Blue Bear Sound
New member
Pygmy Players?? Yeah... I heard of them!
(just pokin' a bit 'o fun at ya Sjoko!)
Bruce
(just pokin' a bit 'o fun at ya Sjoko!)
Bruce
E
enemyofthesun
New member
Blue Bear Sound said:29"????? Well... I guess at 29 inches, standing waves don't have a chance to build up a whole lot... but how do the musicians play lying down on their sides?!?!?!
ahhh hahaha!! Good good good.!!
sjoko2
New member
I'm doing something about the California energy crisis, small studio, small people, small instruments, little bit of electricity
Tekker
New member
29" huh?? I would have never guessed that from looking at your pictures because it sure looks big to me, but it's probably a lot smaller in person though.
-tkr
-tkr
sjoko2
New member
Sigh.......well everything is relative 
crosstudio
New member
about this digital 0 thing... i just don't believe that it's necessary to record as close to 0 as possible. this isn't analog where the tape player itself it going to contribute lots of noise. this is digital where the A/D/A is relatively quiet.
if, in order to get to 0 db, you have to push your pre-amp to the point where it's adding more noise than signal strength what's the point?
if, in order to get to 0 db, you have to push your pre-amp to the point where it's adding more noise than signal strength what's the point?
sjoko2
New member
If you wonder what's the point..... read the posts?
Sense and reason..... Apart from which, a pre should not introduce noise when a signal is boosted to odB? Which is, after all, the point where equipment has been / should have been designed to provide optimum performance / max dynamic range????
Sense and reason..... Apart from which, a pre should not introduce noise when a signal is boosted to odB? Which is, after all, the point where equipment has been / should have been designed to provide optimum performance / max dynamic range????
dobro
Well-known member
My pres don't add noise when I boost the signal to 0 dB, but they add some noise when I make the signal as hot as possible. Here's what I mean:
I'm using two pres a lot these days - a Mindprint Envoice and a Meek VC6Q. I use the input gain knob and the meter on the pre to get the signal as hot as possible, before I use my onscreen software meter to adjust the output gain knob to get the level up to the 0 dB vicinity. And yet, following this procedure, I find that making the signal as hot as possible adds some noise to the signal - it's visible on the onscreen waveform - it thickens the midline of the waveform, even before the music starts. If I cut the input gain one notch or so, that noise, that thickening of the wavefrom midline, disappears.
My point is, the Mindprint is a really quiet peace of gear, but depending how hot the signal is, it still seems to add some noise to the signal if I push it to its internal limit.
I'm using two pres a lot these days - a Mindprint Envoice and a Meek VC6Q. I use the input gain knob and the meter on the pre to get the signal as hot as possible, before I use my onscreen software meter to adjust the output gain knob to get the level up to the 0 dB vicinity. And yet, following this procedure, I find that making the signal as hot as possible adds some noise to the signal - it's visible on the onscreen waveform - it thickens the midline of the waveform, even before the music starts. If I cut the input gain one notch or so, that noise, that thickening of the wavefrom midline, disappears.
My point is, the Mindprint is a really quiet peace of gear, but depending how hot the signal is, it still seems to add some noise to the signal if I push it to its internal limit.
sjoko2
New member
Do you use a master fader in your DAW? Do you have a calibrate function?
If you use your pre the way you do, what level is your fader on average in your DAW before the signal reaches odB?
If you use your pre the way you do, what level is your fader on average in your DAW before the signal reaches odB?
dobro
Well-known member
I don't use a mixer to track. I go straight from the pres into the soundcard, so there's no master fader.
sjoko2
New member
in your DAW?????
dobro
Well-known member
I once asked what DAW meant here - I got a lot of answers. Maybe we're not talking the same language.
I track direct to my computer's soundcard via two channel strip preamps. I use the meters on the preamps to set the input gain there, then I use Cool Edit's onscreen meter to set the output gain from the pres to the computer. Of course, I *listen* to how it sounds, but I find it interesting what the onscreen waveforms do (see above post about high input gain on the pre being registered by onscreen meter).
I track direct to my computer's soundcard via two channel strip preamps. I use the meters on the preamps to set the input gain there, then I use Cool Edit's onscreen meter to set the output gain from the pres to the computer. Of course, I *listen* to how it sounds, but I find it interesting what the onscreen waveforms do (see above post about high input gain on the pre being registered by onscreen meter).
sjoko2
New member
DAW = Digital Audio Workstation
I have never seen of used cool edit, but if you incur noise the way you do, that normally means there is a calibration issue in you input chain.
There are 2 ways to solve this.
1. See if your software has a calibration window and correct it from there.
2. If you have the ability to "create channel" within your software program, see if you can create a "master fader" which allows you overall control.
If not, perhaps you should go and download Pro Tools Free, and try that. Free, and very likely a far superior software program.
I have never seen of used cool edit, but if you incur noise the way you do, that normally means there is a calibration issue in you input chain.
There are 2 ways to solve this.
1. See if your software has a calibration window and correct it from there.
2. If you have the ability to "create channel" within your software program, see if you can create a "master fader" which allows you overall control.
If not, perhaps you should go and download Pro Tools Free, and try that. Free, and very likely a far superior software program.
dobro
Well-known member
Cheers, Sjoko. I'll check it out. Cool Edit has a master gain setting, but it's only for playback - it controls nothing during recording - it just takes what's given it.
sjoko2
New member
I know it doesn't - but it will allow you to see the comparative levels throughout the system.
For example - something every pro studio does (should do might be a better expression), is run tones through the entire system.
Put a tone into your system at 0dB, with your master fader at 0dB, and all tracks armed and at 0dB - now your signal should be bang on 0dB on every channel. Once that is done you go through your I/O and equipment, doing the same thing. That way you know what are the true levels of your systems paths.
For example - something every pro studio does (should do might be a better expression), is run tones through the entire system.
Put a tone into your system at 0dB, with your master fader at 0dB, and all tracks armed and at 0dB - now your signal should be bang on 0dB on every channel. Once that is done you go through your I/O and equipment, doing the same thing. That way you know what are the true levels of your systems paths.
J
JimH
New member
I enjoyed the Dreampoint (Jezar) site mentioned above and I think it's a valuable resource. But I also don't quite buy the whole part about recording at lower levels. In his example about recording the entire band that wants to punch in louder at the end, he makes sense there. But I don't think it should be a universal rule. If you are recording at home one track at a time, I don't see the problem with recording at high levels. He says "What's the point in religously trying to record your cabasa at exactly 16 bits when it is never going to represent more than 12 bits of the final stero mix?" I think there are two reasons why. First, in computer-based multi-track recording the recorded audio gets converted to 32-bit floating point. So even if you end up lowering the volume of that 16-bit recorded track to the equivalent of 12-bit volume, you still retain the 16-bit accuracy. Second, what if you record at a low volume so that the full volume of the casaba is represented by 12 bits but then you later decide to mix it louder? No matter how much you push up the volume, the sound quality will never be better than 12-bit.
So I think he should qualify in which instances it is bad to record at full volume and which instances it is OK.
So I think he should qualify in which instances it is bad to record at full volume and which instances it is OK.
sjoko2
New member
Of cause there is something as logical as --> you record at 16 bit, or 24 bit, your track is 16 bit or 24 bit - if you lower the volume it does not mean you lower bitrate?!
The point I was trying to make - The examples used for this type of argument are normally those for instruments with a penetrating sound, which, by nature, are instruments which operate within a narrow frequency spectrum.
Lets go back to the most used example, the High Hat. If you select your microphone, place it properly, record it at a good level, say -3dB to allow for drummer unevenness.
Now you have a full track at just under odB. Come mixing. You start to isolate those frequencies you wish to keep, shelve those you don't want (or perhaps you can use them to put something else back in the sound? On a seperate track?), now your signal strength will have dropped considerably, and is likely to be close to where you want it to be in the overall kit level.
Had you NOT done that, it is very likely that, after you processed the track, you would have to boost it, or boost certain frequencies of that track, thereby decreasing overall track quality.
Common sense, logic, call it what you like. It's called the art of recording, it has been proven to work
The point I was trying to make - The examples used for this type of argument are normally those for instruments with a penetrating sound, which, by nature, are instruments which operate within a narrow frequency spectrum.
Lets go back to the most used example, the High Hat. If you select your microphone, place it properly, record it at a good level, say -3dB to allow for drummer unevenness.
Now you have a full track at just under odB. Come mixing. You start to isolate those frequencies you wish to keep, shelve those you don't want (or perhaps you can use them to put something else back in the sound? On a seperate track?), now your signal strength will have dropped considerably, and is likely to be close to where you want it to be in the overall kit level.
Had you NOT done that, it is very likely that, after you processed the track, you would have to boost it, or boost certain frequencies of that track, thereby decreasing overall track quality.
Common sense, logic, call it what you like. It's called the art of recording, it has been proven to work
J
JimH
New member
Here's what he is referring to. 16-bit resolution means that the audio signal can be represented digitally by numbers ranging from -32768 to +32767. (This is a range of 64536, which is 2^16.) So if you record a signal at 0dB, the loudest sounds will have numbers as high as +32767 (and as low as -32768). If you then lower the volume on playback, say to 1/16th original volume, you are dividing these numbers by 16. That signal will then be represented by numbers from -2048 to +2047. The highest 4 bits of the available 16 will not be used at all and it will be the same as if you had recorded the signal on a 12-bit recorder at 0dB. Or another way to look at it: it would be the same as if you recording the audio at 1/16th volume in the first place.
That whole reasoning above would be correct if you use integers. But multi-track recording programs don't, they use floating point numbers internally. That's why the argument doesn't hold up.
sjoko2
New member
exactly - its difficult to try and be non-confusing
crosstudio
New member
EDITED NOTE on IEEE-754 Floating Point Precision
bit 31 : sign bit
bits 30-23: exponent field
bits 22-0 : significand
that means that any value with 21 significant digits or less can be represented with no loss of precision. that's great. i'm impressed. d*mn sure impressed.
anyway like i was saying, if you are pushing your gear past it's best sound just to try to get to 0db, then you are fooling yourself. why not record it at its best level and make it louder digitally if necessary? most of these sequencing software packages give you the ability to normalize to 0db anyway. yes, the normalization has artifacts. but you have to be intelligent enough to weigh the artifact vs. you pushing your gear past its limits. the software packages will get better and leave less artifacts as they upgrade. recording bad sound is permanent.
if you are working in a gold room where every pre-amp, mic, and guitar is top notch then you've got nothing to worry about. but if you are like me and you are using a tech21 sans amp going into a mackie 1604 vlz you've got to use your head because neither piece of gear is going to get me to 0db when i'm playing without making the noise on the track rediculous.
bit 31 : sign bit
bits 30-23: exponent field
bits 22-0 : significand
that means that any value with 21 significant digits or less can be represented with no loss of precision. that's great. i'm impressed. d*mn sure impressed.
anyway like i was saying, if you are pushing your gear past it's best sound just to try to get to 0db, then you are fooling yourself. why not record it at its best level and make it louder digitally if necessary? most of these sequencing software packages give you the ability to normalize to 0db anyway. yes, the normalization has artifacts. but you have to be intelligent enough to weigh the artifact vs. you pushing your gear past its limits. the software packages will get better and leave less artifacts as they upgrade. recording bad sound is permanent.
if you are working in a gold room where every pre-amp, mic, and guitar is top notch then you've got nothing to worry about. but if you are like me and you are using a tech21 sans amp going into a mackie 1604 vlz you've got to use your head because neither piece of gear is going to get me to 0db when i'm playing without making the noise on the track rediculous.
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