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New Monitors, Questions Regarding DB
Ok,
So I just got some new tr8's. Brand new, they smell great 
I was just wondering when you guys use your monitors, does it make a difference how loud they are? These are my first pair of monitors, so I'm total noob when it comes to all the technical stuff involved. I set the soundcards input to MAX, and in the back the speakers are set to -5DB.
Hmmm... I have them at -20 dB, with sound card output set to -30 dB, coming through the volume control at 0 dB (at max.) But it should come down to 75 dB sound level while designing, mixing, etc. occasionally switching to 85 dB to get a crisp idea of what you're doing
Get a dB meter at RadioShack, it's about $50.
http://www.digital-recordings.com/audiocd/radio.html
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
Remember always run internal audio from your PC at maximum then turn the volume down on the monitors to maintain the integrity of the sound.
If you have an EXTERNAL mixer, you can adjust the volume from here, but better at the amp level.
| quote: |
| Originally posted by flutlicht junky Remember always run internal audio from your PC at maximum then turn the volume down on the monitors to maintain the integrity of the sound. |
Then it would appear that a small format mixer or other gain device will be an essential next purchase. 
Edited comments until can find source. * Lets hope I read it properly but more than likely I just mis-quoted lol *
FJ
| quote: |
| Originally posted by flutlicht junky Then it would appear that a small format mixer or other gain device will be an essential next purchase. FJ |
I know it adds a bit of colour but i dont have the bank for a proper audio interface atm
| quote: |
| Originally posted by flutlicht junky Then it would appear that a small format mixer or other gain device will be an essential next purchase. Bear in mind when volumes are done digitally inside a pc bits are removed to make the sound quieter resulting in degradation. Its not the same as reducing the voltage in a hardware mixer. FJ |
I'm trying to pull together more information. It maybe that it is incorrect.
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| The idea is that using the master volume control on the computer (in your soundcard mixer program) or winamp etc is not a great idea because you are attenuating the signal in the digital domain, and therefore reducing the bit depth of the signal before it is converted to analog. This means you are losing some available quality of the signal and in the worst cases you may hear graininess in low level signals, reverb tails etc. Even if it is not this obvious, the lost information may negatively effect the stereo image, soundstage etc in more subtle ways. |
| 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 |
i got one of these yokes, handier than adjusting my windows mixer volume all the time...
http://www.smproaudio.com/MPATCH2.htm
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| 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 |
| 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'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.
| 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 |
| 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. |
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.
| 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. |
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.
Relative, well that explains
Thanks!
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.
Thanks again .
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.
Whoa, didn't expect so many replies that quick. Interesting stuff guys, thanks so much for the input!
| quote: |
| Originally posted by camsr 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. |
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