ORIGINAL: jazzbass12

Waltenstein:
What I ment was the fader control. If I read your post correctly, you had said to adjust you trim (Gain) first then you faders.
So how do you adjust the trim if you have the faders down? Do you adjust all the trims so they so 0?
I might sound ingnorant but, I am just trying to understand the concept.
I have never used the trim controls before.
Thanks

ORIGINAL: Saxon1066

Here's something I always wanted to know: on commercial rock recordings, I've noticed (I think) that when the vocals come in, the levels of all instruments drop back into the background. Ten, during instrumental passages, like between verses, the instruments come back to the forefront.

How is this achieved? Do the mixers use automation (or in the old days, a steady hand) to drop the instrument levels and raise vocal tracks at the right times? Can you get the same effect by mixing good levels with the vocals in, and using compression to keep the overall song level when vocals are out?

I'll have to deal with this eventually.

ORIGINAL: kicksville
Which brings me to my question: does anybody know the internal structure of the buses in Sonar (bit-rate)? According to traditional analog summing laws, I should be adjusting a given bus's input Trim to compensate for the incoming signals. Some consoles (like the Soundcraft K2 and MH4) reduce the gain of their summing amplifiers by 6db to deal with this, so the "input trim" is under the hood, so-to-speak. Protools handles summing by having a 48-bit bus architecture. So, does Sonar pass signals from the channels to their destination at the project's bit rate or at 32-bit floating point? Or what?

ORIGINAL: RRabbi

I just discovered something about my new MOTU 828 MKII... I haven't used it yet, but am getting familiar with the manual. One part says I can toggle between +4 db and -10 db. So In order to leave some headroom in Sonar, I'd assume then I should always record with the setting toggled at -10db?

ORIGINAL: kicksville
Keep in mind that dbFS is the scale we usually mean around here when we say "you need to keep your levels at -10db" - dbFS refers to digital 0 (decibels Full-Scale) and not to a measurement of voltage (dbu and dbV)....

Conrad
www.kicksville.com

ORIGINAL: kicksville

This actually touches on something I've been meaning to address for a while now - gain structure. I'll try to explain some basics of signal flow and gain structure for you, but this also brings up some questions of my own about how Sonar handles certain things. Just for the record, I'm ignoring the 16/24 debate for now because proper gain structure is relative to a given system's available headroom.

In general, signal flow in Sonar emulates flow through a traditional console. Each channel/bus has an input and and output gain adjustment. On most consoles, the input is the first knob on the channel strip, and is usually labeled "Input Trim". Sonar adopted this convention, so that's the Trim slider. Again, on a traditional console, the output level after passing through the channel strip is set using the fader at the bottom (we're ignoring more complicated mix architecture like matrix mixing, etc.). Same thing in Sonar - the Volume control/fader controls the signal's output level, after channel processing.

The standard convention for gain structure is to use your input to set optimum peak levels, and your output to set mix levels. Translated to Sonar, in practice this means that when recording, set your input levels using your preamps (or whatever you're using to get signal into the CPU) and set your monitoring levels using either the faders in Sonar (if you're using input monitoring) or in your hardware's output section. When mixing, the Trim control and clip gain envelopes adjust the signal level before any other processing. The Volume fader adjusts the signal level after it has been processed (EQ, inserted FX).

Here's where it can get complicated and absolute numbers matter. If you have a broad-spectrum input signal that peaks at -3db, and you use the channel EQ (for example) to boost 500Hz by 6db, you will obviously clip the output. There are ways to compensate for this (turning down the input or cutting frequencies instead of boosting for example), but you can see why it's important to leave headroom on a given track. This is also why mastering engineers like to see peaks between -10db and -6db on raw mixes.

Now, when you're sending a bunch of signals from different outputs (channels) to a single input (a bus), you have to pay attention to gain structure here as well - and each subsequent stage, for that matter. This is called summing. Here is a quote from the Midas XL250 manual (one of the industry-standard big-ass monitor consoles):
"When two coherent test signals are summed together the resultant signal is 6db higher than the inputs...this pattern continues thus: 4 inputs give +12, 8 inputs give +18, 16 inputs give +24.... Fortunately, real audio signals are not at all coherent and are either transient...or partially cancel in the summing process due to phase and frequency differences. Music instrument material typically mixes to give a 6 to 9db increase in level when 48 inputs are summed together."
So with that in mind, you can see why it's important to have headroom in your mix process as well as your individual tracks. It doesn't matter whether you're using 16 or 24 bit: two coherent signals, each peaking at -6db, will sum together at 0db, regardless of bit-rate.

The thing to remember is that clipping in an analog console is tolerable (within reason), but clipping at any stage in a digital system is absolutely hideous and has to be avoided. Keeping the proper gain structure set up all the way through the signal path will ensure you won't have serious problems controlling your final mix levels.

Which brings me to my question: does anybody know the internal structure of the buses in Sonar (bit-rate)? According to traditional analog summing laws, I should be adjusting a given bus's input Trim to compensate for the incoming signals. Some consoles (like the Soundcraft K2 and MH4) reduce the gain of their summing amplifiers by 6db to deal with this, so the "input trim" is under the hood, so-to-speak. Protools handles summing by having a 48-bit bus architecture. So, does Sonar pass signals from the channels to their destination at the project's bit rate or at 32-bit floating point? Or what?

Bill, I hope I've been helpful as far as mix/record level management. If you have any questions, let me know.... Thanks in advance to anyone who knows the deal with the bus architecture.....

Conrad St. Clair
www.kicksville.com