One more pic from the Home theatre from hell. Please note my custom varnish dipped Toroid, another noise source eliminated.
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As allen said I was meaning for you to start simple with the minidsp, i.e. use a 4th order Linkwitz Riley filter at the same crossover point you use now. REW has an auto EQ function that will give you the values to enter into the parametric EQ section to arrive at a fairly flat axial response without a lot of effort.
The remote will change between inputs and presets and can learn so you can use an existing remote. As you have found out you can control the minidsp in real time but I would urge some caution with this. It is possible to for any dsp to produce an error which if you are listening at high volume will not be nice. Digital clipping is truly horrible and is to be avoided. If you are going to boost a shelf or parametric EQ in real time then I would advise you to apply some attenuation at the input to at least the same value that you think you may boost to give yourself some headroom and I would never do this with volume turned up loud. With an analogue control it is easy to tell if you are overdoing it as the sound will gradually distort as you turn it up. With a digital system everything will be fine until you go just too far and then you get a horrible crunching sound.
I found a situation with my Najda that if I had it connected via usb control and my computer went to sleep at the same time as it was connected it triggered the DSP to go to 100% usage and required a reboot before it would work again. I don't have a minidsp and it is obviously not the same as a Najda but the warning is there.
I don't mean to dampen your hopes but as you are pretty new to all things digital it could be easy to stuff something up.
Nice stack!
W😀W,
I thought I liked it loud! Always liked front VU meters - miss the cool blue ones my McIntosh MC-150 had. Sold that for a down payment on the house I live in now. Odd really, I wanted my own house so I could enjoy being loud, but I had to give up the amp to get it! After living here 17 years, I finally got an amp that is comparable to the McIntosh, no VU meters thou. 😉
The USB mic will be nice with REW, as it will be SPL calibrated, so when You take your measurements, they will show the actual SPL of your system.
W😀W,
I thought I liked it loud! Always liked front VU meters - miss the cool blue ones my McIntosh MC-150 had. Sold that for a down payment on the house I live in now. Odd really, I wanted my own house so I could enjoy being loud, but I had to give up the amp to get it! After living here 17 years, I finally got an amp that is comparable to the McIntosh, no VU meters thou. 😉
The USB mic will be nice with REW, as it will be SPL calibrated, so when You take your measurements, they will show the actual SPL of your system.
Hi Allen;
I too wanted a house to enjoy loud music. I built my own mini 'IMAX' theatre living room. There are 7 non parallel walls, approx 32' x 24' overall, with 17 foot ceiling. I seem to have a problem with moderation.
Regarding McIntosh amps, I think it is a case of 'sour grapes' but I don't understand the design benefits of using output transformers in a solid state amplifier. I am however, in awe of an amp that requires a chain fall to install in the rack!
I too wanted a house to enjoy loud music. I built my own mini 'IMAX' theatre living room. There are 7 non parallel walls, approx 32' x 24' overall, with 17 foot ceiling. I seem to have a problem with moderation.
Regarding McIntosh amps, I think it is a case of 'sour grapes' but I don't understand the design benefits of using output transformers in a solid state amplifier. I am however, in awe of an amp that requires a chain fall to install in the rack!
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Fluid;
Thank you for the warning, re; adjusting bass and treble 'on the fly'. I have a compulsion to adjust the tone controls from one song to another. At this point, I am learning what can and can't be done with the minidsp 4x10 (or other--still welcoming suggestions) units. I will then have to learn the 'hows'. I imagine the bass control would be actually adjusting the db/octave slope of a filter that I have (installed, created (?)). I presume (hope) I could adjust the slope in one db steps and adjust hinge point frequency on the fly, also).
Now here's my real question; I can get used to turning the volume down while the filter (tone control) resets to the new parameters, if there is a glitch, BUT, do you mean turn down the dsp volume OR do I need to have an actual (six ganged, maybe 12 ganged for balanced) physical set of pots to reduce glitches at the output of the dsp?
This is a particular concern with my mid range power amp. I rebuilt the Phase Linear 700B with new PS caps, output transistors and a White Oaks loaded PCB. In the interest of noise reduction and simplification, I eliminated the input pots. They had no detentes, so the only way to insure the system remained balanced would have been to run them wide open anyway. This mod also eliminated considerable lengths of internal low level signal routing in a possibly noisy environment. The amp is sufficiently quiet to run wide open in this manner. Obviously, it (and all the amps) are last on, first off.
PS:
I am having a concern about noise (hiss) issues expressed in some threads regarding the minidsp 4x10. This would be serious, with a wide open amp (no front end pots).
S/N specs seem good, but could that be for the DAC chips, and not measured at the final (analogue) outputs? Or maybe without various software options loaded?
Thank you for the warning, re; adjusting bass and treble 'on the fly'. I have a compulsion to adjust the tone controls from one song to another. At this point, I am learning what can and can't be done with the minidsp 4x10 (or other--still welcoming suggestions) units. I will then have to learn the 'hows'. I imagine the bass control would be actually adjusting the db/octave slope of a filter that I have (installed, created (?)). I presume (hope) I could adjust the slope in one db steps and adjust hinge point frequency on the fly, also).
Now here's my real question; I can get used to turning the volume down while the filter (tone control) resets to the new parameters, if there is a glitch, BUT, do you mean turn down the dsp volume OR do I need to have an actual (six ganged, maybe 12 ganged for balanced) physical set of pots to reduce glitches at the output of the dsp?
This is a particular concern with my mid range power amp. I rebuilt the Phase Linear 700B with new PS caps, output transistors and a White Oaks loaded PCB. In the interest of noise reduction and simplification, I eliminated the input pots. They had no detentes, so the only way to insure the system remained balanced would have been to run them wide open anyway. This mod also eliminated considerable lengths of internal low level signal routing in a possibly noisy environment. The amp is sufficiently quiet to run wide open in this manner. Obviously, it (and all the amps) are last on, first off.
PS:
I am having a concern about noise (hiss) issues expressed in some threads regarding the minidsp 4x10. This would be serious, with a wide open amp (no front end pots).
S/N specs seem good, but could that be for the DAC chips, and not measured at the final (analogue) outputs? Or maybe without various software options loaded?
Hi Mediumroast
I have investigated the Rane line of DSP's. I think the HAL3s would be the most suited as far as number of inputs and outputs. Being a completely commercial unit, it seems to be a further complication to connect a CD player, optically or a laptop as a music source.
The "Halogen" software has everything needed for my application on the output side. The unit is about 3 times the cost of the minidsp 4x10, which I could probably get over, if it was the final solution!
I have investigated the Rane line of DSP's. I think the HAL3s would be the most suited as far as number of inputs and outputs. Being a completely commercial unit, it seems to be a further complication to connect a CD player, optically or a laptop as a music source.
The "Halogen" software has everything needed for my application on the output side. The unit is about 3 times the cost of the minidsp 4x10, which I could probably get over, if it was the final solution!
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Peter,
You will be able to control any parameter you wish is real time with the minidsp hooked up by usb cable. For changing the bass and treble balance it is likely that your NAD is just using a shelving filter at a defined frequency with the knob adjusting the gain. If you want to recreate what you have now, use arta or right mark to see what happens when you change the tone control this should show you if it is a basic shelving filter and at what frequency. When you get the minidsp you can use the overlay function to check if you have correctly programmed the EQ.
Baxandall style tone controls are also popular and you could look into recreating those.
I would think simple is better for this use as otherwise you would spend more time adjusting controls than actually listening to the music!
The minidsp has a built in digital volume control that will act on all the channels so you would just turn the volume down from there before you start tweaking.
Good multi channel analogue volume control is not an easy thing to achieve and is one of the reasons why I bought the Najda because it has it built in.
As I don't have a minidsp I can't give you a definitive answer on the hiss. There are many people on the linkwitz forums who have used this exact device with no issues so I don't think it is an inherent flaw.
Usually this is a matter of gain staging. The minidsp has a switch to change between output voltage levels. Most of these devices are designed to work with a 1V to 2V output voltage and power amps with a voltage gain of 26dB or so. If you use a very sensitive horn system you may need only a fraction of a volt of output from the dsp to get ear splitting volume. A low voltage signal boosted by a high voltage gain in the power amp will give you a lot of noise.
You need to keep the signal level at the right amount throughout the chain to avoid noise.
There is much more to it than this of course but usually high levels of hiss come from bad gain structure. There are so many places to cut and boost in a dsp that it can be easy to get it wrong.
You will be able to control any parameter you wish is real time with the minidsp hooked up by usb cable. For changing the bass and treble balance it is likely that your NAD is just using a shelving filter at a defined frequency with the knob adjusting the gain. If you want to recreate what you have now, use arta or right mark to see what happens when you change the tone control this should show you if it is a basic shelving filter and at what frequency. When you get the minidsp you can use the overlay function to check if you have correctly programmed the EQ.
Baxandall style tone controls are also popular and you could look into recreating those.
I would think simple is better for this use as otherwise you would spend more time adjusting controls than actually listening to the music!
The minidsp has a built in digital volume control that will act on all the channels so you would just turn the volume down from there before you start tweaking.
Good multi channel analogue volume control is not an easy thing to achieve and is one of the reasons why I bought the Najda because it has it built in.
As I don't have a minidsp I can't give you a definitive answer on the hiss. There are many people on the linkwitz forums who have used this exact device with no issues so I don't think it is an inherent flaw.
Usually this is a matter of gain staging. The minidsp has a switch to change between output voltage levels. Most of these devices are designed to work with a 1V to 2V output voltage and power amps with a voltage gain of 26dB or so. If you use a very sensitive horn system you may need only a fraction of a volt of output from the dsp to get ear splitting volume. A low voltage signal boosted by a high voltage gain in the power amp will give you a lot of noise.
You need to keep the signal level at the right amount throughout the chain to avoid noise.
There is much more to it than this of course but usually high levels of hiss come from bad gain structure. There are so many places to cut and boost in a dsp that it can be easy to get it wrong.
To combat hiss and at the same time reatin resolution, the goal has to be to have just the right sensitivity of whatever comes after the dsp, so that you don't have to adjust down the gain of individual channels more than necessary.
If you use a 105dB/W compression tweeter and an amp with 26dB voltage gain, combined with a typical HiFi woofer with a sensitivity of 88dB and an identical amp, the compression tweeter has to be dialed down a lot. The problem is that the hiss from the dsp output section (amp and buffer) will remain the same, so you've just lost 20dB S/N ratio. If you, instead of using the gain controls in the dsp to take down the tweeter level, use the amps level control or a set of inline attenuator plugs, such as the ones from Rothwell, you will have a much lower level of hiss and better dynamics.
The 4x10HD has jumpers on the inside so select the proper input and output sensitivites. This will help with this problem.
I've done this a different way. I'm using a homebrew amp with Hypex UcD180s for the woofers, and UcD 3xMP for tweeters and mids. This gives 26dB for the woofer section and 15dB for mid and tweeter section. I have tu trn down the woofer section a little, but generally I find this works well to keep the hiss down. I haven't tried running a 95-100dB AMT tweeter, yet, so I'm a bit curious if this will be enough to keep stuff quiet... I have a set of 10dB attenuators if needed 🙂
An online aquaintance of mine has had some problems with hiss in his compression tweeters. Adding a protection cap (should be used on all expensive tweeters) and using the level control on his amps completely fixed the hiss problem.
Johan-Kr
If you use a 105dB/W compression tweeter and an amp with 26dB voltage gain, combined with a typical HiFi woofer with a sensitivity of 88dB and an identical amp, the compression tweeter has to be dialed down a lot. The problem is that the hiss from the dsp output section (amp and buffer) will remain the same, so you've just lost 20dB S/N ratio. If you, instead of using the gain controls in the dsp to take down the tweeter level, use the amps level control or a set of inline attenuator plugs, such as the ones from Rothwell, you will have a much lower level of hiss and better dynamics.
The 4x10HD has jumpers on the inside so select the proper input and output sensitivites. This will help with this problem.
I've done this a different way. I'm using a homebrew amp with Hypex UcD180s for the woofers, and UcD 3xMP for tweeters and mids. This gives 26dB for the woofer section and 15dB for mid and tweeter section. I have tu trn down the woofer section a little, but generally I find this works well to keep the hiss down. I haven't tried running a 95-100dB AMT tweeter, yet, so I'm a bit curious if this will be enough to keep stuff quiet... I have a set of 10dB attenuators if needed 🙂
An online aquaintance of mine has had some problems with hiss in his compression tweeters. Adding a protection cap (should be used on all expensive tweeters) and using the level control on his amps completely fixed the hiss problem.
Johan-Kr
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BTW not sure whether it was mentioned here.
Najda has an option Balanced output board. You can use them to replace the existing SE outputs. The board is for stereo set so you can get the boards up to your needs.
Oleg
Najda has an option Balanced output board. You can use them to replace the existing SE outputs. The board is for stereo set so you can get the boards up to your needs.
Oleg
Hi Fluid;
Thanks again for your guidance. I will investigate the "linkwitz" forum and report back. I have a general idea regarding gain structure, having installed analogue Dolby theatre sound processors in movie cinemas in the 80's. One wants to keep levels high enough to make use of each components dynamic range, but low enough to avoid clipping. My collection of 'almost vintage' gear can produce horrendous levels (115 db peaks) at 15 feet with the volume pot at "2" or "3" O'clock, but no perceptible noise, hum, or hiss when the the volume pot is turned down to background levels. In fact even with the master volume up, and the signal source paused, I impress myself by how quiet the system is. I would hate to loose this with the introduction of a dsp.
Fluid, you mentioned; "Good multi channel analogue volume control is not an easy thing to achieve and is one of the reasons why I bought the Najda because it has it built in." Is the built in volume an actual multi-ganged pot or is it a digital volume control? Or some other form of non digital volume control? I was told (from reputable sources here at diyaudio) that even a digital process as 'straight forward' as volume requires a lot of digital recalculating, with it's associated drawbacks. Not my area, but I am guessing; errors, latency, noise, etc. I was told that if I insist on using "VLC player" as my media handling software (and at the risk of sounding like a 'tin ear', I am happy with it)--keep the VLC volume at 100. The Windows volume mixer is also kept at 100 for the sound card and other applications kept at zero. I don't dispute that Jriver may be a better way to go, but at this point it, it is not the most glaring fault in my system.
As to gain structure in the dsp; as mentioned, my requirements were as Pre amp (bass, treble, balance, input switching), EQ, and Crossover, with master gain ideally after the crossover IMHO. Do these individual software 'plug ins' each require level adjustment, and could all these level adjusting processes be a source of noise, in them selves?
Hi Johan;
Could you elaborate further; "Adding a protection cap (should be used on all expensive tweeters) and using the level control on his amps completely fixed the hiss problem." I have one (tantalum) 20 uF cap between my Yamaha P2200 and each array of 8 plannar tweeters. My reasons were (A); safety, if the amp decides to go DC, and (B); to enhance the slope of the crossover, which I thought is critical at high power levels. With a series connection of a capacitor, I would have thought that it would pass the HF, including hiss?
Hi Oleg;
Is this option available in boxed version? What is involved in assembling a complete kit? (Sorry, probably a huge question!)
Thanks again all, Peter
Thanks again for your guidance. I will investigate the "linkwitz" forum and report back. I have a general idea regarding gain structure, having installed analogue Dolby theatre sound processors in movie cinemas in the 80's. One wants to keep levels high enough to make use of each components dynamic range, but low enough to avoid clipping. My collection of 'almost vintage' gear can produce horrendous levels (115 db peaks) at 15 feet with the volume pot at "2" or "3" O'clock, but no perceptible noise, hum, or hiss when the the volume pot is turned down to background levels. In fact even with the master volume up, and the signal source paused, I impress myself by how quiet the system is. I would hate to loose this with the introduction of a dsp.
Fluid, you mentioned; "Good multi channel analogue volume control is not an easy thing to achieve and is one of the reasons why I bought the Najda because it has it built in." Is the built in volume an actual multi-ganged pot or is it a digital volume control? Or some other form of non digital volume control? I was told (from reputable sources here at diyaudio) that even a digital process as 'straight forward' as volume requires a lot of digital recalculating, with it's associated drawbacks. Not my area, but I am guessing; errors, latency, noise, etc. I was told that if I insist on using "VLC player" as my media handling software (and at the risk of sounding like a 'tin ear', I am happy with it)--keep the VLC volume at 100. The Windows volume mixer is also kept at 100 for the sound card and other applications kept at zero. I don't dispute that Jriver may be a better way to go, but at this point it, it is not the most glaring fault in my system.
As to gain structure in the dsp; as mentioned, my requirements were as Pre amp (bass, treble, balance, input switching), EQ, and Crossover, with master gain ideally after the crossover IMHO. Do these individual software 'plug ins' each require level adjustment, and could all these level adjusting processes be a source of noise, in them selves?
Hi Johan;
Could you elaborate further; "Adding a protection cap (should be used on all expensive tweeters) and using the level control on his amps completely fixed the hiss problem." I have one (tantalum) 20 uF cap between my Yamaha P2200 and each array of 8 plannar tweeters. My reasons were (A); safety, if the amp decides to go DC, and (B); to enhance the slope of the crossover, which I thought is critical at high power levels. With a series connection of a capacitor, I would have thought that it would pass the HF, including hiss?
Hi Oleg;
Is this option available in boxed version? What is involved in assembling a complete kit? (Sorry, probably a huge question!)
Thanks again all, Peter
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Hi Peter,
I am not sure about the boxed version, but can not see why not.
Najda has the front and back panel drawing on the web site.
So it would be easy to get it boxed even with the balanced extention. As I have seem they are offering a PSU as well. So thete might be a kit available with the chassis. Or you may need to order a chassis on your own with drawing ready available from them.
Multi way volume controls are not common units this is true, but they are available.
You can try vicol audio Maya
My guess is that muses chips could be daisy chained to one controller as well to replicate several channels at the same time.
Sent from my ONE A2003 using Tapatalk
I am not sure about the boxed version, but can not see why not.
Najda has the front and back panel drawing on the web site.
So it would be easy to get it boxed even with the balanced extention. As I have seem they are offering a PSU as well. So thete might be a kit available with the chassis. Or you may need to order a chassis on your own with drawing ready available from them.
Multi way volume controls are not common units this is true, but they are available.
You can try vicol audio Maya
My guess is that muses chips could be daisy chained to one controller as well to replicate several channels at the same time.
Sent from my ONE A2003 using Tapatalk
Effectively if you are building analogue multi channel pre you will need to start from Multi channel dac.
Feeding multi channel pre, where maya would be used as volume control. It can support any number of channels.
Maya is followed by buffers to make sure you cam transfer the signal outside of the chassis and than feed it to your amplifiers.
You can use Najda to get a stereo digital input using spdif or usb.
Than there are a couole of options
1.use on board dacs
2. Use i2s out and conmect your own i2s dacs
So this gives you flexible options.
On the other hand if you dont want to use dedicated dacs, there is a solution built by Trabquility-bass
It has on board ess dac simialr to Najda amd cam provide you with multi channel analog output l. But wont allow you to have multiple i2s outs at this stage
Sent from my ONE A2003 using Tapatalk
Feeding multi channel pre, where maya would be used as volume control. It can support any number of channels.
Maya is followed by buffers to make sure you cam transfer the signal outside of the chassis and than feed it to your amplifiers.
You can use Najda to get a stereo digital input using spdif or usb.
Than there are a couole of options
1.use on board dacs
2. Use i2s out and conmect your own i2s dacs
So this gives you flexible options.
On the other hand if you dont want to use dedicated dacs, there is a solution built by Trabquility-bass
It has on board ess dac simialr to Najda amd cam provide you with multi channel analog output l. But wont allow you to have multiple i2s outs at this stage
Sent from my ONE A2003 using Tapatalk
There is no reason why adding in the minidsp should make your system any noisier if configured correctly, if it works nicely with the 2V out of a CD player then it should work equally well with the minidsp, my comments before were to explain why some people may complain about noise and attribute it to the dsp when in actual fact that is not the real cause of the noise. In a very sensitive system an analogue volume control can have the benefit of preserving the signal to noise ratio as the noise is attenuated along with the signal, in a digital volume control the noise is constant and the signal to noise ratio is degraded with increasing attenuation.
It uses a Cirrus CS3310 chip which is digitally controlled analogue and very good. Multi ganged pots have poor channel matching.
OK we are starting to get into complicated territory again and it is hard to give definitive answers as they depend on many variables. There is a lot of misinformation about digital volume controls as well and many issues that used to exist which have long since been left behind but are still quoted as reasons against using it.
I wrote this on another thread recently:
ESS has a very good explanation of Digital vs Analogue volume here
http://www.esstech.com/files/3014/40...me-control.pdf
With an analogue volume control as you increase the attenuation you also decrease the noise which maintains signal to noise ratio.
With a digital volume control the noise remains the same and as you attenuate you bring the signal closer to the noise losing signal to noise ratio.
The reason why there is so much debate is because old myths live on long after technology has improved and made them redundant.
For every 6dB that you attenuate with a digital volume control you effectively lose 1 bit of resolution in the DAC, in a 16 bit DAC this is a real problem and anything beyond 10dB or so of attenuation can be heard as a reduction in quality.
If digital volume control is applied to a 24 or 32 bit signal then you can attenuate much more before the difference can be heard. On a 24 bit signal you can attenuate ~40dB and still reproduce a 16 bit file without any degradation. When you go to 32 bit it only gets better.
So if you use a 16 bit DAC or need to attenuate a 24bit DAC more than 40dB for your listening level then you will likely be able to hear a difference.
Many analogue volume controls bring their own issues such as channel imbalance and distortion so the analogue volume control needs to be very very good to beat a digital one if the digital is used with a high bit signal and without excessive attenuation.
The digital volume in the ESS sabre chips is outstanding. It can be adjusted in 0.5dB steps and it uses 64 individual steps per 0.5dB to ramp up or down smoothly.
The minidsp volume control is nowhere near as sophisticated but works well within it's limits.
I don't know how VLC implements their volume control but to dismiss it out of hand suggests a lack of understanding on how these things actually work. If you get a minidsp or Najda I would use the volume control in those.
An analogue volume control will always increase harmonic distortion slightly how much depends on the quality of the chip or device. Even a resistor has Johnson noise. A digital volume control won't introduce harmonic distortion but as you turn the volume down you decrease the resolution in the converter and this will increase quantisation distortion from the reduced number of steps available. The usual method to reduce the effects of this is to add dither which will increase the background noise slightly. These are the tradeoffs.
My advice is to use a good volume control, either the analogue on in the Najda, the minidsp volume or a software one like in Jriver these are all good solutions. The analogue is better if you need to use a lot of attenuation, the digital ones are just as good if not better as long as you don't attenuate too much and they are working on 24bit or higher signals.
This is the structure of the minidsp.
Volume control is done at the start, but as it is done digitally it really doesn't matter in this situation.
The Najda is similar it has input processing, per channel processing and analogue volume control at the end of the chain.
The expansion board for the Najda adds an extra two channels of balanced input and two more channels of unbalanced output, you can only use one of these so this will really not help your situation. The only options for balanced outputs are to build your own unbalanced to balanced boards. Another forum member built a Najda with 8 channels of diy built balanced output. He had a few issues to overcome in getting it working and if he was to do it again he wouldn't bother. Honestly you do not need balanced output in this application it will not help.
If you decide to build a Najda I can send you the CAD files to have an enclosure made the same as mine, without the logo and engraving if you want to keep it cheaper and simpler although the chassis from modushop is not cheap and will cost almost as much as the actual DSP board. You need a 5V and +/-12V power supply, this can be from an SMPS which is the easiest or by using simple linear regulators. None of it is hard to do but it is a lot more work than just buying the minidsp 4x10 and plugging it in!
Sorry for the length of this post but your questions aren't that easy to answer briefly...
Hi fluid, and others who have taken the time to respond;
Please don't apologize for the length of your reply. Your time is hugely appreciated. I suspect there are many audiophiles in my situation, having years of experience on the analogue side of things, but would like to 'get into the 21 century'. Thanks for your help.
Although constructing a unit from PCB sub assemblies and peripheral components would yield a tremendous sense of accomplishment, I am very anxious to address my EQ and Crossover issues. This may be contrary to the 'diy' spirit, but just learning to understand this technology is very satisfying.
At the risk of testing your patience, I have couple of dumb questions;
As shown in your screen shot above, the minidsp 4x10 has 2 analogue inputs and 2 digital. The rear panel has 3 types of digital input. Could I utilize both the S/PDIF (RCA) and Toslink (Optical) by leaving both connected and simply switching off one unit or the other (CD player or Laptop via sound card w/optical out, as source)?
Second issue; You mentioned " Volume control is done at the start, but as it is done digitally it really doesn't matter in this situation. The Najda is similar it has input processing, per channel processing and analogue volume control at the end of the chain." Just to clarify, does the minidsp use digitally controlled analogue volume controls in the boxes marked "Gain/Phase/Delay/RMS"?
Please don't apologize for the length of your reply. Your time is hugely appreciated. I suspect there are many audiophiles in my situation, having years of experience on the analogue side of things, but would like to 'get into the 21 century'. Thanks for your help.
Although constructing a unit from PCB sub assemblies and peripheral components would yield a tremendous sense of accomplishment, I am very anxious to address my EQ and Crossover issues. This may be contrary to the 'diy' spirit, but just learning to understand this technology is very satisfying.
At the risk of testing your patience, I have couple of dumb questions;
As shown in your screen shot above, the minidsp 4x10 has 2 analogue inputs and 2 digital. The rear panel has 3 types of digital input. Could I utilize both the S/PDIF (RCA) and Toslink (Optical) by leaving both connected and simply switching off one unit or the other (CD player or Laptop via sound card w/optical out, as source)?
Second issue; You mentioned " Volume control is done at the start, but as it is done digitally it really doesn't matter in this situation. The Najda is similar it has input processing, per channel processing and analogue volume control at the end of the chain." Just to clarify, does the minidsp use digitally controlled analogue volume controls in the boxes marked "Gain/Phase/Delay/RMS"?
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The good thing about mini dsp it allows you to experiment using analog inputs.
But this is the downside as well.
Analog input means you do ADC. And it reduces the quality of the dac that you have used before.
Than you need to fo the DAC to get analog out on another end. Which in the best case replaces your first dac in the worst cases uses degraded ibout from adc and degrades it further.
I mean for the price that mini dsp is offered it is more than worth it.
Plus the best approach is to keep signal in digital domain first amd than do DAC ones. Using mink dsp build in dac or anotger standalone dacs.
You can do this if you use
I2S stereo input and than route either to built in dacs or I2S outputs and stand alone dacs.
Minisharc from mini dsp or najda can both do i2s in and i2s out.
Tranqukity bass project can do only analog out using built in dac for now
Sent from my ONE A2003 using Tapatalk
But this is the downside as well.
Analog input means you do ADC. And it reduces the quality of the dac that you have used before.
Than you need to fo the DAC to get analog out on another end. Which in the best case replaces your first dac in the worst cases uses degraded ibout from adc and degrades it further.
I mean for the price that mini dsp is offered it is more than worth it.
Plus the best approach is to keep signal in digital domain first amd than do DAC ones. Using mink dsp build in dac or anotger standalone dacs.
You can do this if you use
I2S stereo input and than route either to built in dacs or I2S outputs and stand alone dacs.
Minisharc from mini dsp or najda can both do i2s in and i2s out.
Tranqukity bass project can do only analog out using built in dac for now
Sent from my ONE A2003 using Tapatalk
Actually another solution is
Diyinhk
1. Multinchannel usb
2. Ess based multi channel dac
3. Psu for the usb and dac.
And a software like jriger to do all the cinversion on your pc amd stream a multi channel digital out through usb
Once you are used to this set up you can change to any i2s input dacs you would like to.
Sent from my ONE A2003 using Tapatalk
Diyinhk
1. Multinchannel usb
2. Ess based multi channel dac
3. Psu for the usb and dac.
And a software like jriger to do all the cinversion on your pc amd stream a multi channel digital out through usb
Once you are used to this set up you can change to any i2s input dacs you would like to.
Sent from my ONE A2003 using Tapatalk
Audio Amp Adventures...
Actually, knowing what I know now, I can say, the McIntosh design is indeed very "unique". I was in my early 20's when I bought it, well before I understood amp topographies. I was more driven by aesthetics then actual performance. Although it did indeed perform very well. It left an impression on me that I have not felt until recently with the First One v1.4 I have now. I very different topography altogether. 🙂
I think You have a listening room that many of us envy. 😀
Actually, knowing what I know now, I can say, the McIntosh design is indeed very "unique". I was in my early 20's when I bought it, well before I understood amp topographies. I was more driven by aesthetics then actual performance. Although it did indeed perform very well. It left an impression on me that I have not felt until recently with the First One v1.4 I have now. I very different topography altogether. 🙂
I think You have a listening room that many of us envy. 😀
Pete you are nowhere near testing my patience don't worry 😉
Those inputs really represent Left and Right. Any of the three digital inputs or analogue can be selected as the source and yes you would leave them connected all the time no requirement to switch them off if you don't want to.
The miniDSP does not use any form of analogue volume control it is all digital. The volume control which is at the start will set the overall input level to the DSP from the source, the gain is a per channel setting which can be used in the crossover sections to adjust the relative volume of the drivers to adjust for sensitivity differences or differences in amp sensitivity if they are not all the same.
As mentioned above the analogue inputs are better used with actual analogue sources such as a turn table, reel to reel etc. If the source that you are using is digital keep it that way into the DSP.
It is always better to avoid AD and DA conversions wherever possible but also a single extra conversion at the end does not do a significant amount of damage. The real difficulty comes from being able to set the input levels correctly to avoid clipping while still keeping the signal level high. I have listened to a few tests or the gearslutz forum where studio tracks have been output directly from the DAW and processed through an AD and DA converter and it was really difficult to tell them apart, there were differences but to actually choose which one was which consistently not something I could do. And these were fairly basic converters, not Mytek or anything fancy.
I can't attach the pdf for the ESS presentation as it is too big even when zipped. I also got an error from the link. Weird because if I do a google search I can still get back to the document... Type ess analog digital volume control in a google search and it is the top document.
Buy a minidsp 4x10 and a UMIK and then your questions will relate to your actual setup and the answers should help you improve your system. It's a good way to start and you can always consider other options in the future if or when you reach the limits of what the minidsp can do.
Hi fluid and all;
I found the ESS pdf on line, regarding D verses A volume controls. Quite disappointing at first, but it seems much of the S/N ratio degradation with large attenuation can be avoided with 32 bit processors....Again, I really don't know what I'm talking about! Never the less, the minidsp 4x10 specs claim "24 bit (32 bit internal)". Encouraging.
I am going to re-read the operating manual for the dsp and my Creative Sound-Blaster-X-Fi-HD card to avoid any unnecessary questions, but I still have one burning issue regarding the use of my laptop as media source;
Can I use the sound card's optical out, to connect to the minidsp? If I avoid attenuation and other filters, is this extra component degrading the digital signal?
OK, another dumb question, can the laptop be directly connected to the dsp via a cable that adapted to USB?
Thanks all, I have some reading to do before any more questions!
Peter
I found the ESS pdf on line, regarding D verses A volume controls. Quite disappointing at first, but it seems much of the S/N ratio degradation with large attenuation can be avoided with 32 bit processors....Again, I really don't know what I'm talking about! Never the less, the minidsp 4x10 specs claim "24 bit (32 bit internal)". Encouraging.
I am going to re-read the operating manual for the dsp and my Creative Sound-Blaster-X-Fi-HD card to avoid any unnecessary questions, but I still have one burning issue regarding the use of my laptop as media source;
Can I use the sound card's optical out, to connect to the minidsp? If I avoid attenuation and other filters, is this extra component degrading the digital signal?
OK, another dumb question, can the laptop be directly connected to the dsp via a cable that adapted to USB?
Thanks all, I have some reading to do before any more questions!
Peter
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