move the gain knob of volume fader

[Nick The Man]

which one do you reach for when you first plug in a mic?


I dont know if this is right but i have all my gains set right in the middle (they are knobs) and then i move the fader.. is this incorrect?

 

[Massive Master]

You set the input level with the (pre-fader) pre-gain.

 

[groovyisland]

everythings usually set the way i like it already...

when i reach for a different mic what i want to adjust is the gain on the mic preamp if anything

thats usually called a trim pot on a mixer i think

trim is what youre doing

set your fader to unity gain or 0 or just under the red and twist the gain or trim knob to get it as hot as you want it - thats your starting point. then your faders will give you instant visual of whats going on

 

[Nick The Man]

so just set the faders to 0 and then just use the gain knob?

 

[pandamonk]

so just set the faders to 0 and then just use the gain knob?

Exactly. You wanna set it so you get the most volume without peaking.

 

[Nick The Man]

ok maybe im retarded but im gonna do a step by step on how i precieved these replies..

ok erhmmm

1: (plug the mic in) leave the gain knob right in the center
2: use the fader and raise the volume until the source is right under peaking
3: turn the gain down so that the source is not in danger of clipping


something doesnt seem right .. most people say they let the source peak at around -18dB when they record

 

[Farview]

Exactly. You wanna set it so you get the most volume without peaking.

No you don't.

1. You set the fader at unity gain (0db on the fader scale)
2. hit the solo button (PFL)
3. adjust the trim (pregain, gain, what ever it's called on your board) until the meters on the mixer bounce around 0dbvu


The trim brings the mic signal up to line level, the fader is for adjusting the relative levels of the channels.

 

[Farview]

something doesnt seem right .. most people say they let the source peak at around -18dB when they record

It isn't right. Your levels are supposed to average around -18dbfs, not peak.

The levels can peak where ever they do (as long as they don't clip), you just want the average power of the signal around -18dbfs. The easiest way to do this in your situation is to use the VU meters on your mixer and set the signal around 0dbvu on those meters. That will get you right where you need to be. (BTW, this doesn't work on drums or other percussive instruments, with those you just set them to peak around -6dbfs (in the computer) and forget it)

 

[Massive Master]

Exactly. You wanna set it so you get the most volume without peaking.

I know Farview already nailed that, but I just wanted to add a "roll eyes" for the helluvit.

Here goes:

 

[Nick The Man]

thanks guys.. now the signal runs through different busses on the board and then those send to an interface which also has a trim knob.. hmmmm

 

[Farview]

You are going to have to calibrate your setup.

You will need a test tone (like a 1k sine wave) and the owners manual for your interface.

Look in the manual to find out:
1. Are the line inputs on the interface +4 or -10
2. what the converters are calibrated to (line level input = ?dbfs)


It is a lot easier if everything has a setting for unity gain. (the busses on the mixer might)

What you do is follow my instructions from before and set the input trim so the meter on the mixer reads 0dbvu.

Using the pfl on each buss of the mixer, make sure that the level is still 0dbvu all the way through the mixer.

Plug the output of the mixer into the line input of the interface. If there is a unity setting on the interface, set it there and you are done. If there isn't, you will need to know what level in the computer equals a line level signal at the interface. Once you know that, you can set the trim on the interface so your test tone reads that level in the computer.

Now that you have done that, you can leave the trim control on the interface alone forever. You will always adjust the level with the mixer.

 

[pandamonk]

No you don't.

1. You set the fader at unity gain (0db on the fader scale)
2. hit the solo button (PFL)
3. adjust the trim (pregain, gain, what ever it's called on your board) until the meters on the mixer bounce around 0dbvu


The trim brings the mic signal up to line level, the fader is for adjusting the relative levels of the channels.

Isn't that what I said? It's what I meant...

 

[Nick The Man]

You are going to have to calibrate your setup.

You will need a test tone (like a 1k sine wave) and the owners manual for your interface.

Look in the manual to find out:
1. Are the line inputs on the interface +4 or -10
2. what the converters are calibrated to (line level input = ?dbfs)


It is a lot easier if everything has a setting for unity gain. (the busses on the mixer might)

What you do is follow my instructions from before and set the input trim so the meter on the mixer reads 0dbvu.

Using the pfl on each buss of the mixer, make sure that the level is still 0dbvu all the way through the mixer.

Plug the output of the mixer into the line input of the interface. If there is a unity setting on the interface, set it there and you are done. If there isn't, you will need to know what level in the computer equals a line level signal at the interface. Once you know that, you can set the trim on the interface so your test tone reads that level in the computer.

Now that you have done that, you can leave the trim control on the interface alone forever. You will always adjust the level with the mixer.

when i look at the manual all i can find is this:

By the way its a Tascam FW-1804 just so you know.

thanks a ton for all the help already far!

LINE input 1-4 (Balanced)
Connector 1/4 inch TRS (combo jack) x 4 (T: Hot, R: Cold, S: GND)
Input impedance 10 k ohm
Adjustable input range –43 dBu (TRIM max) to +11 dBu (TRIM min.)
Maximum input level +27 dBu


LINE input 5-8 (Balanced)
Connector 1/4 inch TRS jack x 4 (T: Hot, R: Cold, S: GND)
Input impedance 15 k ohm
Adjustable input range –42 dBu (TRIM max) to +4 dBu (TRIM min.)
Maximum input level +20 dBu

 

[Nick The Man]

Plug the output of the mixer into the line input of the interface. If there is a unity setting on the interface, set it there and you are done. If there isn't, you will need to know what level in the computer equals a line level signal at the interface. Once you know that, you can set the trim on the interface so your test tone reads that level in the computer.

Now that you have done that, you can leave the trim control on the interface alone forever. You will always adjust the level with the mixer.

Ok so make sure the test tone reads 0dB on the mixer (thats how far i am) then send the signal to the interface (which goes to the computer) and then turn the trim on the interface until it reads 0dB on the computer also?

 

[Nick The Man]

ok this is what ive done. let me know if its right.

Made a CD with a 1k sine wave and a 500 sine wave.

hooked the cd player into the mixer. THEN:

1. Set fader to 0dB and hit the PFL solo and set the gain so that the wave hit 0dB right on

2. selected a bus (1) then hit PFL solo on the bus and the signal dropped down to about -7dB so i raised the gain again until it was back at 0dB (bus faders also slid up to 0dB)

3. Opened my recording software and armed a track for recording on input 1

4. Adjusted the trim on the interface until the wave was at -12dB on the computer

I did this for each bus therefore i have set levels on the everything including the trim levels on the interface.

still feel like i may be doing somehting wrong .. dont know why when i select a bus and check the signal soloed on the bus and its less than just the signal by itself??

 

[Farview]

Isn't that what I said? It's what I meant...

nope, you said

You wanna set it so you get the most volume without peaking.

 

[Farview]

Ok so make sure the test tone reads 0dB on the mixer (thats how far i am) then send the signal to the interface (which goes to the computer) and then turn the trim on the interface until it reads 0dB on the computer also?

Nope. 0db(VU) on the mixer and 0db(FS) in the computer are not the same.

Since the specs don't help much, calibrate the input trim on the interface so your 0dbVU (the mixers meter) sine wave sits at -18dbfs (the computers meter). Do that for all the inputs and tape the trims on the interface so they don't move. The system is now calibrated.


The part that some people have a hard time wrapping their head around is that the mixer is using a different db scale than the computer is.

0dbfs (computer) is the absolute ceiling, you can't go beyond it.
0dbVU (Analog gear-preamps, mixers, etc...) is line level. You normally have at least 18db of headroom before it clips.

There has to be 1000 threads explaining this (I've certainly typed it enough) in greater detail, but that is the general idea.

 

[Farview]

ok this is what ive done. let me know if its right.

Made a CD with a 1k sine wave and a 500 sine wave.

hooked the cd player into the mixer. THEN:

1. Set fader to 0dB and hit the PFL solo and set the gain so that the wave hit 0dB right on

2. selected a bus (1) then hit PFL solo on the bus and the signal dropped down to about -7dB so i raised the gain again until it was back at 0dB (bus faders also slid up to 0dB)

3. Opened my recording software and armed a track for recording on input 1

4. Adjusted the trim on the interface until the wave was at -12dB on the computer

I did this for each bus therefore i have set levels on the everything including the trim levels on the interface.

still feel like i may be doing somehting wrong .. dont know why when i select a bus and check the signal soloed on the bus and its less than just the signal by itself??

You did everything right (I suggested -18dbfs instead of -12dbfs, but I just like the extra headroom)

The reason the busses were quieter was because the gain control on the buss wasn't set at unity, now it is.

 

[Massive Master]

0dbVU (Analog gear-preamps, mixers, etc...) is line level. You normally have at least 18db of headroom before it clips.

And although Farview understands and has typed this 100 times also, I just thought I'd add for the sake of explanation -

A signal in the digital realm is more or less "there." You can bring the volume up, you can bring the volume down. You can take that -18dBRMS sine wave and push it to -0.1dBFS and it isn't going to sound any different (although your converters will hate you to the very depths of your soul for doing it). It's going to be perfect, perfect, perfect, perfect, CLIP.

Analog isn't like that. There is great analog gear that can handle and awful lot before it clips, but generally, you're looking at (at or below line level and going up) Perfect, a little distorted, unfocused, loss of clarity, spectral distortion, harmonic distortion, CLIP.

The CLIP is the point of FAILURE - Not the point where the signal loses its integrity and character. Getting a signal "hot but without clipping" is normally overdriving the input chain. With a lot of "budget friendly" gear, that really has no decent measure of ("real") headroom, that quickly adds up to "Why don't my mixes sound like 'pro' mixes?"

 

[Nick The Man]

wow guys i couldnt thank you enough .. finally i have a calibrated system and some sound definitions.

ill send you guys some fruit baskets or something