Max digital volume and Monitors

[ecc83]

Attached is the output control panel for the Jul@ card. The "faders" can be seen to be at max (0dB) by default and indeed they are so set for all other views of the control panel.

You can of course overload the card's INPUT but this is under the control of whatever the sound source is, e.g. a mixer.

To tie it all up: Record 24 bits (44.1kHz unless you like buying hard drives) and average as Bob says at -18dBFS, even as low as -25dBFS will not hurt if the program is beyond your control, a live band for example.

The "mixer" in the DAW software should default to "0" with the usual +10dB available on each channel. I am NO mixer! But I believe the process should be more "subtractive" than "additive"?

Control the card's output with an analogue device, i.e. a pot in a tin.

This, ^ all assumes ASIO drivers and control by the soundcard and the DAW. If you let Windows into the equation all level settings go to H in a pram!

Dave.

 

[iqi616]

This...

There's no exact conversion between dB(u) and dB(FS) but most manufacturers assume you should have 18 dB headroom above the zero level in analogue. for this reason, digital "zero" tends to be considered around -18dB. This means that to equate to recording in analogue you probably want your "average" around -18 with peaks in the -10 to -6 range (depending on the material).

And especially this...

The myth of using every bit is just that...a myth.

The "use every bit" came from people who didn't have the first idea of how digital audio actually functions.

A lot of the bollocks is a hang-over from the early days of digital when most "16-bit" gear was at best 12-bit + noise and from narrow-track analogue where it was a case of squeezing the signal in between the noise floor and clipping. None of that applies to 24-bit.

 

[iqi616]

... by that I meant Bobbsy is absolutely spot-on. [Internet Exploder won't let me edit my post.]

 

[Pinkertel]

Now I know what you are talking about with the volume changing the eq curve. That was only a feature on home stereo systems. It was an attempt to replace the loudness button.

I don't know if they still do that anymore, but that was cheap home audio, never having anything to do with and studio or pro audio stuff.

Aah alright. So generally software volumes don't do that.

"Is that I just want the professional standard correct way of doing things. And I always thought that a max volume on the computer is a standard practice"

No it's not. I think it may be time for some basics.

There are several different dB scales in use for metering. Analogue tends to use dB(u) or dB(VU). On these scales, the zero level is a semi-arbitrary voltage (David can give his lecture about 600 ohm loads if he wants) but the result is that any decent gear has plenty of headroom above that zero level for you to use while recording.

Computers/digital, on the other hand, uses dB(FS) (Full Scale). This means that the zero level here is the absolute maximum the system can take without running out of bits. If you're recording at 24 bit, a sample at 0dB(FS) will be 111111111111111111111111, i.e there's nowhere else to go.

There's no exact conversion between dB(u) and dB(FS) but most manufacturers assume you should have 18 dB headroom above the zero level in analogue. for this reason, digital "zero" tends to be considered around -18dB. This means that to equate to recording in analogue you probably want your "average" around -18 with peaks in the -10 to -6 range (depending on the material). The myth of using every bit is just that...a myth.

If you go too high, you gain no quality, run the risk of digital clipping and, since levels are cumulative when you mix, you'll just have to turn everything down anyway.

I get what you are saying. I shouldn't say "max volume" I should say 0db or digital 0. "any decent gear" So the monitors I bought don't have enough headroom?

I understand what you mean. With more tracks in the mix I have to turn things down anyway. So I shouldn't be to fidgety with cutting dbs with the last master fader of my soundcard before sending them to my monitors(still I like them bits ). But it is curious. Other stuff like my amp and my mixing board never clipped. So they have that 18db required headroom. Why not these active monitors? And if the digital zero is considered to be around -18db. Shouldn't an active monitor be capable of handling this?

Attached is the output control panel for the Jul@ card. The "faders" can be seen to be at max (0dB) by default and indeed they are so set for all other views of the control panel.

You can of course overload the card's INPUT but this is under the control of whatever the sound source is, e.g. a mixer.

To tie it all up: Record 24 bits (44.1kHz unless you like buying hard drives) and average as Bob says at -18dBFS, even as low as -25dBFS will not hurt if the program is beyond your control, a live band for example.

The "mixer" in the DAW software should default to "0" with the usual +10dB available on each channel. I am NO mixer! But I believe the process should be more "subtractive" than "additive"?

Control the card's output with an analogue device, i.e. a pot in a tin.

This, ^ all assumes ASIO drivers and control by the soundcard and the DAW. If you let Windows into the equation all level settings go to H in a pram!



Dave.

So I should leave it at 0db (or like I called it earlier max volume).

"more "subtractive" than "additive"?"

My thought exactly.

"This, ^ all assumes ASIO drivers and control by the soundcard and the DAW. If you let Windows into the equation all level settings go to H in a pram!"

Yeah I have no idear what windows does.... or controls. I just have the little speaker icon in the right corner like always. And it had a volume control( I always leave it on max and if I ever want to control the volume I do it with the esi control panel). If I click on it it opens the mixer. On the left it says "device" and it is the juli@ with a volume control and on the right from that there are the volume controls for system sounds and application I might me running. If I open up sound in control panel. It just says that the juli@ is the standard device for recording and playing. The standard soundchip of my motherboard I just turned off by the way in device manager.

So yeah no idea or how to turn it off.

 

[garww]

The motherboard devices can often be disabled in BIOS, but Windows device manager should be fine

 

[Farview]

The volume control on the soundcards control panel should be set at 0 (unity) and never touched again. If your mix is clipping, the master output in your DAW is what you need to bring down. If the signal is clipping in your DAW, the soundcard control can't fix it. It will simply lower the volume of the clipped signal.

You still haven't confirmed that you are using the ASIO driver. Doing so would take the windows mixer out of the equation and help with latency and a bunch of other things.

Digital zero is not -18dbfs. Analog zero (0dbVU) is about -18dbfs in the DAW. In other words, line level in analog will read about -18dbfs in the computer. (about half way up the meter) So, if you have really hot signals that are bouncing around 0dbfs in the computer, you are sending a signal 18db higher than line level to the speakers.

 

[JR15108]

If you are just using the internal sound from the computer, get a recording interface. It will bypass all the windows garbage, output a line level signal, give you a volume control, zero latency monitoring, etc...

If you do have an interface, the windows mixer should not be involved at all. You should be using balanced TRS cables to go to the monitors and everything should just work out.

Thank you, Farview - just had a light bulb moment. I have been recording and mixing in different locations and it never occurred to me to use an AI at the latter, where only playback was necessary. (Eegit?)
Been running from 1/8 stereo pc into RCAs on not so great, but good 'nuff Audix monitors where I mix.
Will grab a loose AI to test this revelation asap.

JR

 

[Pinkertel]

The volume control on the soundcards control panel should be set at 0 (unity) and never touched again. If your mix is clipping, the master output in your DAW is what you need to bring down. If the signal is clipping in your DAW, the soundcard control can't fix it. It will simply lower the volume of the clipped signal.

We are still talking about monitors that are clipping. Who said anything about clipping in the daw. There is no problem.

You still haven't confirmed that you are using the ASIO driver. Doing so would take the windows mixer out of the equation and help with latency and a bunch of other things.

In adobe audition ofcourse I selected the asio driver. Again for windows I don't know. I don't know what to tell you. Nowhere there is a an option for selecting the asio driver. I just asumed it does. Windows mixer, just has a volume bar on the right for "device" (juli@) and next to them volume bars for the applications. Sound at control panel just says the juli@ is the main device for recording and sound and controls everything. So I still don't know what you mean or how to change it.

Digital zero is not -18dbfs. Analog zero (0dbVU) is about -18dbfs in the DAW. In other words, line level in analog will read about -18dbfs in the computer. (about half way up the meter) So, if you have really hot signals that are bouncing around 0dbfs in the computer, you are sending a signal 18db higher than line level to the speakers.

Aaah ok. I am a bit confused when to put the minus sign. It is a matter of perspective I guess? So active monitors are not made to be driven by 0db coming from the computer? But almost everyone hooks it up to a computer. And shouldn't a mixing board also clip when sending a 0db digital signal?

 

[Farview]

Aaah ok. I am a bit confused when to put the minus sign. It is a matter of perspective I guess? So active monitors are not made to be driven by 0db coming from the computer? But almost everyone hooks it up to a computer. And shouldn't a mixing board also clip when sending a 0db digital signal?

It isn't a matter of perspective. All digital levels are below 0. In digital 0dbfs is the ceiling, there is no higher level. Digital meters track the peak level of the signal.

In analog, 0dbVU is line level. Also, VU meters are tracking the average level of the signal, not the peak level. The peaks will be much higher than what the meter reads. The idea is to keep the meter reading around 0dbVU, and the peaks will generally fit into the headroom above 0dbVU.

This is why your average level in digital needs to sit around -18dbfs. It leave room for the peaks, without hitting 0dbfs.

 

[Pinkertel]

It isn't a matter of perspective. All digital levels are below 0. In digital 0dbfs is the ceiling, there is no higher level. Digital meters track the peak level of the signal.

In analog, 0dbVU is line level. Also, VU meters are tracking the average level of the signal, not the peak level. The peaks will be much higher than what the meter reads. The idea is to keep the meter reading around 0dbVU, and the peaks will generally fit into the headroom above 0dbVU.

This is why your average level in digital needs to sit around -18dbfs. It leave room for the peaks, without hitting 0dbfs.

Aah ok it is starting to make sense.

But apart from all that and back on topic. Why aren't active monitors designed to run of digital 0dbfs? Aren't they made for mostly computers?

And if not why not have a monitor that can run on analog 0dbVU and digital 0dbfs with a switch?

 

[snow lizard]

Everything is designed to run at line level, as described. There are 2 different ones for consumer or professional grade equipment. +4 dBu (pro) and -10 dBV (consumer). Some soundcards or interfaces will allow you to switch between the two to suit whatever you are using to monitor. Trying to run a +4 signal into a system designed for -10 would cause problems.

 

[garww]

....
"Why aren't active monitors designed to run of digital 0dbfs? Aren't they made for mostly computers?

And if not why not have a monitor that can run on analog 0dbVU and digital 0dbfs with a switch"?

Another consideration to ponder BEFORE going self-amped. I do have a few of those little 3-4-inch boxes, though, and find them handy when it's a chore( like distance) to connect something to an amp.

 

[Farview]

Why aren't active monitors designed to run of digital 0dbfs? Aren't they made for mostly computers?

You aren't running a digital signal to them. Since 0dbfs only means something while the signal is still digital, you are stuck dealing with whatever analog equivalent your interface decides.

And if not why not have a monitor that can run on analog 0dbVU and digital 0dbfs with a switch?

Because there is no standard between the two different scales. Hell, there are two different line levels, and some mixers (mackie) that decided to make their 0dbVU end up somewhere between the two standard analog line levels.

Add to that the fact that every interface and soundcard maker has a different idea of what the dbfs equivalent of line level (which ever one they use) is.

The whole thing is a shit show with no single standard for anything. So the chance of you overpowering the inputs on a very cheap set of power speakers is actually pretty high.

 

[Bobbsy]

As others have said, speakers aren't designed to operate at 0dB(FS) because they operate on a signal that has been converted to analogue and run through an analogue amplifier.

Most decent amps/speakers CAN handle a brief signal at 0dB(FS)/+18 dBu but, just as with the headroom when recording, they are not meant to run flat out all the time. To be able to do this safely would require a considerably more powerful (i.e. expensive) amplifier. But please let's not move this discussion to peak/RMS/continuous etc. power specs for amps or we'll be busy until Christmas.

 

[snow lizard]

I did some digging on Pinkertel's setup and found that the Juli@ card is modular and can run at +4 or -10 levels by swapping out modules that have either RCA (-10) or 1/4" TRS (+4) connections. Not sure if there is any other way to switch operating levels. I can't find any specs on the Hercules speakers, but they do have RCA and 1/4" balanced connections. One would think the balanced connections would handle a +4 signal but it didn't get mentioned in the manual. Suggested uses are for TV's, computers, home theatre.

Most decent amps/speakers CAN handle a brief signal at 0dB(FS)/+18 dBu but, just as with the headroom when recording, they are not meant to run flat out all the time. To be able to do this safely would require a considerably more powerful (i.e. expensive) amplifier. But please let's not move this discussion to peak/RMS/continuous etc. power specs for amps or we'll be busy until Christmas.

It shouldn't be that complicated. The Juli@ specs are actually fairly detailed regarding RMS and peak voltage for the respective modules. For example, the consumer module does not output upwards of 3 volts peak. The balanced one does. Since Pinkertel said he had to drop the signal around 15 dB so the speakers didn't kack, I'm thinking it's a consumer level thing.

The whole thing is a shit show with no single standard for anything. So the chance of you overpowering the inputs on a very cheap set of power speakers is actually pretty high.

That's the spirit!

 

[garww]

"Suggested uses are for TV's, computers, home theatre".

Can't say that suggests +4

 

[Bobbsy]

Doesn't sound like the best manual ever written does it?

Although it's not a hard and fast rule, the -10 level is used by domestic gear--like 'TVs, computers, home theatre" stuff that have unbalanced connectors on RCA/Phono or sometimes 1/8th inch jacks. The +4 set up is pretty well always pro level gear with balanced connection on a quarter inch TRS or XLR. (Yes, some pro stuff uses XLRs for line level--it's not restricted to microphones.)

 

[ecc83]

"
But apart from all that and back on topic. Why aren't active monitors designed to run of digital 0dbfs? Aren't they made for mostly computers?

And if not why not have a monitor that can run on analog 0dbVU and digital 0dbfs with a switch? "

Nice idea! If every source was capable of say +22dBu then monitors could be configured to deliver their maximum SPL AT that level. Trouble is of course, many, even most "prosumer" sources do not get close to a +22dBu capability. Vadrering a few specifications.....

2i2 max +10dBu
Steinberg UR22 +10dBu
NI KA6 +11dBu

These are all bus powered (or bus power capable) AIs of course and those that need external juice such as my 8i6 will be a bit hotter but you see the problem? A monitor with a sensitivity of say 100dB SPL for +22dBu could never be driven to decent levels by the vast majority of AIs and be an even bigger wimp driven from internal OBS cards (which they need to be able to do).

One downside to this necessary extra gain is that some active monitors generate audible hiss and this was quite a common complaint on forums a couple of years ago. The mnfctrs SEEM to have their act together now but about bloody time! Building a power amplifier that did not generate audible noise (at say 1mtr) has been beer into water for many decades! But even now you will rarely see a specification for 'self noise'.

A sensitivity switch? Fork! Cost money, there can be few markets as competitive as the HR active monitor one (maybe the middle-of-the-road guitar amp?) A $1 spent on a switch and associated components at design turns to around $10 at Sweetwater.

Dave.

 

[Pinkertel]

Trying to run a +4 signal into a system designed for -10 would cause problems.

Hmm as far as I know. I am going from TRS jack to TRS jack. so +4

You aren't running a digital signal to them. Since 0dbfs only means something while the signal is still digital, you are stuck dealing with whatever analog equivalent your interface decides.

Hahaha. I really have to be careful what I say XD. Yes you are right. I understand the signal is not digital. This is all really confusing. I set the computer on 0dbfs digital (the last mixer of the juli@) . I play normal music on the computer. Like Vevo on youtube or something. And this signal is to loud for the hercules monitors. The analog equivalent is just a Balanced TRS with +4dBu I guess? Sorry I have trouble with the semantics.

The whole thing is a shit show with no single standard for anything. So the chance of you overpowering the inputs on a very cheap set of power speakers is actually pretty high.

Aaah I get what you are saying. Yes no standards suck indeed. My underlying assumption was that there was a standard. My mistake trusting the world!

Most decent amps/speakers CAN handle a brief signal at 0dB(FS)/+18 dBu but, just as with the headroom when recording, they are not meant to run flat out all the time.

Hmmm well you just hear clipping in with the loud parts. But it all depends on the source ofcourse.

I did some digging on Pinkertel's setup and found that the Juli@ card is modular and can run at +4 or -10 levels by swapping out modules that have either RCA (-10) or 1/4" TRS (+4) connections. Not sure if there is any other way to switch operating levels. I can't find any specs on the Hercules speakers, but they do have RCA and 1/4" balanced connections. One would think the balanced connections would handle a +4 signal but it didn't get mentioned in the manual. Suggested uses are for TV's, computers, home theatre.

Thx for the digging!
YES! I would think so too. Tried balanced and unbalanced. Started clipping with both. (and no I didn't mix between the two)

Since Pinkertel said he had to drop the signal around 15 dB so the speakers didn't kack, I'm thinking it's a consumer level thing.

*she

"Suggested uses are for TV's, computers, home theatre".

Can't say that suggests +4

Strange I remember reading it was +4 somewhere. Can't seem to find it now.
But I must say this manuals is pretty bad. There are differences between languages also in specs.

But what the hell. Balanced TRS that isn't +4??? Does that even exist?

But even if that were the case. It has the exact same problem if I connect it unbalanced.

But apart from all that and back on topic. Why aren't active monitors designed to run of digital 0dbfs? Aren't they made for mostly computers?

And if not why not have a monitor that can run on analog 0dbVU and digital 0dbfs with a switch? "

Nice idea! If every source was capable of say +22dBu then monitors could be configured to deliver their maximum SPL AT that level. Trouble is of course, many, even most "prosumer" sources do not get close to a +22dBu capability. Vadrering a few specifications.....

2i2 max +10dBu
Steinberg UR22 +10dBu
NI KA6 +11dBu

These are all bus powered (or bus power capable) AIs of course and those that need external juice such as my 8i6 will be a bit hotter but you see the problem? A monitor with a sensitivity of say 100dB SPL for +22dBu could never be driven to decent levels by the vast majority of AIs and be an even bigger wimp driven from internal OBS cards (which they need to be able to do).

One downside to this necessary extra gain is that some active monitors generate audible hiss and this was quite a common complaint on forums a couple of years ago. The mnfctrs SEEM to have their act together now but about bloody time! Building a power amplifier that did not generate audible noise (at say 1mtr) has been beer into water for many decades! But even now you will rarely see a specification for 'self noise'.

A sensitivity switch? Fork! Cost money, there can be few markets as competitive as the HR active monitor one (maybe the middle-of-the-road guitar amp?) A $1 spent on a switch and associated components at design turns to around $10 at Sweetwater.

Dave.

No standard stuff.
Glad I dodged that hiss bullet a couple of years ago XD. Did have some hum problems though

 

[RFR]

Maybe those speakers are just shit?