I'm nearly all digital and bi-amped:
1. MacBook (important to note: no digital attenuation available for some of my music sources (such as web using Safari) when Breeze or monitor are connected as output
2. "Breeze" USB to SPDIF/coax box
3. Behringer DCX2496 DSP (using the SPDIF/coax input and analog output)
4. 4-gang analog volume control pots with a handy knob near my chair
5. analog amps.
Now i am going tri-amped and must upgrade the 4-gang pot. Somewhat nutty to make a little box with 6-gang pots and likely there'll be tracking errors in among the pots.
So the natural replacement would be an adjustable digital attenuator on the USB line between the MacBook and the Breeze converter - with a knob and in a little box. Or in the middle of the SPDIF/coax line.
Obvious as this sounds, I haven't found such gizmos.
Any help re-thinking my circuits or finding the gizmo greatly appreciated.
Ben
1. MacBook (important to note: no digital attenuation available for some of my music sources (such as web using Safari) when Breeze or monitor are connected as output
2. "Breeze" USB to SPDIF/coax box
3. Behringer DCX2496 DSP (using the SPDIF/coax input and analog output)
4. 4-gang analog volume control pots with a handy knob near my chair
5. analog amps.
Now i am going tri-amped and must upgrade the 4-gang pot. Somewhat nutty to make a little box with 6-gang pots and likely there'll be tracking errors in among the pots.
So the natural replacement would be an adjustable digital attenuator on the USB line between the MacBook and the Breeze converter - with a knob and in a little box. Or in the middle of the SPDIF/coax line.
Obvious as this sounds, I haven't found such gizmos.
Any help re-thinking my circuits or finding the gizmo greatly appreciated.
Ben
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Thanks for the software suggestion.
Presently, the stock "Volume:" slider doesn't control loudness for the Breeze SPDIF converter (identified as "Speaker" in menu bar) or docked monitor. Does the egMac2?
Main problem is "with a handy knob near my chair".... so I can turn conveniently down the music at my chair when the phone rings or other reasons.
Lots of outboard USB DACs with volume control for headphones, etc. But odd not to find this type of gizmo.
B.
Presently, the stock "Volume:" slider doesn't control loudness for the Breeze SPDIF converter (identified as "Speaker" in menu bar) or docked monitor. Does the egMac2?
Main problem is "with a handy knob near my chair".... so I can turn conveniently down the music at my chair when the phone rings or other reasons.
Lots of outboard USB DACs with volume control for headphones, etc. But odd not to find this type of gizmo.
B.
SPL Volume 8
I run an active four way horn system. I use this product for 8 channel analogue volume control and can highly recommend it.
Volume 8 – SPL
Volume 8 – SPL
I run an active four way horn system. I use this product for 8 channel analogue volume control and can highly recommend it.
Volume 8 – SPL
Volume 8 – SPL
Thanks for that link. The "Volume-8" active analog gang volume control - in a great little box - would be an excellent solution for me in most ways. (Or possibly a clone on Alibaba.) Like my present system, I conserve all my digits until past the DSP box. At which point the analog gang VC operates.
In fact, the Volume-8 is just the sort of thing I was thinking of making using, I assume, voltage-controlled resistors. Except I'm on the steep learning curve for that circuitry.
There are two big problems for me with the Volume-8 device: (1) us$700 and (2) DB25 connectors input and output which requires substantial construction effort to build a larger box to match RCA cables input and output to the two DB25 cables.... wow.
Today, I am going to measure distortion in my system, comparing:
(a) low digital signal to the DSP controlled by digital VC at the digital source
versus
(b) high digital signal with analog VC after the DSP.
In fact, the Volume-8 is just the sort of thing I was thinking of making using, I assume, voltage-controlled resistors. Except I'm on the steep learning curve for that circuitry.
There are two big problems for me with the Volume-8 device: (1) us$700 and (2) DB25 connectors input and output which requires substantial construction effort to build a larger box to match RCA cables input and output to the two DB25 cables.... wow.
Today, I am going to measure distortion in my system, comparing:
(a) low digital signal to the DSP controlled by digital VC at the digital source
versus
(b) high digital signal with analog VC after the DSP.
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Testing was rather crude and not fully conclusive. Varied the signal into the SPDIF converter and inversely reduced the gain of the amp feeding a ribbon tweeter at 4500 Hz, about 90dB SPL at 5 inches where the mic was placed.
Figures below are all loose and relative.
Signal from the laptop ranged from -45 to -5 dB (no bars on the Behringer DSP to 4 bars input). Output from the tweeter kept near 90 SPL.
Result: THD from the tweeter read about .2-.3% at all values of drive from the laptop, but with lots of ambient noise, which is pretty good for a driver.
My first impression is that the laptop (or downstream USP attenuator) can run the DSP at "no bars" and yet show no detectable distortion. Hence the loss of digits even with my inexpensive system does not harm the music.
Crude testing.
B.
Figures below are all loose and relative.
Signal from the laptop ranged from -45 to -5 dB (no bars on the Behringer DSP to 4 bars input). Output from the tweeter kept near 90 SPL.
Result: THD from the tweeter read about .2-.3% at all values of drive from the laptop, but with lots of ambient noise, which is pretty good for a driver.
My first impression is that the laptop (or downstream USP attenuator) can run the DSP at "no bars" and yet show no detectable distortion. Hence the loss of digits even with my inexpensive system does not harm the music.
Crude testing.
B.
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compromise?
Ben, have a read of this thread Apogee Rosetta 800 + especially posts #6 & 8 where Krivium talks about using a three preset analog attenuator between the DACs and amps (with high spl, mid and late night position).
Prior to apogee, I used a Behringer DCX2496 & was pleasantly surprised with its sound quality after reading several threads about how bad they sound.
Rgds
Barry
Ben, have a read of this thread Apogee Rosetta 800 + especially posts #6 & 8 where Krivium talks about using a three preset analog attenuator between the DACs and amps (with high spl, mid and late night position).
Prior to apogee, I used a Behringer DCX2496 & was pleasantly surprised with its sound quality after reading several threads about how bad they sound.
Rgds
Barry
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You can buy cables with a DB25 on one side and RCA or XLR on the other. It's a stock item with small home studios and soundcard-based systems.
It even says 'Tascam standard' on the back of the box. (In this case, Tascam D-88).
Jan
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Many thanks to you and to Mr BA (esp for directing me to krivium's two learned posts). A device like Apogee Rosetta 800 (even if still available) would be far too sophisticated solution for me (read: expensive) and still not achieve my objective of being less than an arm's length from my chair.
If I understand torgeirs' "rpi minicomputer" correctly, would that be for me to finally learn how to program my dust-gathering Ardurino or a Raspberry Pi?
Or, in the spirit of DIY and in a world of inaccessibly high-priced audio goodies, a solution that meets my needs if I opt for a digital attentuator would be simply to get a cheap old laptop or pad or maybe smartphone to do the variable attenuation chores upstream of the DSP.
Digital attenuation is now looking more feasible to me (see post 6 above). Pano (in the DCX2896 mother-thread) points out that his loudest to softest volume settings are maybe 12dB apart. If so, I can attenuate my amps ("gain management") to be super-loud on full digits and then it is operating late-at-night softly at only a few digits less. And if I need super-low, at the loss of a few more digits, it isn't critical listening anyway.
For Pano's valuable insight, see post #4458
Behringer DCX2496 digital X-over
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Thanks. I didn't know but google does. But I'll never be ready to spend almost $200 for each of two cables (plus shipping).
And besides, is my colour vision good enough to tell 8 RCA or 8 XLR plugs apart?
B.
Comming back to me: digital attunation before resampling can be benefitial
Software to detect +0dBFS waveforms and overload times
Going rpi and camilladsp also gives a free high end resampler.
The resampling device in behringer has some minor flaws. 96khz in is the best.
Software to detect +0dBFS waveforms and overload times
Going rpi and camilladsp also gives a free high end resampler.
The resampling device in behringer has some minor flaws. 96khz in is the best.
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Link to tcelectonics article from 2006 is broken
The start of this covers it: https://tech.ebu.ch/docs/techreview/trev_310-lund.pdf
The start of this covers it: https://tech.ebu.ch/docs/techreview/trev_310-lund.pdf
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And even the DAC is behaving better at -4dbFS
And not much worse at -20dbFS
AKM AK4393 | It's only Audio
So my choice is:
Source
Spdif or usb or other
24bit to decimal number
Digital attunation
Resampling 96 k
Decimal to 24bit
Spdif
Dcx
Maybe some resistive attunation or pots on amp
Amps
Guess that chain is described a million times before;-)
And not much worse at -20dbFS
AKM AK4393 | It's only Audio
So my choice is:
Source
Spdif or usb or other
24bit to decimal number
Digital attunation
Resampling 96 k
Decimal to 24bit
Spdif
Dcx
Maybe some resistive attunation or pots on amp
Amps
Guess that chain is described a million times before;-)
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Just one last point.
The Pano solution is the best and cheapest.
There are some challenges with several digital boxes in a chain and no master clock
This can be mittigated but it is not simple
So at 0.01 distortion limit and the use of the analog input, it is easy to achive with an inexpensive home studio soundcard between source and dcx
The benefit of analog in and out is that the same masterclock is used on ad, dsp and da.
The Pano solution is the best and cheapest.
There are some challenges with several digital boxes in a chain and no master clock
This can be mittigated but it is not simple
So at 0.01 distortion limit and the use of the analog input, it is easy to achive with an inexpensive home studio soundcard between source and dcx
The benefit of analog in and out is that the same masterclock is used on ad, dsp and da.