Input Trim Matters
You’re going to understand, at the end of this session, some terms, and some of the basics of proper sound system setup with rational explanations for choices you’ll be making. Let’s begin by laying out the various tools and tests that we will be applying to our mixer. In this case, I’m using an Allen & Heath board, but nearly all have similar behavior that I’ll try to highlight when/where there’s differences that will matter.
I’ll be using a B&K analog signal generator, a Siglent digital oscilloscope, an analog dB meter, several kinds of test leads with fanouts for quarter inch and XLR connectors, SM 58, and mic cord. Later on when evaluating different mics, there will be a speaker employed instead of me singing.
Since a vocal mic input is something that jumps around constantly, it presents a difficulty in measuring on the split second. The o’scope we’re using is going to help a great deal to see these waveforms because of its ability to capture, store, and display short term fleeting events. A mic stimulus, (air movement / sounds), cover a vast range resulting in vast mic output size too. It covers so much range, that a specific notation is employed to describe audio signal sizes, the dB. Ears are fabulous organs. They can discern a fly buzzing in the next room, and a Harley with open pipes accelerating past you. Because of this huge range of volume, (air movement) signals are expressed in logarithmic form… that is, powers of ten. An absolute dB level relative to a particular load is given an (m) behind the dB, and looks like dBm. That (m) refers to “meter”. Dang, makes sense. Okay, you’ve seen that a few places, but you should also know that when comparing signal levels, such as gain or the relative ratio between signals of two sizes, you’ll not generally or accurately use the (m). This signal may be X times as large or different from anther you’re comparing it with, so in this case you’ll say it’s X dB different, without the (m) because you’re referring to their ratio, rather than absolute values. The log functionality, or powers of ten particular to signal amplitude use 20log10 notation, so when measuring amplitude, 20dB means a multiplication times ten. If you amplify a signal ten times larger than it was, you’ve amplified it 20dB. Now, to understand how the dB relates to very large range measurements, another multiplication by ten (100 times total), is 40dB of gain. So, as that dB tale goes, 60dB is 1000 times multiplication or ratio, 80dB is ten thousand times multiplication or the relationship between signal sizes. You’ve gotta have that method of measurement when covering the vast range of the tiny air disturbance of that fly in the other room… to the enormous (you can FEEL it) air activity from that open pipe Harley since air movement is what ears detect. They do so magnificently. The signal size measured directly from a mic (SM58) for a decent singer, at a nice proximity to the mic, at a hardy lung exhalation is equivalent to almost a tenth of a volt RMS. A tenth of a volt would be 100 millivolts. Our stand-in singer is slightly under that at 70 something millivolts. His vocal volume is going to vary… a lot. It will be as soft as a whisper (I hate whisper singers) and a good bit more when he hoots or hees ( I hate hooting and heeing singers, well, except Ron. Ron gets a pass) A practical live event range for a vocalist’s volume will be about 25 to 30 dB. For now though, we’re preparing for his greatest singing volume level that we want out of him. Another measure that’s going to be important to our discussion is Volts Peak to Peak, (VPP). That will come later. So back to our simulated singer’s mic output level of 70 millivolts. That’s not a perfect size to work with, so mixers use a preamp to get it up to a better working level. The good working level within the mixer is typically close to a volt. It should be clear here, that to raise that puny mic signal up to a decent working voltage level for the mixer is going to take multiplying it roughly ten times. Did a light bulb just go off? That ten times amplification is 20dB. Our test lab is going to prove this to you. You’ll see the levels, the control adjustments, and the meters telling you this great news, which is indisputable fact.
All righty, lab 1.
I’ve chosen a tone generator stimulus just to simulate what the signal size that will come from a mic when sung into by a strong singer using the mic at an inch or so from the windscreen. The generator feeds a DI box using a transformer to make our signal a balanced signal like a mic cartridge. PHOTO 1 is generator and DI.
PHOTO 1
The first statement of microphone output measurement we’ll legitimize on the scope in TRACE A. It’s me saying “Hello” strongly (the level you want from your singer) measured right at the mic’s XLR connector. RMS measurement isn’t applicable to a non-constant waveform, so disregard that, and instead, see that the VPP level and the scope amplitude setting is consistent with the simulated singer signal in the next trace. The VU meter’s response will follow the peak characteristic, rather than the RMS.
TRACE A.
TRACE B shows an 800 cycle wave at 74 millivolts. That’s our equivalent singer’s voice for now right off the mic XLR….. he’s holding an “AHHHH” as long as we need him to, to look at levels.
TRACE B.
Lookie at the Trim knob’s position in PHOTO 2. It is set at 20dB. We’re multiplying the input level ten times. (20dB)
PHOTO 2.
TRACE C is the insert point after the preamp and before the channel’s strip with the EQs and the routing knobs. It’s essentially ten times larger than the mic signal. Some loss of precision rests with the markings of the trim knob and the actual gain of the preamp, but that doesn’t scuttle our lab lessen. Take a peek at TRACE C. That signal is 900 millivolts… almost one volt.
TRACE C.
We have our mic sized signal coming in, amplified only 20 dB… we have the PFL button pushed, so voila, the meter in PHOTO 3 reads 0db. Give this a “sink in” moment.
PHOTO 3.
We’ve amplified a decent singer’s mic signal just ten times and it’s zero dB on the PFL meter, and it's just about a volt that properly fits the best level for the guts of the mixer. BAMM
That’s where we want it guys. Strong enough for the rigors of modification and combining with other signals, yet sized to allow somewhat energetic performance increases from our singer without using up the clean zone where the mixer expects signal to abide within. We will rely on the span of the faders to give us the control over this and all other inputs to create our mix. Set this way, placing the channel fader at the zero marking, and the mains output fader at the zero marking, the signal level is right on the zero meter. Mixers do vary as to the actual dB produced at the physical output connectors in my experience, but generally, that board output is going to be plus 4 dB. (a bit lower, 900 millivolt RMS, with this A&H and the tone generator input signal)
Perhaps the most important control in the whole chain is setting the trim correctly. The signal must be large enough to facilitate whatever modifications and combinations you’re going to do with it, but NOT so large that it overwhelms the pathways through our mixer. The first place that can potentially be over-driven is the preamp that raises the relatively weak mic signal up to the working range for use throughout the mixer. That knob (amp) has up to 60dB of gain to increase your nominal tens of millivolts mic signal by. That 60dB is multiplying by one thousand, yikes! There’s the potential for really boning up the input waveform. It’s sometimes suggested that, “some” or a “little” red is fine. You may even find operators that go to red, then back off a little. A pox on these posers, I say! They are beginning the whole trip down our audio path drunk and out of control, destined to wreck against the walls somewhere towards the eventual end of the road to the speakers. I’ve used an analogy for those guys that tell me, “Hey, I get the job done”. I say to them, “Sure, you can actually drive your car in reverse around the beltway… and you might even be Joie Chitwood good at it, but the fact remains, that your car wasn’t designed to be used that way, and will undoubtedly work better, used as it was intended.” Put that on my tombstone.
Back to justifying our Zero level on PFL after the preamp. As you add EQ, you’re making the signal larger, remember how when used at their extremes, those knobs read +12 or +15? Well friend, you’ve just made a part of your signal as much as 5.6 times (15dB) your original one volt level. That’s going to light the next 3 or 4 LEDs of our meter. I hear the defense…. That’s only on the most-loud parts. That is true, and your mixer had this in mind. It has room built in for that… BUT, if you steal that built in margin away making your preamp gain much larger than you needed, you WILL hit the stops, so to speak. Those stops are the voltage limits of the power supply in the mixer. Some are plus and minus 15, some 18, some larger, but there’s still a finite level due to those limits that will result in clipping when exceeded. Here’s where VPP mentioned earlier plays in. Remember that preamp output level of 900 millivolts we got from adding about 20dB at the preamp? That 900 millivolts is 2.74 volts peak to peak (VPP). If, just for example, we use the A&H from our lab session with power supplies that are plus and minus 16 volt, or 32 volts overall….. brace for it….. you have only about ten times your margin left….ten times 2.74 volts relative to 32 volts max turns into just 20dB headroom.
The details…. You don’t get all the way to the power supply rails. There’s about volt from each you should discount for your signal to traverse used up within any amp stage. Nerdie math-wise, 30 volts divided by 2.74 leaves 10.95 worth of multiplication you can exert on the signal till you crash or clip against the maximum. 10.95 is more than 20dB, but less than 21dB. Gosh, that sounds important. This is why I harp so hard over input trim. Low and behold, I grabbed a shot of their specs…. PHOTO 4.
Photo 4
Exciting live mix shows will have over 40dB of dynamic range, 50 if your audience doesn’t yack all through the set. Look at orchestra… they have bonzo 70dB or greater. If you throw away even 10 dB of it in excess preamp gain that eliminates using that much of your headroom, YOU SUCK, (what?, too harsh? I’m old, and you know sound men tend to be grumpy), and you will likely also be hitting the rails.
Am I talking out my butt? I come up with that number by taking a good loud live indoor show of 115db to 120dB (the cops are coming), take away your 40dB to 50dB of program dynamic range, and that leaves a room ambient noise level of 75 to 85. Go measure it… I’ll wait here. Numbers are truth.
Okay, so how bad is hitting the rails by over doing your mix. Clipping consciousness is subjective and can vary according to the ear's perception at different volume levels. It creates hordes of harmonics that were not in the original signal, and to me, is most obvious in the vocals. It's a scratchy, kazoo sounding quality. On an evaluation of power consumption, the duration that a signal is clipping is equivalent to a DC voltage, and during that time, it uses more energy. (AC signal power vs DC power. That's a good subject for a lab in the future.) The power dissipated by the speakers is greater during clipping so there's an equipment longevity issue to consider as well. Clipping matters where you have limited amperage from your wall sockets or range plug that can impact the show.
Here’s a real world example: I walked into a nightclub with a great band playing… yeah, it was loud. Then three times the breakers for the range plug powering the system tripped. That’s death to your show. Well, after the first system collapse, I watched the board and saw how much red was going on…. It was “whore porch red” I say. The solution is simple…. During the periods of clipped signal, the power amps were generating DC voltage, and that calculation yields more amps being drawn than the clean AC signal would have consumed and substantially more than the circuit breaker could deliver. I shouted into the ear of the band’s agent, “I can fix that”, and he pulled that sound man off the board and shoved me in. All I did was reduce trim and fader levels while restoring monitor feeds and increasing outboard gain eliminating the clipping all over the board. I “cooled” off that guy’s excess trim level settings and immediately, that show became dynamic, intelligible, the breakers didn’t trip….. and I made it louder than is was when I walked in, possibly twice louder. Aright, a “yeah me” moment, but proof that using large signals inside the mixer is a real problem that is avoidable using better gain structure methods.
Some may ask… “So, you made me get rid of my signal…. now what shall I do?” The answer is greater gain AFTER the mixer, and more speaker diaphragm area. In older days you had a system EQ and a compressor chained before the crossover in an outboard rack. They each had some gain range to kick it. These days you’re driving a powered speaker right out of the mixer. Selecting the additional gain here is okay, up to the point that you’re still not moving enough air and the amp in that box starts banging the speaker to the end of its excursion limit. Ultimately, speaker area will always be the key.
You can get the same volume out of an 8 inch powered box as you can from a double 15, but that little 8 incher has gotta be moving way more in and out than those 15s. What makes the little speaker move further is greater gain. Unfortunately, gain is an evil force. Theoretical gain suffers real world limitations and inaccuracies due to mechanical and electrical reactive behaviors that make large gain unstable. Much of ringing problems are due to too much gain for either electrical or acoustic stability. Your remedy is more speaker area…. more boxes of speakers, more power amps driving those speakers with moderate gain rather than high gain.
What you should be able to take away from our lesson:
Good singers give good signal.
A typical vocal mic produces the equivalent of a tenth of a volt.
You only need 20dB of gain to bring that signal to the level it should be internally to the mixer.
You want your headroom to stay clean.
A clipping signal consumes more amperage from your power source than a clean signal, and is harder on amps and speakers.
You want a lively large dynamic range show.
Gain has an evil cost.
You need the appropriate amount of speaker area for a proper show at the level you desire.
These sessions will include my rants, but there’s value here that will benefit your improved skills and your understanding. This is intended for those who have struggled with system problems and control, or who may just desire to learn a hobby that I find to be extremely satisfying.
Some of my day career duties included tech writing, so this project has a nostalgic happy feeling for me. The only reference material used was a dB/multiplier chart (cause I didn’t trouble to actually do the equations), https://www.technicalaudio.com/reading/handydB.html, and the schematic with Spec data of the mixer to get the power supply info, WZ16-WZ12-MK2-DX-SCHEMATICS.pdf. I’m an Electronics Engineer, and make careful effort to mockup, energize, then actually measure what I claim, as well as sharing from my professional education and experience. Every bit of this is original content of mine from education and experience, with no plagiarizing of any other’s thoughts or writings.
Future sessions will cover mic models, amp gut stuff, speaker jazz, processing gear, compression, mixing tips, some theory, and yarns from years of both pain and glory.