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 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
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 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 :-)
Quick start Thread for 2-way crossover with FreeDSP
Hello FreeDSP friends,
if you need some help to have a first 2-way crossover design with FreeDSP (Classic) have a look to my thread. You find there also information how to include volume control (with buttons and IR control) and to prevent power on/off noise.
You find the link here:
aes/spdif pcb and free USBi board's
I ordered an aes/spdif board, but because transport was the major part of the costs, I ordered some more.
Please contact me if you're interested in the pcb. They are 9 euro each, send by standard post in Europe.
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.
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:
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 :)
I think the problem with the left HF actually is that the old PIO's in the x-over has died on me.
I got the output I thought was a problem to work when connected to something else.
I've ordered replacement caps for the x-overs, MKP's this time.
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.
Habe a look here:
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