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Should i use an analog level monitor control to keep the signal cleaner ?
Old 21st April 2017
  #1
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Should i use an analog level monitor control to keep the signal cleaner ?

Hi gang,


I just received my new monitors and before getting them i also purchase a used echo audiofire 4 interface. I have no analog monitor level controller so i have to use the audiofire digital interface level monitor. Note in the picture below that the "Play A1 & A2" is linked to my windows volume control and the "Out A1 & A2" is the main volume control of the interface.

The monitors with their sensitivity at 0dB default position run way too hot and i have to put the interface level control at -50dB to work at reasonable level. Even at lowest -60dB i still have little sound that comes out. Don't know if this is normal or not..

So what i did is turn down the sensitivity of the monitors of -10dB and raise a bit the interface level control and what i found out is that the sound quality somehow get better, now it make me things even more confused..

I remember have read somewhere in the past that someone advise that the better thing to do while having only a interface level control is to use an analog level control and keep the digital interface level control all the way up to not alter the signal quality.

So how much this statment is valid ? Do the digital signal need to be at its maximum amplitude level to keep the signal clean as possible ? Seems to be fair to me because i got this impression that when i turned down the sensitivity and raise up the interface level i got a better sound quality but still confused about how much this theory and impression could be valid

So now for working at confortable level the way i have setup my level chain is my monitors sensitivity turned down about -10dB, my windows level all the way up at 100% and my interface level control at around -40dB ( if i raise it up it become way too loud for me ).




Thank you guys
Old 21st April 2017
  #2
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Animesh Raval's Avatar
Hello mate, I just posted a link to this article somewhere else and think it might help you:

Monitor Wizard |

You can calibrate your speakers and your interface together to get an ideal volume for listening/mixing depending on the size of your room and what volume reference level you wish to work to.

Note that the dB scale in the screenshot is dBfs...I'd recommend using a analogue style methodology of keeping RMS levels at 0VU (+4dBu) which translates (I hope I get this right) to -20dBfs digital ref level.

So basically when listening, you would be hearing a -20dBfs signal at your volume reference level which could range anywhere from 74dB SPL in a small/medium room up to 85dB SPL in a very large room.

The article above will help you calibrate things. Essentially, I would find a reference position on your audio interface volume knob - say for example 12 O'Clock position and set it there. Then turn the gain on your speakers right down, and using Pink Noise (RMS -20dBfs, bandwidth 500 to 2KHz) test first right speaker then left and increase gain on the speaker until it reaches your desired volume SPL reference level in the listening position. You will need an SPL meter for this.

You can download pre made test files from here: Blue Sky » Blog Archive » Blue Sky Calibration Test Files

I hope this kinda makes sense? Others more experienced/knowledgeable than me may pitch up and tell you I'm being too complicated/leading you astray! But I think based on what you've said, you essentially just need to get your interface volume and speaker volume in a happy medium...and calibrating your monitors with pink noise will hopefully be a fairly methodical way of getting there.

Note that music played through your windows system - i.e. iTunes and CDs will be mastered near to 0dBfs, and using the method above you will have built in a headroom of 20dB so it will sound massively loud compared to your music in your DAW if you're working to -20dBfs. So yes you will have to turn the interface volume right down when working outside of your DAW. But in calibrating your system as above, you can get back to the ideal listening volume for mixing easily.

I hope this helps rather than confuses!
Old 21st April 2017
  #3
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Best practice to keep the signal cleaner is to avoid an analog level monitor control. Turn your monitors sensitivity down all the way and do any other attenuation digitally.
Old 21st April 2017
  #4
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The answer doesn't seem to address the question. I think the question might be, when you digitally attenuate your monitor level, are you compromising the quality and resolution of the digital signal? Passive or active analog attenuation does not normally compromise the quality of the audio signal.
I have asked the same question, and lacking a clear answer, I run digital outputs maxed and control my monitor level with a Radial passive monitor controller. As you might guess, I'm not extremely knowledgeable (nor very trusting) about digital signal attenuation.
Old 21st April 2017
  #5
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You should read up on the K-System used by most studios now. You can find some references to it on line. Bob goes into it in great detail in his book which is well worth the investment of reading.

Here's a brief outline on it.

K-System Metering 101

This is from his book which has more details on the How and Why the system is used.

https://www.digido.com/portfolio-ite...ctices-part-2/
Old 21st April 2017
  #6
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A monitor controller needs to be transparent (what's your budget?) and is probably something to consider only if you really need the added functionality it would give you i.e. switching between multiple speaker sets, easily moving from stereo to mono, dim, talkback, headphones etc. It doesn't seem you're at that stage yet and would seem an unnecessary expense.

You're still going to need to set good levels coming out of your interface and into the controller and then the speakers, so I can't really see how it would solve your problem.

So to add to my earlier post, good gain-staging between your interface and speakers should be all you need for now to hear your signal cleanly. And that's kinda brings us back to my original statement about calibrating your speakers.
Old 21st April 2017
  #7
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Quote:
Originally Posted by Animesh Raval View Post
I really am not an expert in this area, so anyone more knowledgeable don't beat me up...

Attenuating your monitoring level digitally won't compromise the quality and resolution of the signal. The sample rate (number of samples taken each second) and number of bits of information in each sample (i.e. the resolution, 24-bits) doesn't change just because you're turning down the level. So if I sample a sine wave and I attenuate it by 50% I'm just mathematically halving all my amplitude sample values. I still have the same number of samples per second and number of bits of information per sample, that I did for the original signal - so it should be a perfect reconstruction of the original sine wave, but just at half the amplitude. I hope that makes sense?

In talking about K-metering systems etc, we're suggesting a way to calibrate your listening volumes that will help you be more discerning when mixing - i.e. the optimum resolution for your ears!

Sorry man, but this is completely wrong. Attenuating the signal inside the audio interface's software DOES reduce the bit depth of the outgoing signal. In some very rare cases, the output level trims are controlling an attenuator that comes after the D>A converter inside the interface, but this is not the case with the interface that rds is using.

What rds is doing by turning down the output faders / trims on his audio interface software is reducing the level of the digital signal coming from his DAW software before it goes to the D>A converters, thereby providing a signal of reduced bit depth for those converters.

rds - what you assumed was correct, as you discovered by experimentation. It is always better to leave any software output level trims at their maximum level and then reduce the signal level going into your monitors by analog means.

An analog monitor controller will be what you want. It doesn't have to be an $8,000 Grace Designs unit - it can be a $600 Drawmer unit, or an $89 TC Electronics Level Pilot, or anything similar depending on the features you need in terms of hookups for multiple sets of speakers, mono+dim buttons, inputs for alternate playback units like iPhones etc.

This same scenario is still an issue with the new breed of monitors that have digital audio inputs - if they don't have a means of controlling the level of their internal amplifiers (like a remote volume knob) then the user must attenuate the level of the digital signal going out of the audio interface - and this is bad, bad, bad. Doing this definitely means you will be listening to a signal that is effectively 13 bits or whatever, instead of the nice and fluffy 24 bits being generated by your software. I use Dynaudio AIR monitors which have digital input and an included remote volume control - this knob connects directly to the monitors via CAT5 cable and digitally controls the level of their amplifiers - it does NOT affect the digital audio stream itself. This is the correct way to do this, so beware of any monitor system that does have digital audio inputs but does not have a means of remotely controlling its amp levels.

So, rds, what's going on in your setup is that your speakers are expecting an analog signal coming in, and when that analog signal is at max level the speakers will basically be playing as loudly as they can go without exploding. Adjusting the monitor sensitivity, whether by a knob on the back or a +4 / -10 switch, is providing a small range of adjustment intended for precise calibration and lineup with the gear that's sending the signal - not as an overall "volume control".

So, yeah..... what you initially suspected is correct, rds. Get thee to the music store and grab an analog monitor level control of some kind. Then set your output levels in the audio interface software back up to their default (max) position and use your shiny new level controller as your "volume knob".
Old 21st April 2017
  #8
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Quote:
Originally Posted by charlieclouser View Post
Sorry man, but this is completely wrong. Attenuating the signal inside the audio interface's software DOES reduce the bit depth of the outgoing signal. In some very rare cases, the output level trims are controlling an attenuator that comes after the D>A converter inside the interface, but this is not the case with the interface that rds is using.

What rds is doing by turning down the output faders / trims on his audio interface software is reducing the level of the digital signal coming from his DAW software before it goes to the D>A converters, thereby providing a signal of reduced bit depth for those converters.

rds - what you assumed was correct, as you discovered by experimentation. It is always better to leave any software output level trims at their maximum level and then reduce the signal level going into your monitors by analog means.

An analog monitor controller will be what you want. It doesn't have to be an $8,000 Grace Designs unit - it can be a $600 Drawmer unit, or an $89 TC Electronics Level Pilot, or anything similar depending on the features you need in terms of hookups for multiple sets of speakers, mono+dim buttons, inputs for alternate playback units like iPhones etc.

This same scenario is still an issue with the new breed of monitors that have digital audio inputs - if they don't have a means of controlling the level of their internal amplifiers (like a remote volume knob) then the user must attenuate the level of the digital signal going out of the audio interface - and this is bad, bad, bad. Doing this definitely means you will be listening to a signal that is effectively 13 bits or whatever, instead of the nice and fluffy 24 bits being generated by your software. I use Dynaudio AIR monitors which have digital input and an included remote volume control - this knob connects directly to the monitors via CAT5 cable and digitally controls the level of their amplifiers - it does NOT affect the digital audio stream itself. This is the correct way to do this, so beware of any monitor system that does have digital audio inputs but does not have a means of remotely controlling its amp levels.

So, rds, what's going on in your setup is that your speakers are expecting an analog signal coming in, and when that analog signal is at max level the speakers will basically be playing as loudly as they can go without exploding. Adjusting the monitor sensitivity, whether by a knob on the back or a +4 / -10 switch, is providing a small range of adjustment intended for precise calibration and lineup with the gear that's sending the signal - not as an overall "volume control".

So, yeah..... what you initially suspected is correct, rds. Get thee to the music store and grab an analog monitor level control of some kind. Then set your output levels in the audio interface software back up to their default (max) position and use your shiny new level controller as your "volume knob".
To quote you, "Sorry man, but this is completely wrong." The bit reduction would still leave you with well over 20 bits, in other words not even close to an issue. Compared to that, the analog products you suggested will blanket your signal.
Old 21st April 2017
  #9
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Originally Posted by Deleted User View Post
To quote you, "Sorry man, but this is completely wrong." The bit reduction would still leave you with well over 20 bits, in other words not even close to an issue. Compared to that, the analog products you suggested will blanket your signal.

Yeah.... no. If rds is trimming the output of his sound card by 40db that's basically shaving off nearly 7 bits' worth of signal - so if he's got a fully-normalized 24-bit file that's being trimmed by 40db in software, the output will be around 17 bits - or less if he's not almost clipping at the output of his DAW. I don't know where you're getting "well over 20 bits" from.

Each bit is worth 6db.

144db (theoretical max dynamic range of 24 bit signals) minus 40db (the amount of attenuation he's using in his driver's software) equals 104db = 17.3333 bits.

Now he's chomped off a huge chunk of his signal. Whether that matters to him depends on his music and ears - but he's already heard a difference just by fiddling with his speakers' sensitivity setting vs output level trim on his driver software.

And those analog products won't "fully mask the signal". Most are just passive controls that drop signal to ground. Attenuating a digital signal before D>A is not best practice for digital audio.

What means do you use for level control?
Old 21st April 2017
  #10
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Quote:
Originally Posted by charlieclouser View Post
Yeah.... no. If rds is trimming the output of his sound card by 40db that's basically shaving off nearly 7 bits' worth of signal - so if he's got a fully-normalized 24-bit file that's being trimmed by 40db in software, the output will be around 17 bits - or less if he's not almost clipping at the output of his DAW. I don't know where you're getting "well over 20 bits" from.

Each bit is worth 6db.

144db (theoretical max dynamic range of 24 bit signals) minus 40db (the amount of attenuation he's using in his driver's software) equals 104db = 17.3333 bits.

Now he's chomped off a huge chunk of his signal. Whether that matters to him depends on his music and ears - but he's already heard a difference just by fiddling with his speakers' sensitivity setting vs output level trim on his driver software.

And those analog products won't "fully mask the signal". Most are just passive controls that drop signal to ground. Attenuating a digital signal before D>A is not best practice for digital audio.

What means do you use for level control?
Your math is right this time, not 13 bit like you said before. Honestly, I would way rather have a clean 17 bit signal than the TC level pilot, etc.

The number I used referred to a more normal setup. I agree that 40 dB of digital attenuation is huge and should be addressed, which is why I asked earlier if he can attenuate his monitors more (or change the d/a calibration settings or something.) Mine (Adam S2x) go down 22 dB. I only need about 12 dB max digital attenuation to bring a hot mix down to a normal level. So that's a 22 bit signal. If I want to listen real quietly we're talking 20 bits.

Last edited by Deleted User; 21st April 2017 at 11:00 PM..
Old 21st April 2017
  #11
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Quote:
Originally Posted by charlieclouser View Post
I use Dynaudio AIR monitors which have digital input and an included remote volume control - this knob connects directly to the monitors via CAT5 cable and digitally controls the level of their amplifiers - it does NOT affect the digital audio stream itself. This is the correct way to do this, so beware of any monitor system that does have digital audio inputs but does not have a means of remotely controlling its amp levels.
Unless you have information that says otherwise, the Dynaudio AIR speakers are digital throughout. That includes the volume control on the way to a Class D (PWM) amplifier found on those speakers. There isn't a stage of conversion from D to A to do volume adjustment (post or pre DSP work.) You are, in fact, attenuating a digital signal before output through the PWM amps to the drivers. It was one of the genius ideas of the TC/Dynaudio collaboration of the time: make everything digital on the way in and then distribute and control all functions digitally. The remote (or the software) just manages the same digital attenuation.
Old 21st April 2017
  #12
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Quote:
Originally Posted by pentagon View Post
Unless you have information that says otherwise, the Dynaudio AIR speakers are digital throughout. That includes the volume control on the way to a Class D (PWM) amplifier found on those speakers. There isn't a stage of conversion from D to A to do volume adjustment (post or pre DSP work.) You are, in fact, attenuating a digital signal before output through the PWM amps to the drivers. It was one of the genius ideas of the TC/Dynaudio collaboration of the time: make everything digital on the way in and then distribute and control all functions digitally. The remote (or the software) just manages the same digital attenuation.
Yeah, I know. Class D amps in the AIR speakers, yadda yadda yadda. I have been using the AIRs for almost 15 years and have over thirty (!) individual speakers spread across a number of rooms. For the purposes of simplifying my description of why it's bad practice to attenuate digital signals before any D>A I didn't get into that. I only mentioned that because we are starting to see some monitors that do have digital inputs but do not have a means of controlling their amp levels, and thus require the user to attenuate the level of the digital output coming from their DAW.

This is bad.
Old 21st April 2017
  #13
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Told you guys I was no expert!! apologies to the OP if my posts were misleading.

I am now however slightly confused by the above discussion... I don't wish to hijack this thread, but is this then the case with all interfaces? That if you aren't outputting at close to 0 dBfs from your DAW, you are introducing reduced bit depth to your converters on the way out to the studio monitors? Apologies if this is a dumb question!
Old 21st April 2017
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Quote:
Originally Posted by charlieclouser View Post
Yeah, I know. Class D amps in the AIR speakers, yadda yadda yadda. I have been using the AIRs for almost 15 years and have over thirty (!) individual speakers spread across a number of rooms. For the purposes of simplifying my description of why it's bad practice to attenuate digital signals before any D>A I didn't get into that. I only mentioned that because we are starting to see some monitors that do have digital inputs but do not have a means of controlling their amp levels, and thus require the user to attenuate the level of the digital output coming from their DAW.

This is bad.
Wait, so....you're doing the same digital attenuation you're criticizing?! In fact much more than me, since I'm doing 20 dB of analog attenuation on the Adam controls and the same or more on my Auratone/headphone amp.
Old 21st April 2017
  #15
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Originally Posted by Deleted User View Post
Your math is right this time, not 13 bit like you said before. Honestly, I would way rather have a clean 17 bit signal than the TC level pilot, etc.

The number I used referred to a more normal setup. I agree that 40 dB of digital attenuation is huge and should be addressed, which is why I asked earlier if he can attenuate his monitors more (or change the d/a calibration settings or something.) Mine (Adam S2x) go down 22 dB. I only need about 12 dB max digital attenuation to bring a hot mix down to a normal level. So that's a 22 bit signal. If I want to listen real quietly we're talking 20 bits.

Really? You'd rather truncate a digital signal than use an analog monitor level controller? Plugged into an Echo AudioFire-4? That Echo interface doesn't have "d/a calibration" - it's a simple, ten-year-old 4x4 FireWire interface fer chrissakes. It doesn't have trimpots like a 192 does... so if you're talking about a "more normal setup", well.... that's not what rds has, and that's not the question he asked. He had a specific question about his specific setup and had experienced a difference in sound based on an obvious (and correct) assumption.

Attenuating a digital signal before D>A is bad. Period.

Using the master output level of your DAW or your driver's control panel software to adjust your listening level is bad. Period.

How bad depends on how much attenuation - and 40db is a lot.

For rds's use, a simple analog monitor level controller will be fine.

Level Pilot, NanoPatch, etc. Cheap - n - cheerful, just like his audio interface.
Old 21st April 2017
  #16
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Quote:
Originally Posted by Animesh Raval View Post
Told you guys I was no expert!! apologies to the OP if my posts were misleading.

I am now however slightly confused by the above discussion... I don't wish to hijack this thread, but is this then the case with all interfaces? That if you aren't outputting at close to 0 dBfs from your DAW, you are introducing reduced bit depth to your converters on the way out to the studio monitors? Apologies if this is a dumb question!
Your assumption is correct. Trimming the output level of your DAW, whether by its master fader or by some control in the driver software, will result in less than 24 bits or whatever going to the D>A.

Unless you have specific knowledge that a particular audio interface's driver trim controls are trimming the analog output level, the above is exactly what's happening.

Always leave your master output level of your DAW at zero, and get right up there close to full scale and that's how things should work.
Old 21st April 2017
  #17
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Thank you for your explanation it's much appreciated! So if I understand right, I'll do my gain staging nicely in the DAW when mixing and leave peaks at max -6dBfs for the mastering guy to go their thing. In terms of me monitoring my mix, I'm leaving 6dB of headroom so losing only 1 bit on the way out to my monitors, which is still way too good for me to even be the slightest bit concerned I might not be hearing the mix perfectly...
Old 21st April 2017
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Originally Posted by Deleted User View Post
Wait, so....you're doing the same digital attenuation you're criticizing?! In fact much more than me, since I'm doing 20 dB of analog attenuation on the Adam controls and the same or more on my Auratone/headphone amp.
Yes, technically the Dynaudios are trimming the bit depth somewhere inside their electronics - how the hell else would the level be controlled in a Class-D amp? But I'm not trimming the digital signal anywhere before a 24-bit D>A circuit. And that's what we're talking about here - reducing the bit depth of a digital signal before it hits a D>A will result in effectively lowering the bit depth of the signal that's being output. Period.

I don't have to trim my digital output, which means no confusion about what's going on when that same digital output is used for multiple purposes - feeding the monitors at the same time as it's feeding a separate ProTools layback recorder for instance (which is what I do every day). I don't have to wonder if I'm "really at zero" because I trimmed my digital level somewhere other than with the Dynaudio's volume control.

I've been recording digitally since 1983. The Sony PCM-601 encoder hooked up to a Sony BVU-950 gave us 14 bits on a good day - I know what it's like to deal with not enough bit depth, and how you've got to get right up there as close to full-scale as you can. I've been down every digital dusty trail there is. Now that every bedroom has a 24-bit setup, people don't need to be as careful as we had to be back in the bad old days.

But the math hasn't changed - and neither have best practices for digital audio. Reducing the level of a digital signal before D>A reduces the effective bit depth of the signal that's being output. Simple.
Old 21st April 2017
  #19
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Quote:
Originally Posted by Animesh Raval View Post
Thank you for your explanation it's much appreciated! So if I understand right, I'll do my gain staging nicely in the DAW when mixing and leave peaks at max -6dBfs for the mastering guy to go their thing. In terms of me monitoring my mix, I'm leaving 6dB of headroom so losing only 1 bit on the way out to my monitors, which is still way too good for me to even be the slightest bit concerned I might not be hearing the mix perfectly...
Yeah, your assumptions are correct. I mean, in a 24-bit system you don't have to be as careful about that last bit as some of this discussion might indicate, but as long as you understand the principles correctly (and you do), then you'll be okay.

But, yeah... whacking 40db off the top of your signal with the output control in your audio interface's driver control panel, while the signal is still digital, is not a great way to do it - that's what rds was dealing with in his original post - and it's not a great way to go.
Old 21st April 2017
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Quote:
Originally Posted by charlieclouser View Post
Yes, technically the Dynaudios are trimming the bit depth somewhere inside their electronics - how the hell else would the level be controlled in a Class-D amp? But I'm not trimming the digital signal anywhere before a 24-bit D>A circuit. And that's what we're talking about here - reducing the bit depth of a digital signal before it hits a D>A will result in effectively lowering the bit depth of the signal that's being output. Period.

I don't have to trim my digital output, which means no confusion about what's going on when that same digital output is used for multiple purposes - feeding the monitors at the same time as it's feeding a separate ProTools layback recorder for instance (which is what I do every day). I don't have to wonder if I'm "really at zero" because I trimmed my digital level somewhere other than with the Dynaudio's volume control.

I've been recording digitally since 1983. The Sony PCM-601 encoder hooked up to a Sony BVU-950 gave us 14 bits on a good day - I know what it's like to deal with not enough bit depth, and how you've got to get right up there as close to full-scale as you can. I've been down every digital dusty trail there is. Now that every bedroom has a 24-bit setup, people don't need to be as careful as we had to be back in the bad old days.

But the math hasn't changed - and neither have best practices for digital audio. Reducing the level of a digital signal before D>A reduces the effective bit depth of the signal that's being output. Simple.
LOL, so you're digitally attenuating down to the same 17 bit or so as the OP. I'm at 22 or 23 bit. This has been a hilarious episode of those in glass houses shouldn't throw stones.
Old 21st April 2017
  #21
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Quote:
Originally Posted by charlieclouser View Post
Yeah, your assumptions are correct. I mean, in a 24-bit system you don't have to be as careful about that last bit as some of this discussion might indicate, but as long as you understand the principles correctly (and you do), then you'll be okay.

But, yeah... whacking 40db off the top of your signal with the output control in your audio interface's driver control panel, while the signal is still digital, is not a great way to do it - that's what rds was dealing with in his original post - and it's not a great way to go.
Honestly, thank you very much for educating me! I really appreciate it!
Old 21st April 2017
  #22
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Originally Posted by Deleted User View Post
LOL, so you're digitally attenuating down to the same 17 bit or so as the OP. I'm at 22 or 23 bit. This has been a hilarious episode of those in glass houses shouldn't throw stones.
Well, that's not exactly how Class-D amps work, so it's not an apples-to-apples comparison. Reducing the bit depth of the AES signal going into the Dynaudios has a different effect than what their level control is actually doing in terms of apparent or actual bit depth. To get deeper into that issue is a question for Brian Thompson at TNR Electronics, the Dynaudio service tech that keeps all of my systems up to spec and knows those amp modules like the back of his hand.

Anyway, none of that has any relevance to OP's question about lowering the output level by forty freaking db in the digital domain, on the way out of his Echo AudioFire-4 interface, which is still not the "correct" or best way to do things - as he could plainly hear when fiddling with his speakers and audio interface.

But, hey, I'm just trying to help a dude who had a simple question about a simple setup.
Old 22nd April 2017
  #23
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Originally Posted by charlieclouser View Post
Well, that's not exactly how Class-D amps work, so it's not an apples-to-apples comparison. Reducing the bit depth of the AES signal going into the Dynaudios has a different effect than what their level control is actually doing in terms of apparent or actual bit depth. To get deeper into that issue is a question for Brian Thompson at TNR Electronics, the Dynaudio service tech that keeps all of my systems up to spec and knows those amp modules like the back of his hand.

Anyway, none of that has any relevance to OP's question about lowering the output level by forty freaking db in the digital domain, on the way out of his Echo AudioFire-4 interface, which is still not the "correct" or best way to do things - as he could plainly hear when fiddling with his speakers and audio interface.

But, hey, I'm just trying to help a dude who had a simple question about a simple setup.
Well, it may be relevant given that you say your system sounds good and you may be working at the same 17 bits as the OP. But I agree that would be too much digital attenuation for me. However I think a small amount is perfectly fine and better than a cheap analog product, so I think your severe warning about it is overboard and likely being majorly violated by you yourself.
Old 22nd April 2017
  #24
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Quote:
Originally Posted by Deleted User View Post
Well, it may be relevant given that you say your system sounds good and you may be working at the same 17 bits as the OP. But I agree that would be too much digital attenuation for me. However I think a small amount is perfectly fine and better than a cheap analog product, so I think your severe warning about it is overboard and likely being majorly violated by you yourself.

Oh jeezus freaking keeeerist.

You agree that 40db is too much digital attenuation - which was the whole freaking point of OP's post! So he's not attenuating by a "small amount" - 40db is huge, and will be a lot worse than a simple analog level control.

A passive volume attenuator is the simplest possible audio circuit. It'a an audio taper pot that dumps signal to ground. You can build one for $20 in parts that will sound a whole lot better than chomping 40db off a signal in the digital domain. What do you think the level trim knob /switch on your speakers is doing? Exactly that. It's dumping signal to ground. So if speakers have any kind of level trim knob, there's already a "cheap analog product" in the circuit.

And your Adam S2x monitors definitely have a whole shedload of analog controls on the front - bass, treble, and level controls - that's exactly what's in a cheap analog level controller, plus a healthy chunk of RC network circuitry to do the eq - so that's already way more than what's in something like a NanoPatch or Level Pilot.

The OP needs a simple analog level controller. It's not going to "mask the sound" any more than the knobs on the front of your Adams will.
Old 22nd April 2017
  #25
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No I'll reiterate what I suggested to the OP in my first reply - I think the better place to adjust first is the monitor sensitivity. After that, then yes an analog control may be needed. In my case, not.
Old 22nd April 2017
  #26
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I'd like to highlight charlieclouser for patience and clarity in this thread. It requires dogged persistence to not allow incorrect information be the last word. You don't get much for spending the extra time, but you deserve at least thanks and respect.
Old 22nd April 2017
  #27
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charlieclouser's Avatar
 

Quote:
Originally Posted by Bushman View Post
I'd like to highlight charlieclouser for patience and clarity in this thread. It requires dogged persistence to not allow incorrect information be the last word. You don't get much for spending the extra time, but you deserve at least thanks and respect.
Thanks Bushman.

I could tell from rds's post what he was up against, and since his ears had already told him most of the answer, it's a short journey from there to "spend $100 on a NanoPatch or Level Pilot and problem solved". Given the price point of his AudioFire interface, he's probably not going to be picking up a Grace, Antelope, or Dangerous monitor controller!

Although rds didn't explicitly say it, it was kind of obvious that his speakers (like most) don't have level controls, other than perhaps a -10 / +4 switch or a trim pot with limited range - so suggestions to use those to control listening level probably won't do him much good. Plus those adjustments are usually on the back of the speaker - not convenient, hence his use of the output level controls on his audio interface driver software (or DAW) - which will have the effect of reducing the effective bit depth of the signal being sent to the speakers.

In some cases, it won't matter all that much - but 40db (aka 7.333 bits) is a healthy chunk.

rds is correct in what his intuition, experience, and ears told him.
Old 22nd April 2017
  #28
Gear Addict
 

Wow.. i didn't expect so much replies ! Thank to you all

Quote:
Originally Posted by Animesh Raval View Post
Hello mate, I just posted a link to this article somewhere else and think it might help you:

I hope this helps rather than confuses!
To be honest i'm not into the technical side at all, my strengh are more into production and mixing. So yea, basically when technical stuff come into play i'm more like





Quote:
Originally Posted by Animesh Raval View Post
Told you guys I was no expert!! apologies to the OP if my posts were misleading.
No worries, at least this thread help you too to get things clarifyed as well so was worth it

Quote:
Originally Posted by Deleted User View Post
Best practice to keep the signal cleaner is to avoid an analog level monitor control. Turn your monitors sensitivity down all the way and do any other attenuation digitally.
My monitor sensitivity is already all the way down at 10.5dB exactly and still digitaly above -40dB it start to become to loud to work in long sessions. So something has to be done in a way or another because -40dB digital attenuation is way to much from the good advise from this thread.

Quote:
Originally Posted by Bushman View Post
The answer doesn't seem to address the question. I think the question might be, when you digitally attenuate your monitor level, are you compromising the quality and resolution of the digital signal?
Spot on.

Quote:
Originally Posted by Bushman View Post
I'd like to highlight charlieclouser for patience and clarity in this thread. It requires dogged persistence to not allow incorrect information be the last word. You don't get much for spending the extra time, but you deserve at least thanks and respect.
Indeed, thanks Charlie and everyone else who contribute to this thread

Quote:
Originally Posted by wrgkmc View Post
You should read up on the K-System used by most studios now. You can find some references to it on line. Bob goes into it in great detail in his book which is well worth the investment of reading.

Here's a brief outline on it.

K-System Metering 101

This is from his book which has more details on the How and Why the system is used.

https://www.digido.com/portfolio-ite...ctices-part-2/
Thanks for the links, i'll give a look.

Quote:
Originally Posted by charlieclouser View Post

rds - what you assumed was correct, as you discovered by experimentation. It is always better to leave any software output level trims at their maximum level and then reduce the signal level going into your monitors by analog means.
So it seems to make sense, thank for confirming as well Charlie.


Quote:
Originally Posted by charlieclouser View Post
An analog monitor controller will be what you want. It doesn't have to be an $8,000 Grace Designs unit - it can be a $600 Drawmer unit, or an $89 TC Electronics Level Pilot, or anything similar depending on the features you need in terms of hookups for multiple sets of speakers, mono+dim buttons, inputs for alternate playback units like iPhones etc.
It seems analog control it's going the way to go indeed. Actually i use only main monitors and one single mixcube for mono purpose. The thing is that the mixcube has an analog level trim on its back and its hooked to the output 3-4. So i can raise the digital 3-4 output level all the way up and controlling the volume from the mixcube, so no problem here.

So basically i'll need just a level control for my main monitors only and i don't need fancy options like mute or mono things since i can do this in my interface and daw already.



Quote:
Originally Posted by charlieclouser View Post
So, rds, what's going on in your setup is that your speakers are expecting an analog signal coming in, and when that analog signal is at max level the speakers will basically be playing as loudly as they can go without exploding. Adjusting the monitor sensitivity, whether by a knob on the back or a +4 / -10 switch, is providing a small range of adjustment intended for precise calibration and lineup with the gear that's sending the signal - not as an overall "volume control".

So, yeah..... what you initially suspected is correct, rds. Get thee to the music store and grab an analog monitor level control of some kind. Then set your output levels in the audio interface software back up to their default (max) position and use your shiny new level controller as your "volume knob".
Yes the sensitivity isn't a volume control but in this case and for experiment purposes i tryed to found a way to lower the level signal somewhere to be able to raise the digital one to see if any improvment occurs and that's how i notice a slight difference.

Quote:
Originally Posted by charlieclouser View Post

I could tell from rds's post what he was up against, and since his ears had already told him most of the answer, it's a short journey from there to "spend $100 on a NanoPatch or Level Pilot and problem solved". Given the price point of his AudioFire interface, he's probably not going to be picking up a Grace, Antelope, or Dangerous monitor controller!
Indeed

So now i'm leaning toward the Jbl nano patch since i just need to control my main monitors and i need TRS outputs only so it fit my need plus as i said i prefer no fancy other options than a volume knob control. I think the lesser electronics components the better to avoid extra signal parasit ( just my idea ) and if passive even better i guess.

If anyone has any other suggestion or experience with a good volume controller feel free to share.

Also if i set an analog volume controller i won't need to crank the sensitivity all the way down like now, so should i put my monitors sensitivity to its 0dB initial position or i can leave it at -10.5dB ? What recommendation here ?


Thanks again guys
Old 22nd April 2017
  #29
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charlieclouser's Avatar
 

The JBL nPatch+ (which is a rebranding of the tried and true SM Pro Audio NanoPatch) is definitely a good bet. Totally passive, not expensive, and it's got the connectors you need. Job done!

Here's what it says on the product's web site:

"One of the benefits of passive volume attenuation is that you no longer have to control volumes with your software’s master volume fader. Reducing levels from your software only reduces the audio bit depth. It is much more appropriate to keep your software masters at unity and attenuate the audio to your active monitors with a passive controller like the Nano Patch+. This way you can maintain maximum audio resolution and attenuate your monitoring volume with the confidence that the sonic integrity of your audio is not compromised.

No active circuitry, buffers, transistors, batteries, or power supply of any kind can interfere with the main passive output of the Nano Patch+. It simply delivers attenuated signal. Pure, simple transparent."

There ya go.
Old 22nd April 2017
  #30
Gear Addict
 

Quote:
Originally Posted by charlieclouser View Post
The JBL nPatch+ (which is a rebranding of the tried and true SM Pro Audio NanoPatch) is definitely a good bet. Totally passive, not expensive, and it's got the connectors you need. Job done!

Here's what it says on the product's web site:

"One of the benefits of passive volume attenuation is that you no longer have to control volumes with your software’s master volume fader. Reducing levels from your software only reduces the audio bit depth. It is much more appropriate to keep your software masters at unity and attenuate the audio to your active monitors with a passive controller like the Nano Patch+. This way you can maintain maximum audio resolution and attenuate your monitoring volume with the confidence that the sonic integrity of your audio is not compromised.

No active circuitry, buffers, transistors, batteries, or power supply of any kind can interfere with the main passive output of the Nano Patch+. It simply delivers attenuated signal. Pure, simple transparent."

There ya go.
Fantastic

Just as side note, when i notice the difference in sound quality it was only when listening some of my mp3 which are 16bit. I mean it wasn't in my daw but still i felt this difference even on 16bit mp3.

Also have you any suggestion about this ?

Quote:
Originally Posted by rds View Post
Also if i set an analog volume controller i won't need to crank the sensitivity all the way down like now, so should i put my monitors sensitivity to its 0dB initial position or i can leave it at -10.5dB ? What recommendation here ?
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