freeDSP main thread

This thread is a place for links to other threads that are related to the freeDSP project. Feel free to post with these links and a brief comment on what the thread discusses. Occasionally the moderators will consolidate them into fewer posts.

Please create individual threads (and link them from here) to connect with other people working with the freeDSP for discussion and to support each other. Please keep in mind that freeDSP is a spare-time project and not a commercial product. If you want to get a freeDSP you need to build it yourself (manufacture board, order parts, …) or organize centralized buying with other DIYers.

The freeDSP is a low-budget open-source digital signal processor family, which is published under a creative commons license. It allows the unrestricted use and modification of the modules. The applications range from active loudspeaker concepts and room equalization over advanced musical effect processors to car audio signal processing. We would be happy if you join us and improve or extend the project.

GitHub is used for file exchange. If you want to join the development team, just send us a private message with your ideas and your GitHub user name. Most freeDSP PCBs will be designed using KiCad. Some guidelines were defined to make future freeDSP development and extensions as compatible as possible. These layout guidelines can be found in the freeDSP-Wiki.

In the following you’ll find a summary of the current freeDSP plans:

green = sources tested and available,
black = work in progress,
gray = on the wish list

freeDSP motherboards:
freeDSP CLASSIC (ADAU1701 / 2 x In & 4 x Out Analog via RCA) freeDSP thread, SigmaStudio AutoEQ
freeDSP CLASSIC SMD (ADAU1701 / 2 x In & 4 x Out Analog via RCA)
freeDSP INSANITY (ADAU1452 / 4 x In & 4 Out Bal. Analog via Jack, alt. 8 In x 8 Out Unbal. / 1 x In & 1 x Out SPDIF via RCA & Toslink)

freeDSP compatible motherboards:
PiDSP (ADAU1450 / RasPi In + Out / 3 x I2S In + Out ) PiDSP thread

freeDSP programmer:
freeUSBi + EZ-USB

freeDSP IO expansions:
freeDSPx AES/SPDIF IN (1 x In AES/EBU via XLR / 1 x In SPDIF via RCA)
freeDSPx SPDIF IO (1 x In & 1 x Out SPDIF via RCA & Toslink)
freeDSPx BAL OUT x16 (16 x Bal. Out Analog via SUB-D)
freeDSPx ADAT IO x3 (3 x In & 3 x Out ADAT via Toslink - maybe even 4 IOs)
freeDSPx BAL IO x4 (4 x In & 4 x Out Balanced Analog via Jack, alt. 8 In x 8 Out Unbal.)
freeDSPx UNBAL IO x2 (2 x In & 2 x Out Analog via RCA)

freeDSPx PHONES AMP
freeDSPx AMP
freeDSPx HDMI IO
freeDSPx DOLBY/DTS/AC3 IO

freeUSBi kits and freeDSP classic kits are almost always available - please use the contact fomular on our website :)
 
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Any advice on how to set up a 2-way x-over with dipole compensation on the Low pass?
I've played around a bit with Sigma Studio, but apart from using behringer dcx2496 I am a complete novice when it comes to DSP.
I'm refering to what filters or other tools to use.
The High pass filter, I'd like the signal to go "unaltered" to the passive x-over in the mid/high part of the dipole speakers.

I will try to use the FreeDSP classic with I²S in via an audio-widget board. And once I have it working, use two es9023 DAC's as output via I²S from the FreeDSP board.
 
Does this look like a decent starting point?
(attached screenshot of SigmaStudio)

I do have two fully populated (and tested as far as that they take programming), so I guess I could use one FreeDSP Classic/channel, allowing for more complex filters etc.
I've also made a measuring mic with amp according to the info on Linkwitz site, including the modifications to the mic.

When the system is up and running enough for me to measure it with REW, I'll need to read up on how to use REW etc with SigmaStudio to get the best sound quality possible in the room.
 

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Hello Mayday,

if you have measuring capabilities I would first to start with a measurement of the different loudspeakers. It is crucial to understand the frequency response and SPL levels each speaker is providing.
From that you can choose/plan the filters and level adjustments you need.
Important to know is also which amplifiers you use. Are they identical? Or do they habe different output power and amplification?

For a first implementation I would go for the infernal DACs and one Freedsp as it offers sufficient computing power for many solutions.

Good luck! :xfingers:
Walter
 
Hello Mayday,

if you have measuring capabilities I would first to start with a measurement of the different loudspeakers. It is crucial to understand the frequency response and SPL levels each speaker is providing.
From that you can choose/plan the filters and level adjustments you need.
Important to know is also which amplifiers you use. Are they identical? Or do they habe different output power and amplification?

For a first implementation I would go for the infernal DACs and one Freedsp as it offers sufficient computing power for many solutions.

Good luck! :xfingers:
Walter

Hi,
I have a mic/mic-amp built/modded according to Linkwitz site.
The speakers are dipoles, Dayton 15" IB's in H-baffles, passively x-overed mids/highs are Dayton 7" Reference and B&G Neo 8.
The amp situation is a bit fluid at the moment.
I tested the first of the populated FreeDSP classic boards today using a 2xTDA7293/channel amp for the woofers and my Arcam Alpha 8 power amp for the mids/highs.
As this was a test, I used analog in and the internal DAC's of the ADAU1701.
Source was raspberry pi 3 via XMOS USB/ES9023 DAC.
It worked apart from the left HF being dead silent. The LF needs some gain as well.
When I get the FreeDSP classic board working on all outputs, I'll measure the system/room with the mic and REW.
I did test the second FreeDSP board as well, though I'm not sure that I got the sw written to the EEPROM correctly as that board did not produce any sound at all.

I have two ES9023 boards capable of I²S slave mode at home already, intended for the FreeDSP. I also have a ES9018K2M board to try/compare to the others.
My PC is in another room, so adjusting the FreeDSP on the fly would be difficult.

Then I need to figure out what to do with the measurement from REW, how to implement them in Sigma Studio.
I am new to DSP, my only prior experience is using a dcx2496.

Any and all help is appreciated :)
 
Hello Mayday,

unfortunately measurement of dipols are the most difficult ones as
frequency response is depending on room, distance to micro etc.

I never worked with REM but with ARTA which you can use in a basic version free of charge.

On Linkwitzlab website you might find a measurement setup thats works up to few hundred Hz. Above you can make measurements having the micro on your listening position. If I find the link i will send it.
Walter
 
German below!

Hi,
I want to build a 'freeDSP Classic BAL A' for my speaker-system. I made a request on itead.cc for building the whole PCBA but say said that they need another BOM List than that which is on GitHub.
That's why I want to ask, if I can use the 'freeDSP Classic' Part List and what do I have to change to have all parts needed. (I need this on Digikey but any other source is as well welcome, I can copy that my own to Digikey).

German:

Hi,
ich möchte mehrere 'freeDSP Classic BAL A' für mein Lautsprechersystem herstellen lassen (hier: itead.cc als PCBA). Nach dem ich alle Dateien von GitHub für die Bestellung angegebene hatte, wurde mir gesagt, dass die BOM Liste im falschen Format mit falschen Infos sei.
Aus diesem Grund Frage ich hier, ob man die Part Liste der 'freeDSP Classic' benutzen kann und was man ändern muss, damit ich alle Teile für die 'freeDSP Classic BAL A' bestellen kann. (Anbesten bei Digikey aber ich kann die Teile auch von anderen Webseiten dort hin übertragen).
 
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Hi.
I have two freeDSP classic modules.
I connecet them by attached schematic.
My idea is to use first freedsp as main DSP. The second one as ADC converter for more analog inputs.
Į connected 3,3v to 3,3V;
GND to GND
SDATA out1 from secont to SDATA IN from main board.
BCLK OUT from main to BCLK IN to second.
the clock jumper from second board is on external mode.
sigma project configured. But i have no CLOCK on second board.
Can sombody help me? Maybe i connected wrong pins? How sinchronise boards on the same CLK?
 

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CS8422 boards? FreeDSP "SPDIF In"

Hi,
I'm looking for a way to explore the CS8422 (standalone, rather than necessarily using the FreeDSP).

With my limited SM soldering skills was hoping there's a board out there that will make things easier. I found this:

freeDSP | An Open-Source Low-Budget Audio DSP
I/O expansions:

with the documentation here
freeDSPx AES:SPDIF IN - Google Docs

and that looks excellent. I wouldn't need the XLR, the pots and switches. Just a bare board, possibly with the SMD fitted, would be great.

Anyone else interested?
 
Does anyone else find the organization of the FreeDSP project a bit daunting? Finding the bits of each sub-project is difficult.

The table on the bottom of our website shows the relevant source files for each sub-project.

There's a GIT repository for each sub-project. All GIT repositories share a common structure.

There's a Getting-Started-Guide for the main sub-projects available.

If someone's offering ready-built devices or kits, we will also provide a link on our website.

What would you recommend? We're always open to suggestions :)
 
Hello and thank you for your response.

I'm sorry to be critical I've been looking at FreeDSP for a long time now. I just didn't want to hijack this thread. I think that's were my confusion starts. The fact there is no permanent section for FreeDSP on the forum makes it difficult to decipher where the projects are being discussed. My confusion is that when I look at the website I see a bunch of projects but some of them I simply can't find any information about them. I suppose they are spread around through this Diyaudio forum.... But not seemingly organized into a neat section so i can browse around them.

For example, the INSANITY or LUMIÈRE. Where are they being discussed? :)

Again, not to hijack this thread. But I'm deep into the research of trying to figure out how to make a USB Soundcard, with a built in DSP. So the idea is to have USB OTG (slave device) connected to a PC, playing audio through USB Audio, into the DSP, processed, and then outputted through to the DAC. I'm trying to look at all the FreeDSP projects, and determine if any of them can do that. Any advice?

I'm also interested in contributing an expansion card for the FreeDSP family (if I can find out that we can accomplish the above) which would be based on the TLV320AIC3104 codec chips. (because they are cheap, and flexible. My project requires just that). Would you be interested in having an expansion board that had 4x3104 chips so it would produce an 8x8 balanced/single ended card with built in mic/instrument gain?

Thanks for your help!
 
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