I think I've seen similar before PC sound cards, which have the option of multiple channels. What you need is one that can be programmed then able to be used as a standalone. Sure I've seen stuff that was equally as programmable but cheaper, not sure what sort of musical sound enjoyment could be had from their DACs though. Maybe nowhere near as good as some 'good or better DACs known for music'.
IME, the diyinhk DACs I have tried were junk to me. Can't recommend one based on the experience so far.
It is clear that there is some demand for multi-channel dacs for people who want to multi-amp. Ockto makes one that measures pretty well, but last time I looked it appeared they couldn't meet market demand. Don't know how they sound either. Expect its probably about like the typical 'ESS sound.'
It is clear that there is some demand for multi-channel dacs for people who want to multi-amp. Ockto makes one that measures pretty well, but last time I looked it appeared they couldn't meet market demand. Don't know how they sound either. Expect its probably about like the typical 'ESS sound.'
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I could have guessed that functionality but the quality is not there, a lot like a lot fo these cheaper DSP units, or even some of the more expensive for that matter.
Its all about having a good DAC for quality music enjoyment, or of course in you're analogue'ing it, something decent for it to go through to be your active crossover.
Its all about having a good DAC for quality music enjoyment, or of course in you're analogue'ing it, something decent for it to go through to be your active crossover.
What about running this with a receiver with pre outs? Does going through this into the receiver turn the quality into something less than just going from amp to receiver like I’ve been doing? I plan on building two 3 way speakers and using one of these to do the crossovers and eq. I can turn off the eq on the receiver end.
It can be used and will work. The debate is the actual quality of the units so far. You get amazing control of the sound and ability to shape it but often it goes through a complex process box of tricks to achieve this. Although analogue crossovers can also be complex I think they take less away from the signal / sound, yet the only problem is they are not as easy to tailor on the fly, and perhaps out of the reach of designing a good one for many or even most hobbyist diy builders.
Many say active crossovers are better, but they are still crossovers. They do the same job just a different method. I would not say they are better, maybe more versatile, certainly not better.
Many say active crossovers are better, but they are still crossovers. They do the same job just a different method. I would not say they are better, maybe more versatile, certainly not better.
Hi, How can input signal distortion be lowered?FYI: FreeDSP Classic SMD A/B plus-II performance
To know the ADAU1701 device's audio performance and it's characteristics, I measured my FreeDSP Classic SMD A/B plus II. (Crossover 24dB/oct@400Hz HPF output)
1st figure is IMD/THD vs. Input Level. (Green trace is 0404USB it's self)
2nd figure is 1kHz Sin-wave 1.8V input distortion performance.
3rd figure is Remain Noise Spectrum.
Comparing this measured result and DSP408's result shows, there might be some room to improve the Large level signal input distortion performance of DSP408.
I saw a video on YouTube that anything over 55 (56 and up) on the volume of the dsp started to clip. That was in a car though but it might be good to limit the volume to 55 and under.
I think that reference level is of wired remote? How much would it translate to PC GUI volume level?
No idea really. I don’t know how to measure it with my receiver.
here’s the video. I keep mine at 55 even though I have no idea if it does the same thing connected through a receivers pre outs.
Member
Joined 2018
My meaning of distortion is not a waveform Clip. A non-linear characteristic below the clipping level might be the root cause of this behavior.
So I plan to input a phase inverted signal for another side of ADAU1701 embedded ADC and mix them digitally to improve the distortion performance near the full-bit level inputs. I'm not sure, but I'll try with my FreeDSP Classic SMD A/B Plus-II later.
Member
Joined 2018
Hello Again!
I just tested with my FreeDSP Classic SMD A/B board.
The first half part is Single-Ended Input as same as DSP-408, second-half part is using 2 ADAU1701's inputs using like as differential ports. Sampled signals were substructed In a digital domain.
The Result:
There is a big difference between single-ended and differential-Input. Especially even order harmonics are suppressed.
Total Harmonic Distortion decreased as follows...
(100Hz) 0.00723% --> 0.00287%
(1kHz) 0.00642% --> 0.00171%
So, I made a plan for designing a single(mono) channel ADAU1701 4way crossover FreeDSP project.
CyberPit
I just tested with my FreeDSP Classic SMD A/B board.
The first half part is Single-Ended Input as same as DSP-408, second-half part is using 2 ADAU1701's inputs using like as differential ports. Sampled signals were substructed In a digital domain.
The Result:
There is a big difference between single-ended and differential-Input. Especially even order harmonics are suppressed.
Total Harmonic Distortion decreased as follows...
(100Hz) 0.00723% --> 0.00287%
(1kHz) 0.00642% --> 0.00171%
So, I made a plan for designing a single(mono) channel ADAU1701 4way crossover FreeDSP project.
CyberPit
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Member
Joined 2018
I did some unusual experiments on DSP408. It consists of two audio paths that are connected in series.
If I invert the output phase in the first stage, the second harmonic level in the final output signal was decreased slightly.
On the other hand, there's no difference if I changed the second stage output phase.
According to that, ADAU1701 embedded ADC has some non-linear distortion characteristics and it can be decreased if I use it in the differential input technic.
It seems too hard to hack the DSP-408 firmware for me, developing a new ADAU1701 board is much easier for me...
CyberPit
If I invert the output phase in the first stage, the second harmonic level in the final output signal was decreased slightly.
On the other hand, there's no difference if I changed the second stage output phase.
According to that, ADAU1701 embedded ADC has some non-linear distortion characteristics and it can be decreased if I use it in the differential input technic.
It seems too hard to hack the DSP-408 firmware for me, developing a new ADAU1701 board is much easier for me...
CyberPit
Attachments
A few more measurements of this DSP. Hopefully I'm not polluting the thread but this seems to be the defacto data repository.
I'm using GUI (V1.24), Firwmare (V6.21), Hardware (V??). I found the GUI is easy and intuitive to use and I'm keeping mine because I find it useful. I'm just hoping for better firmware in the future, or (preferably) having it unlocked and loading my own firmware.
The baseling loopBacks (pic#1 and #2) show what the H/W is capable of (single ended I/O) and I would actually be happy with that . However, once you start applying filters the performance suffers and there's no valid reason for that. I've included filter templates (in black) that were created by RePhase and overlaid them on the loopback measurements. There is only one filter template deviation (HP-6000) and it only occurs at higher freq . <speculation> it appears to be from freq warping as you approach the Nyquist freq. I've tried using Sigma Studio's filter generator and it does the same thing and I could not find pre-warp compensation. </speculation>. The more relevant issue (for me) is the increase in digital noise for all filters and probably EQ as well (have not measured that yet). <speculation> It appears to be related to the way the fixed point math is performed where small numbers divided by small numbers yields a noisy result. There is also potentially the filter coefficient representation where the LF filter coefficients are very small compared to the HF coefficients. </speculation> This may not be a real problem sending HF noise to a subwoofer (averages it out) but I suspect it can be fixed in firmware. Unfortunately ADI does not provide a disassembler or even details on it's DSP instruction set, nor does Sigma Studio disclose its code implementation. I would expect the processed output result to approach the pass through performance (within a few dB). This also makes the output noise levels more sensitive to input levels.
A fixable, and annoying feature is the "auto power off". On my 3way horn system it shuts off at 74dBc and there is no way to disable this "feature". There is a remote enable pin but there is no 12vdc output for self enabling it. It will power down when the USB is connected and even mid USB transfer. I have an unused input connected to a signal generator to keep it alive.
Pic#1 - measurement soundcard loopback
Pic#2 - loopback with DSP in pass through (no processing, no gain)
Pic#3 - REW measurement with filters
I'm using GUI (V1.24), Firwmare (V6.21), Hardware (V??). I found the GUI is easy and intuitive to use and I'm keeping mine because I find it useful. I'm just hoping for better firmware in the future, or (preferably) having it unlocked and loading my own firmware.
The baseling loopBacks (pic#1 and #2) show what the H/W is capable of (single ended I/O) and I would actually be happy with that . However, once you start applying filters the performance suffers and there's no valid reason for that. I've included filter templates (in black) that were created by RePhase and overlaid them on the loopback measurements. There is only one filter template deviation (HP-6000) and it only occurs at higher freq . <speculation> it appears to be from freq warping as you approach the Nyquist freq. I've tried using Sigma Studio's filter generator and it does the same thing and I could not find pre-warp compensation. </speculation>. The more relevant issue (for me) is the increase in digital noise for all filters and probably EQ as well (have not measured that yet). <speculation> It appears to be related to the way the fixed point math is performed where small numbers divided by small numbers yields a noisy result. There is also potentially the filter coefficient representation where the LF filter coefficients are very small compared to the HF coefficients. </speculation> This may not be a real problem sending HF noise to a subwoofer (averages it out) but I suspect it can be fixed in firmware. Unfortunately ADI does not provide a disassembler or even details on it's DSP instruction set, nor does Sigma Studio disclose its code implementation. I would expect the processed output result to approach the pass through performance (within a few dB). This also makes the output noise levels more sensitive to input levels.
A fixable, and annoying feature is the "auto power off". On my 3way horn system it shuts off at 74dBc and there is no way to disable this "feature". There is a remote enable pin but there is no 12vdc output for self enabling it. It will power down when the USB is connected and even mid USB transfer. I have an unused input connected to a signal generator to keep it alive
.Pic#1 - measurement soundcard loopback
Pic#2 - loopback with DSP in pass through (no processing, no gain)
Pic#3 - REW measurement with filters
Attachments
Are there any other workarounds for this? I'm running mine from a Schiit Audio Sys passive pre and the only fly in the ointment I've found is this. Would using a buffer to boost voltage and impedance serve as a workaround? I really hate that it shuts off during quiet passages.A fixable, and annoying feature is the "auto power off". On my 3way horn system it shuts off at 74dBc and there is no way to disable this "feature". There is a remote enable pin but there is no 12vdc output for self enabling it. It will power down when the USB is connected and even mid USB transfer. I have an unused input connected to a signal generator to keep it alive
Member
Joined 2018
opened up a support ticket @ Parts Express- it was a simple matter of putting it in "remote" mode, which keeps it on all of the time. measurements and tuning to follow.
After testing the unit for a while, it was put away, waiting for a project. I suppose you could use it's auto wiring harness (P1) and wire the 12VDC supply to the remote on. I would prefer a firmware fix (switch) to allow setting the unit always on. I did have a chat with PE support to see if there was another way, and not at this time (firmware 6.21).
I have not seen schematics for the unit, so I'm not sure how the auto-mute is implemented. My wish list would also include digital audio over USB to bypass the A/D stage, and being able to load my own firmware (SigmaStudio).