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flutlicht junky
in das haus



Registered: Oct 2001
Location: Bournemouth, UK

quote:
Originally posted by evo8
i got one of these yokes, handier than adjusting my windows mixer volume all the time...
http://www.smproaudio.com/MPATCH2.htm


Weird in the spec of the hardware you brought it discusses the situation I mentioned above, I'll quote it:

quote:
No more bit loss! One of the benefits of passive volume attenuation is that you no longer have to control volumes with your software master volume fader. Reducing audio levels from software only reduces your bit depth. It is much more appropriate to keep your software masters at unity and passively attenuate the audio to your active monitors.


I would actually like to know more about this. Does anyone here know anything about unity gain and the affect of dropping levels with the digital domain and making up the gain later on the signal chain?

Old Post Feb-26-2007 21:32  United Kingdom
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DigiNut
You kids get off my lawn!



Registered: Dec 2002
Location: Toronto, Self-proclaimed Centre of the Universe

I've heard this bizarre assertion about "bit loss" several times and still can't understand where it could possibly have come from. It's completely untrue, and my only guess is that it must have been something spewed by a marketing weasel that got misinterpreted by a few viewers or buyers.

Bit depth in the digital domain is a constant 16-bit, 24-bit, or 32-bit floating-point. Period. There are no bits "removed" when you lower the volume, nor is there ever a scenario where you "don't get the full 16/32 bits" (this is not like a motor assembly where you only get 90% efficiency, you get 100% of the swing in a digital signal).

Only exception is when you're using a bitcrusher, but that's obviously a deliberate bit reduction (duh).

In actual fact, it's much better to use software to reduce gain because there's no noise. In the analog domain, if you have a noisy stage that outputs low gain, and you amplify that at the next stage, you're amplifying the noise with it. This is a non-issue when working with digital audio.


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Old Post Feb-27-2007 01:51  Canada
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Derivative
Bipolar Bear



Registered: Jun 2004
Location: Dublin

quote:
Originally posted by richg101
if your soundcard is pushing out 0db and your monitors are set to 0db (with matching impedance) then the sound will be the loudest before the amp starts to be over driven.

you should have the power amp in the monitors set to 0db and then adjust the volume using your mixer. you then know that when you get to 0db on your mixer, the speakers will be near/at their limit.

running monitors too loud will adjust their characteristics and make the frequency response jagged. you will know when the speakers are at their limit so just be sure to stay below that limit by a bit. also prolonged listening at loud volumes will make it almost impossible to get a good mix. you should mix at a average level and then turn up now and then to see what its like at high volumes.

hope my comments help a bit


This I guess is what you are supposed to do but you have to take a disciplined approach to turning on your gear and you have to be careful with gain faders - the more powerful the amps are, the bigger risk there is. This is mainly for the benefit of people who don't realise this - I'm sure you know your shit in this regard.

Driving my Dynaudio BM5as to capacity is absolutely deafening. I could not tolerate this listening level for more than a few seconds with the speakers about 4 feet away from me. And FL Studio defaults to 100% gain every time you start a new project so you have to lower it dramatically.

Also, if you do this you must always turn on your gear in a very specific order but the most important one is that you turn on your amps/monitors last. Certain processors, synths and audio interfaces emit loud popping sounds when you turn them on and the output is live. My Access Virus B for instance emits a massive low level popping sound when you switch it on with the amps live. I know that the Focusrite Saffire emits a loud noise when you hotplug it.

Also, you must never hotplug signal carrying lines as you can get bursts of noise/static when doing so. At 110dB this is not cool.

Also, every processor/synth/module that has a gain stage - you must turn down the gain for introducing it into the signal chain and before turning on your amps. So whenever you switch on your monitors you have to slowly increase gain stages one by one to avoid damaging your speakers and your ears. Hope this helps.

Old Post Feb-27-2007 10:57  Ireland
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richg101
1010101010101010101010101



Registered: Apr 2005
Location: a universal nation

quote:
Originally posted by Derivative
This I guess is what you are supposed to do but you have to take a disciplined approach to turning on your gear and you have to be careful with gain faders - the more powerful the amps are, the bigger risk there is. This is mainly for the benefit of people who don't realise this - I'm sure you know your shit in this regard.


what i will say is that you generally it is worth running your power amp gain (on the back of the monitors) to 0db. this poses more of a risk due to the higher gain of anything that pops. but remember that aplying a methodical turn on procedure you will never hear a turn on pop again. just remember to turn on the intitial sound sources first (external synths) follow on with your interface/mixer that all your hardware goes into. then turn on your pc, then turn on your volume controller/mixer, and then your monitors/power amps on. just follow the direcetion tha sound signal goes. having your power amplifiers set to 0db level is good becuase you can then monitor the input level on your mixer/vol controller. you know that you are not clipping your amps until you go over the 0db level on the mixer output meters.

i guess for using a setup where your soundcard mixer is used as a volume control then maybe a lower level on the monitor gain is a good idea. but really the same situation applys as with the external mixer/vol control setup.


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Old Post Feb-27-2007 13:48  England
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Derivative
Bipolar Bear



Registered: Jun 2004
Location: Dublin

Thats really the point. If you do this method of amping up your monitors you have to be disciplined about it or you can destroy your monitors and your ears. Also, you better make sure you don't clip in your DAW as if you clip in your DAW you are clipping your power amp and putting undue stress on the drivers. Stress that they were not designed to cope with.

So what I do is give myself a little leeway and set the output to -6dB in my software mixer. That means I can clip in soundforge and have about 6dB over before I clip the power amp. I do this because I don't want to accidentally damage my speakers. I rarely monitor at full capacity anyway as its so freaking loud I couldn't tolerate it for more than a few minutes.

Old Post Feb-27-2007 14:00  Ireland
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Storyteller
Supreme tracneaddict



Registered: Feb 2005
Location: The Netherlands

quote:
Originally posted by DigiNut
I've heard this bizarre assertion about "bit loss" several times and still can't understand where it could possibly have come from. It's completely untrue, and my only guess is that it must have been something spewed by a marketing weasel that got misinterpreted by a few viewers or buyers.

Bit depth in the digital domain is a constant 16-bit, 24-bit, or 32-bit floating-point. Period. There are no bits "removed" when you lower the volume, nor is there ever a scenario where you "don't get the full 16/32 bits" (this is not like a motor assembly where you only get 90% efficiency, you get 100% of the swing in a digital signal).

Only exception is when you're using a bitcrusher, but that's obviously a deliberate bit reduction (duh).

In actual fact, it's much better to use software to reduce gain because there's no noise. In the analog domain, if you have a noisy stage that outputs low gain, and you amplify that at the next stage, you're amplifying the noise with it. This is a non-issue when working with digital audio.


Are you saying the digital domain is relative? If I lower the volume would that mean the same bits and bytes have less of a difference in dynamic capabilities, but just as much steps? ie. 16 bit audio will have all 16 bits up to -inf up to 0dB. Would it be possible to lower the volume and get 16bits to max at -inf to -3dB? Decreasing the dynamic space in between the bits?

I found in quite logical to see why lowering the volume digitally would reduce the quality. Why? Because *I think* (not sure though) the digital domain is absolute. This means every difference between the bits is fixed. Which means the max would always be 0dB adn nothing less. This means rounding would occur when lowering the volume digitally and would maybe cause quite similar effects like bit reduction?

Funny thing is, I've never really been able to spot any (disturbing) difference when lowering the volume digitally. I'm very curious how it works... anyone?


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quote:
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Old Post Feb-27-2007 14:43  Netherlands
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Derivative
Bipolar Bear



Registered: Jun 2004
Location: Dublin

No. bitdepth doesn't work like that.

If you imagine a sine wave and draw a grid over it, the vertical *resolution* (amplitude) is described with bits. The horizontal *resolution* is incrementally described by sampling bitdepth so many times per second (44,100 times per second for a 44.1khz recording).

Think of it in terms of pictures. Take a 24 bit bitmap file and save it as an 8 bit bitmap then you no longer have the number of bits to describe 16.7 million colours. You have more like 256 colours. Or something. So the picture will lose alot of detail because you can't describe smooth transitions between thousands of colours anymore.

Bitdepth in audio describes signal to noise ratio so it describes the peak signal in relation to silence. The higher the bitdepth the greater the SNR (sometimes called misleadingly dynamic range).

If you render a 24 bit project down to 8 bits you are throwing away 16 bits and you no longer have the number of bits to describe an SNR of over 100dB. You have so few bits that you will find that in relation to silence, sounds will not fade out smoothly to absolute silence, they will appear to sound gated and will suddenly stop instead of fading out. You can no longer represent instruments acurately that have a very large dynamic range and the quieter segments in relation to the peak signal will just disappear or sound distorted (like a gate where the threshold is set too high).

Bit depth does not describe absolute volume because all of this is relative. If you monitor at 0dB or at -30dB, your signal to noise ratio will be the same provided you stay at the same bitdepth. Your ears won't be sensitive enough to tell this dynamic range if you monitor really really quietly but the important thing is that it is always proportional.

Think about it for a second. If by lowering volume digitally, you were truncating bits, then you would end up with bitcrushed noise if you monitored at really low level in your software mixer. This is obviously not the case

Also don't forget that your monitors/amps will be connected to the analogue outputs on your soundcard. So your soundcard will convert a digital signal to analogue anyway before it outputs to the pre amp stage in your monitors. Which then takes this very small signal and scales it up using a mains power supply.

Old Post Feb-27-2007 16:54  Ireland
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Storyteller
Supreme tracneaddict



Registered: Feb 2005
Location: The Netherlands

Relative, well that explains

Thanks!


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quote:
If less is more think about how much more more would be.
-Frasier

Old Post Feb-28-2007 00:29  Netherlands
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DigiNut
You kids get off my lawn!



Registered: Dec 2002
Location: Toronto, Self-proclaimed Centre of the Universe

Derivative is essentially correct, but the fact that it's a "relative" scale is a totally moot point and really doesn't have much to do with what he said.

Storyteller, we had a thread about this before where I explained that dB is a relative scale no matter what equipment you're using. If there's no suffix after the "dB" (like "dBW" or "dBm"), then the reference point is undefined and the signal is usually in reference to whatever the maximum output of the system is.

dB is relative in the analog domain as well. 0 dB at the output of a power amp is a drastically different signal level from 0 dB at the input of a mic preamp. In this respect, there's no difference between digital and analog. Even your monitoring setup makes a difference. If I use passive speakers, I'm putting out something like -20 dB (that's a guess, don't quote me on it) from my power amp to the speakers, but that's a drastically higher power than the 0 dB "line-level" signal which usually gets sent to powered studio monitors. The only "absolute" scale here is the dBA that describes the sound waves coming out of the speaker.

The only significant loss of quality in the digital domain is quantization error, and that only occurs on the way in, or sometimes but rarely on the way out if you have awful DACs. It's quantization error that contributes to the lower SNRs that Derivative is talking about at lower bit depths.

On the way in, let's say that the signal you're recording has a swing of -12 to +12 V. Now let's say you're recording at only 8 bits, that gives you 2^8 or 256 discrete signal levels in the digital domain, which means that the analog signal has to go up by 0.05 V (50 mV or close to 0.5% of the total swing) to register a change in the digital domain. That's actually a fair bit, and if you quantize a sinewave and then convert it back to analog, you get kind of a stepwave on the output which sounds noisy (hissy, really).

16 bits gives you 2^16 or 65536 discrete levels, which corresponds to less than 1 mV of swing, or .002% of the signal. It's so negligible that it's actually lower than the ambient noise floor of most studios, hence the term "CD quality". You're more likely to get noise on your cables than you are through quantization error in 16-bits, and you'll never ever be able to hear quantization error in 24 or 32 bits.

So essentially, the only thing to take home from this is that in modern digital equipment, any loss of quality compared to the analog domain is totally inaudible, even if it's passed back and forth to the digital/analog domains several times in succession.

Analog equipment may colour the sound, produce 2nd-order distortion, have a warmer quality to it, etc. - those are all valid arguments in favour of analog equipment in some circumstances, but there is no significant loss of quality when working with digital equipment at a depth of 16 bits or more.


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Old Post Feb-28-2007 01:27  Canada
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Storyteller
Supreme tracneaddict



Registered: Feb 2005
Location: The Netherlands

Thanks again .


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quote:
If less is more think about how much more more would be.
-Frasier

Old Post Feb-28-2007 03:05  Netherlands
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camsr
Suspended User



Registered: Sep 2006
Location: nor cal

I use my Grado cans straight from the soundcard, so I HAVE to turn down the software mixer level to listen at a reasonable level. But since the slider for the master level is linear, I have it usually set at one-quarter of its full range. 1/4 of a 0dB signal is -12db. This means I only lose two bits of resolution at 16 bits (onboard sound). 14 bits is 2^14 = 16384 discreet volumes for monitoring, OR 84 dB of dynamic range. It's 84 db of range because each bit corresponds to 6dB of headroom. The point quantization noise really becomes super noticeable is at about 12 bits. So you might not be losing as much headroom as you think using the software mixer.

The only time when bitdepth is important is when you are processing with plugins. Phase shifts and the like will sound bad at low bitdepths. High bitdepths in the processing stage allows you more headroom for the effects to work without raising quantization noise.

Old Post Feb-28-2007 05:16 
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Jmanch
tranceaddict



Registered: Sep 2006
Location:

Whoa, didn't expect so many replies that quick. Interesting stuff guys, thanks so much for the input!


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Old Post Feb-28-2007 05:53  United States
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