesting notes on buffer filtering for ADC here..

    I was looking at frequency ranges just above audio but the filters job is to cutoff anything from 6mhz and higher ! 

    update simulation using opa1637 as the preamp buffer to ADC.

    better results in simulation at least.

    Why I can't use the OPA1632 to attenuate the PG2500 with +/-5v power rails

    OPA1602 attenuation and low 6mhz low pass filter

  • one part case & DAC headphone buffer / filter design

    ben biles05/14/2021 at 03:46 0 comments

    The main case is just one part with top and bottom access panels.

    I'm making connector panels that can be changed so if somebody wants for instance 4 x full XLR

    and a DSUB for the rest of the channels its just 4 bolts and an FPC connector to change it! 

    I will make a panel for the AES sockets next. 

    There is a recess for the oled confidence screen and Time Code clock battery. 

    Also a recess for BT module and programmer connector.

    I'll need to learn how to make a nice sliding panel for the battery change as I hate swing out doors that just snap off like a lot of sony pro cameras etc.

    Later down the line when I'm happy with its functionality and I work out better waterproofing I will work on beveling the edges and work on the horrible sharp corners! The bottom edges have a routed channel for a waterproof strip the will mate with the bottom panel to form a water seal.

    The preamps are working great with the new buffered VQ ( vcom ) going down a bolt that holds them in place.

    I can't say the same for the headphone preamp at all !! 

    I had not really thought through the bias injection to the buffer properly ( tried to inject it into each input leg of the negative rail which has the negative feedback loop. the positive was going to GND  ) wrong. 

    ignore the note about 100uF decoupling ! 10uF c1,c2,c3,c4 should be enough to remove the DC bias :)

    Now I need to incorporate this 50khz corner frequency filter in the negative feedback loop of my 4 channel OP4A134 or OPA1604 if this basic version works.

    from a cirrus logic app note : 

    A two-pole Butterworth with a corner
    frequency of 50 kHz attenuates the signal at
    20 kHz by approximately 0.1 dB and has nearly
    ideal phase linearity within the audio band

    As I type this I realise it may have been more sensible to breadboard the buffer with the filter and test it 1st before sending off for the pcb.. 

    I can test the filter with a sweep  generated by the DSP from say 15khz -> 100khz and make sure the filter is doing the job.

    I assume these ultra high frequency's can effect the lower ones harmonically if they are not removed? anyone know why i am actually making this filter ? it's just the recommended one in the datasheet !

    I connect the differential DAC audio signals to the non inverting inputs of the 4 channel op amp IC and bias it with the buffered VQ (VCOM) form the DAC. I think if I need some attenuation of the DACS outputs I'll need to use voltage dividers on the inputs as the non-inverting inputs since any non-inverting op amp input has a minimum gain of 1 ? in other words in this configuration I cannot use the buffer for attenuation.

    Please drop a message in the comments section if I am wrong there? thanks in advance :)

    EDIT:  I just realised I can set the output level of the DAC using I2C registers , so would never need to set attenuation anyway.

  • strong mounts and pcb mounts

    ben biles03/10/2021 at 11:21 0 comments

    I beefed up the upper lower case mounts where there was available space.

    Also added some pcb mounts that don't require spacers. 

  • 3d printed enclosure ver 2

    ben biles03/04/2021 at 05:02 0 comments

    Good bad and the ugly, 

    This will have to do while I work on more important updates with analouge boards.

    Will try carbon firbre nylon composite later. 

  • 3d printed enclosure

    ben biles02/25/2021 at 02:54 0 comments

    As you can see this is my 1st go with a 3d printer! It's actually my 2nd go at the lid. The 1st lid was completely wrong , always measure twice !

    my specially ordered miniXLR sockets with the nuts on the outside of the case are stupid. I realised the pcb on the back is worth 1$ so it's far better to put the mini xlr sockets in through the holes ( should be D shape 11mm ) then tighten the nuts / washers from the inside the case before soldering the pcb onto all the pins. need to add thread lock glue to those nuts I think or you would end up with a bunch of washers and nuts flapping about in between the sock and pcb eventually.

    cramped ugly nuts on the exterior of the case.. no ! got to go :)

    top right corner it a whole for the programmer cable.

    the uSD cards need more thumb room , it's a tweezer job to get them in and out !

    The whole case needs beveled edges , way too sharp everywhere!

    The bulge in the case is caused by the 40pin analogue board connector from ADC DAC card

    that protrudes over the edge of the board edges. easy mistake to forget that those IDC connectors are much wider than the box around the footprint on the PCB !

    for now I will just cut a whole in the case so it doesn't warp the box as I want to change the connector for light weight FPC ribbon cable , so its a problem that will eventually go away :)

    Well I guess this is the great thing about having a 3d printer , you just keep refining the case until your happy then get it made professionally. I didn't bother with vase mode or whatever is the best way to print. I used sun PLA+ which is slightly stronger than the usual PLA. But could well have used anything. 

  • preamp and line driver updates

    ben biles01/26/2021 at 16:19 3 comments

    time to do some work on the preamps and line driver cards now I have learned more ! 

    Originally I tried to avoid using buffers between the ADC and preamps and DAC and line drivers. 

    not really my best idea ! I think I avoided them since I just didn't really know what they did :)

    .. since then I have learned the PGA2500 vcom is in fact an input !!! pretty basic information but something I had completely overlooked.

    There is an advantage of having the PGA's audio signals swinging around 0V so I had connected VCOM to ground.

    BUT... the ADC has to swing around about 2.5v or near since it just used +5v. The exact bias voltage is available on VQ the pin.

    you can't just hook up VQ to VCOM on the the pga or ADC buffer driver ( 10uA maximum current sink for VQ )

    OK , I need a VQ buffer so I can send this bias reference across all 8 preamps.

    why not just add a buffer / proper filter as recommended in the data sheet and keep the PGA2500 VCOM connected to 0V with decoupling capacitors in between the buffer biased at VQ ( buffered )

    the buffered VQ bias ref voltage has to get around all the preamps !!

    Staring at the cards and backplane it seemed like VQ was going to make a massive trip all over the design and pick up who knows what EMI. 

    Ages ago I thought at some stage I will need to add a threaded bar right through all the preamps to stop them flapping about in the header sockets. then I though , just send the VQ voltage down that bar.

    then I briefly thought I had a clever idea , followed by another moment when I realised they have very likely been doing tricks like this since the days of vacuum tubes.. followed by another thought of , maybe the massive 2m threaded bar will pick up even more noise than if had just supplied the buffer with 2.5v from voltage dividers of the the 5V rail. The same 5V the ADC uses anyway.

    I don't know what's inside the ADC IC , how is VQ derived ? Its ridiculous to think it probably is just voltage dividers !   

  • back to where i was before the MC upgrade

    ben biles10/31/2020 at 10:59 0 comments

    finally pretty much back where i started before the stm32f7 to H7 upgrade.

    I still have an annoying bug in the BT app -> Micro where I need to press a button 3 times before it makes it to the app.. the sliders are probably only sending every 3rd value to. Its obviously a buffer or just the annoying fact I have no EOL character being recognised in the UART between stm32 and HC06 module.

    I'm sure there is an obvious fix. I'm using DMA and streaming out tons of meter value data on the same UART. 10 characters go one way and 3 come in for device control. I did try and align the buffer of TX and RX but it didn't help. It could also be the way the App Inventor code sends the characters!


    Fixed ! as soon as i finished writing this log , I tried disabling the FIFO buffer on the UART and bingo.

    one press on a button and instant reaction in the hardware. The sliders feel more accurate too since there not skipping values.

    It's working with an interrupt that happens when 4 character arrive in DMA. my incoming commands are a fixed 4 bytes long. the 1st byte is just a marker to check I am receiving a real command and not garbage.

    I don't send back any acknowledge byte yet to the app. 

    [end edit]

    The individual channel power switching is working. I will need to force a mute on the powered down preamp channels as there is some small noise there. I can do that in the ADC or the DSP , not sure which is better , possibly better to mute the relevant ADC channel so it's quiet when powered back on. Now there is a large thud ( boom ! )

    I don't really like the 1.27mm headers , pins can easily snap off and the connections are not as solid as 2.54mm. looking forward to the day I can just solder those preamp cards onto the backplane !  There is no way I would have got 8 preamp channels in such a small box with the larger headers so no choice ! I do wonder if I would have been better off using flat flex ribbon cables like in a mobile phone.

    I think the next job after fixing the BT uart will be ltc timecode IO which will come through the TI Codec on SAI3. I'm really looking forward to that from a C code perspective. Then right after that I have to wake up the RTC TXCO and figure out how to jam sync incoming timecode to it :)

    Way off into the future but hopefully before next year I will get the Xmos chip awake and get AES3 IO in and out of the DSP ! Hopefully I can setup TDM8 on the XMOS to sync up the the DSP as a slave without to much trouble. 

    AVB seams like pipe dreams now , but the gigabit ethernet PHY IC is there waiting and connected to XMOS.

    I doubt USB C sound card mode will work at the same time as AES3 and AVB but who knows ! that will be decided by XMOS computational power and my coding ability in the XMOS IDE i think. 

View all 92 project logs

  • 1
    Step 1


    STEP 2


    STEP 3

    3D PRINT BOX or buy CARBON FIBRE one from here

    STEP 4


    Hopefully I will eventually sell pre made modular boards of this mixer recorder.

    Users will be able to use pre existing firmware or write their own. 

View all instructions

Enjoy this project?



Chris Eloranta wrote 09/23/2021 at 23:14 point

So humbled and inspired by your insanely fantastic work, Ben.

I'm embarking on a project which is a spiritual subset of yours:  16ch of 24bit 48k low-latency ADC sent over a LAN along with OSC. 

Did you ever experiment with multiplexing/processing the two Cirrus chip TDM streams into a single 16ch bus/output/stream?  I scanned the project and saw it mentioned as a future goal, but I'm unclear whether it was ever implemented, and I must admit I've been stumped for a bit at that point.  (It's OK, I won't see my cirrus chips for half a year anyway due to the chip shortage; so there is time!)

Would I be correct in assuming that two Cirrus chips (CS4385) into a XMOS Core (Need to research which IC) would serve as the basis for outputting 16ch of ADC through AVB?  (I do realize that buffering circuits, clocking, power supply, microcontroller would all be involved in supporting roles as well...)  

Thanks so much for posting your project.  It seems to be the only thing out there even close to my target!

  Are you sure? yes | no

ben biles wrote 09/24/2021 at 00:22 point

Hi , your project sounds interesting. AVB to !! I am not sure you can delay by 8 32bit channels  / daisy chain for TDM16 mode with the CS5386 ( ADC ) . have a look in the datasheet and see if its possible. you might be better using  AKM AK5538 or similar. that supports daisy chain. I'm using the ADAU1467 DSP which supports multiple TDM8 or TDM16 IO. I took out the 2nd ADC from my project for now to free up some ports on the DSP.  later I will probably make a dual ADC card using TDM16 but just choose an ADC IC that supports daisy chain mode. I might also try doubling the ADC's for higher dynamic range. I'm using the Xmos X200 and I basically copied the ref design for the ethernet phy. I sounds like it should be possible to combine 2 x  CS5386 streams into the AVB stream in the Xmos200 code , but I would take a good look at the AVB reference design code 1st and try and understand the way xmos timesclice etc works 1st to see if it looks feasible. Sounds like it should be to me :)

  Are you sure? yes | no

Chris Eloranta wrote 09/24/2021 at 01:17 point

Sounds like I could use an ADAU1467 DSP like you did… 48in/48out.  Overkill for a digital bus combiner, but I’d imagine it would do the job?

Sadly, the AKM aren’t even available for backorder at this point. 

  Are you sure? yes | no

ben biles wrote 09/24/2021 at 03:50 point

Using the ADAU DSP just to combine some channels into XMOS would be a bit overkill. I would go onto the XMOS forum and ask if someone has already combined multiple TDM8 streams for AVB or other things. sounds like it should be totally doable ( possibly already in the AVB demo code ! )  ;) you can define GPIO on the xmos to control your ADC's and DAC's to so would be If quite simple hardware that way. If you want to use ADAU1467 and get stuck ask any questions you like , would be happy to help. The AD engineer forum is a great place for help to. 

  Are you sure? yes | no

FORART fe wrote 05/21/2021 at 11:43 point

Wow, black finish looks really great !

Anyway, as you know, I would really like to see a 1U rackmount version like this:

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FORART fe wrote 03/12/2021 at 08:17 point

Some years ago we've found this interesting project(s) idea(s) that could inspire further improvements:

(our backup:

As always, hope that helps/inspires !

  Are you sure? yes | no

FORART fe wrote 10/05/2020 at 08:30 point

As already suggested in the Public Chat, it would be great to adopt/collaborate with these IntRoLab's open hardware multichannel audio card projects:
(posted on thei GitHub too, of course:

Hope that inspires !

  Are you sure? yes | no

ben biles wrote 03/04/2021 at 23:39 point

ODAS is also a really great project from Inrolab

I built a 4 channel direction finder that uses odas :)

  Are you sure? yes | no

jackiredale wrote 03/23/2019 at 01:08 point

This project is amazing , I would love to have a go at making one myself , are you going to sell diy kits or the pcbs ?

  Are you sure? yes | no

ben biles wrote 04/08/2019 at 17:22 point

Hi jackiredale, sorry for the late reply. 

I was thinking of selling the 4 layer digital main board and ADC-DAC card modules with the firmware / android app already populated later when I've properly got everything working and tested. I don't really want to sell a board with problems :) so far I only found one error and there was an easy work around on this revision of the 4 layer digi main board but there are loads of gremlins on the other 10 or so module boards!

I am also working towards a final bluetooth only field recorder in a waterproof box that I want to give away to sound recordist friends that work in TV film etc. Idea being they could help me fine tune the thing and eventually get some kind of useful thing to sell ! no idea of the channel count or anything yet until I get the timecode / realtime metering working on the app. I got the test tone and test sweep 20hz -> 20khz working today :) pretty cool !

The sound quality is awsome from the preamps / headphone combination. better than my sound devices mixer. One thing though. I'm not using differential op amp buffer between ADC and PGA2500's ( just Panasonic FC DC blocking caps as in datasheet ) but this might not suite other preamp designs. usually you would have the op amp buffer to protect you ADC inputs from DC bias. So that might be annoying for people that want to add there own analogue hardware. 

on a plus side the ADC/DAC cards have split ground plane and there is really no noise coming across into the analogue from digital which I'm really happy about.

If I sell the mainboards I suppose I'll just do that here though hackaday but no idea how that works :)

have you seen this site ? I think they have some pretty cool DSP boards that are open source and one that works with rasbery pi ? that might also work as an audio recorder although i'm not sure.

  Are you sure? yes | no

Ulysse wrote 04/22/2017 at 16:16 point

I like. Of course. Music !

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Craig Hissett wrote 04/22/2017 at 15:56 point

This is great matey!

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Yann Guidon / YGDES wrote 04/22/2017 at 13:38 point

Impressive development !

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Francesco wrote 02/21/2017 at 13:44 point

Hi Ben, i have a question for you. I'm working on a similar project, taking a lot of inspiration from yours. You say you're controlling the ADAU1446 with arduino via i2c. I didn't find any specification about it, the datasheet only says "program it with sigmastudio" without giving any information about i2c via an external controller. Did you found some more information? Or do you have any tip for me, how to do it?

Many, many thanks... and again, your project is awesome!


  Are you sure? yes | no

ben biles wrote 02/21/2017 at 14:09 point Basic uC Integration Tutorial.pdf

You create files from sigmastudio when you have made your design. you need to add macros to SigmaStudioFW.h so that that the code can write / read to I2C depending on the platform you are using. So in arduino that would be  Wire.beginTransmission(44); // transmit to device #44 (0x2c)// device address is specified in datasheet
  Wire.write(val);             // sends value byte

  Wire.endTransmission();     // stop transmitting

sorry , I have no idea how hackaday does the formating , if I try and paste code into this box I get some rediculas things happening including user names poping up ?

I switched to intel edison recently and so I'm adding mraa I2C macros to SigmaStudioFW.h now since I need a micro that can record / playback TDM 8 channel audio. I could not find any way to record high quality multichannel audio in arduino. Not to say there is'nt a way. everyone points to teensy but I am interested in 24bit 96khz 8 channel audio etc..In fact I never added the macros for arduino athough I did get arduino & bluetooth controlling the preamp with SPI and of course controlled the DSP basic initialize power on to get the board working with I2C. I had some confussion with I2C in the begining but it turned out to be a faulty board ( heat damaged by myself ) . I have the board working in selfboot mode also now with eprom :)

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ben biles wrote 02/21/2017 at 14:12 point

Also I don't see your project on hackaday , can I take a look anywhere ? sounds cool ! what ADC / DAC 's are you using ? 

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Francesco wrote 02/21/2017 at 22:50 point

First of all, many many thanks for your answer, that's exactly what i was looking for and i simply missed it. Now everything is much more clear for me :)

My project is very similar to yours, i'm not interested in recording but in general my goal is to build a digital audio mixer. At the moment, it's only theoretical, i'm waiting for the components i ordered some weeks ago from china. My choices are an ADAU1442 (pretty much the same as your 1446) and your same ADC/DAC. I was interested in 8 ch TDM converters and this two are the most common. For example, the "behringher x32" mixer uses those converters as well, they are perfect for the purpose. But I use self puilt preamps, from old projects of analog mixers.
My project is different from yours because i will use rotary encoders, motorized faders, screens and so on.... like i said, the intention is to build my own digital mixing consolle. 
When i'll begin to build things i will create a project page somewhere :) Maybe i will find something usefull to you too :D
Again, thanks for your advices and your help! Will keep you updated.
- Francesco

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ben biles wrote 02/21/2017 at 23:28 point

interesting to know behringher x32 uses the same convertors. I would say that if your going for the highest dynamic range possible there are better ADC's. but I think you would need to use more chips and expense goes up dramitically. I also got my chips from china and managed to keep the cost down a bit. You will need to buffer your preamps with opamps to remove the 2.5v bias on the ADC or use AC coupling capacitors to remove ADC DC bias. There is an APP note from cirrus logic with 50khz corner freqnency filter and DC bias removal. I'm using PGA2500 so had to use AC coupling caps for increased dynamic range. PGA2500 only +-5v swing. anyway , if you get stuck your welcome to ask questions here. the shared clock and fanout buffer is working well for me. keep the clock line traces short and try to keep them equal leaghth. use split ground planes to sepperate analouge and digital ground return paths. I'm using I2C logic level translator as isolator to help keep noise away from preamp / analouge parts of the board. you could quite easily add 2 or more CS5386 ADC's by setting unique I2C addresses. ADAU1442 has 8 stereo asynchronous sample rate converters where the ADAU1446 does not. I did'nt need sample rate convertion and the adau1446 uses less power. same pinouts so you can change between them I think. I put ground pad on the pcb in case i wanted to use ADAU1442. I bought the cirrus logic programmer but I think there is a cheaper one out there.. might be an idea to build the programmer into your board? that way you could just plug your desk into sigmastudio and do realtime DSP from there software also. or just makes it look tidy. 

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Francesco wrote 02/22/2017 at 08:48 point

Cirrus logic programmer? you meant the analog devices one, right? I'm building the freeusbi programmer designed by, total cost is about 6$ so it's much, much cheaper. I'm really trying to keep the cost as low as possible. Build the programmer into the board? Yes, it's one of my goals, it would be a very good thing.

I will for sure follow your advices about the wiring, at the moment de "pcb design" part is far away, i will first try things in some "breadbord version". At the moment one  of my biggest concerns (like yours, i read) is if i will be able to solder the ICs, i'm not an expert with that kind of "small" soldering. Hopefully.......

  Are you sure? yes | no

ben biles wrote 02/22/2017 at 14:02 point

wooops !!! yer , I should say usbi programmer ! only 6$ thats pretty cool .. I'll build the programmer into my next board I think. I first used ic adapter boards to test things out , but to be honest I think you might be better just going for it and drawing a pcb. even if you make a mistake on the pcb you could run a bodge wire or 5 !! I managed to fit ADC DAC and DSP on one 10cm x 10cm board and when I drew that board I did'nt even know how to make ground planes !! just a 4 layer board routed badly and it works really well. 

you can get the ADC DAC and DSP working on adpater boards / bread boards, sort of. but it will be noisey and will probebly crash quite a bit. decoupling caps will be far away from pins on chip, ground plane will probebly be shared with digital etc.. you can download the Eval board guides for each IC and take a look at there schematics to get the idea of how to make the board.

I should'nt advise you to use naked DAC audio pins without buffering IC's, but I have to say I just hooked them up to my powered speekers for testing directly.. use some large 47uf 60WV electro AC coupling capacitors though if your going to do that since there is 2.5v bias ! don't blame me if you blow the IC's doing that also ;)

I found the ADC more tricky to get working than the DAC. theres a few traps in there , so ask me questions if you get stuck !! also I'm sure I could get ideas from your project to ! so would be great if you document it somewhere !

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ben biles wrote 06/05/2017 at 11:20 point

Hi Francesco , could you ask me the question about the DAC buffer stage again here? sorry , I tried looking for your comment somewhere in the logs and couldn't find it. In the mean time the output bias of the DAC is 2.5v which is incorrectly labeled VQ? instead of the normal Vcom ! you can AC couple the outputs to your buffer line driver or use an opamp.  you can bias the OpAamp inputs so that the audio swings around the 2.5v on the buffer input and outputs the audio swinging around 0v, essentially removing the bias and protecting you DAC from unwanted current etc

I'm using a OPA1632 balanced op-amp which outputs the differential channel swinging around 0v to an audio line driver. you could just use AC caps maybe but you can add some filtering to the DAC output if you need it with an opamp. I'm running the DAC at 24.576mhz rather 12.288mhz so filtering is'nt really necessary as far as I can tell but I thought its good protection for the DAC since no nasty current can pass through the OpAmp. Personally if I did the design again I would go for a more basic OpAmp package that the OPA1632. something like the OPA4134 that I'm using on my headphone drivers as the buffer now maybe. Anyway , I'm no expert at opAmps but will try answer your question if you post it here..  I got My ADC DSP and DAC all working together really nicely with an ARM micro. started working on multi-channel recording and playback over 8 chnannel TDM. Looking hopeful :)

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Francesco wrote 06/05/2017 at 16:07 point

Hi Ben! Yesterday i wrote the comment, then had dinner, came back to pc and... "oh, that's how it works, what a stupid question" :D I figured out the answer, I only had some issues understanding the datasheet. Of course it works like you are saying here!! You didn't reply yet, the question was usefull, i simply removed the comment, that's why you can't find it :P

Let me use this comment to congrat again... This project looks amazing to me, I check this page almost everyday hoping for news. Can not wait to see how well this works once finished :D

P.S. I decided to follow your suggestion and go directly with pcbs... as you know, now seeedstudio offers 10pcbs 10x10 cm for less than 5$... it's not worth it to do tests via breadbords :D At the moment I have almost the whole project designed, only some parts already printed, thanks to your advices it works like a charm :)

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ben biles wrote 06/05/2017 at 17:53 point

OK  ,  good to know your not wasting time with to much soldering :) take your time checking the circuits before you order boards. I usually leave it a day and check over again before ordering. I put a new pic up in the log of my 1st test backplane.. its only 4 channel. I'll make the final one and 8 channel :)  

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john.loftus wrote 08/03/2015 at 09:24 point

Thanks Ben. Glad you like it. We have a programmers reference guide here that may help with the LEDs

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ben biles wrote 08/03/2015 at 09:34 point

amazing thanks!  looks like all the info is there!  I Will code in some lights on / off after my holiday,  off to see if I can stand on a surf board for more than 3 seconds in hawaii without midi assistance :) 

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john.loftus wrote 08/03/2015 at 09:36 point

Lucky guy. That sounds like paradise.  Have fun.

Let me know how this project develops.

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john.loftus wrote 08/03/2015 at 08:18 point

Cool project Ben. We like seeing new uses for Launch Control.

Launch Control XL can also be used 'standalone' as shown below (no Arduino required)

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ben biles wrote 08/03/2015 at 09:16 point

great!  Did'nt know you could get usbmidi - midi boxes!  Just did a search and found midi - dmx controllers too for lighting! Awesome :)

Love the way 80's simple solid midi hardware lives on!
Going to try and trigger the lights on the launch control next when you press the buttons.. Any idea where I download the midi map to trigger the lights?  
Oh, also love your use of the phone charger to power the launch control XL :)

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