Here’s a good place to start… Link —VCM Compressor 376

Link — Equalizers Part 1

Link — Equalizers Part 2

Link — Equalizers Part 3

Maybe I should be using the EQ more freely on every Part? I try to stay with the original Voice as programmed, but maybe that philosophy is not appropriate. So compression SEEMS like a better tool -- save for my total ignorance about it.

How the original Voice sounds as (originally) programmed does not take into account *how you are using it*. As you will (hopefully) get from the articles, you have EQ to fix issues within the individual Waveform, but you also have other EQs that are designed to help the instrument blend with its surroundings… You must consider what other instrument is fighting for the same frequencies… you will find that depending on the musical part that is being played the acoustic piano, sooner or later, is in every other instrument’s range. Lots of crunchy guitars will call for one kind of EQ strategy, while if that same piano is surrounded by lush pads and strings this will call for an entirely different approach.

Use your ears in each and every case. As soon as you start thinking there is one setting that will work in all situations … you would be wrong.

Thanks. I am going to read and re-read the Compressor article a few times to get really familiar with the terminology and explanations and examples. And I'm already getting some good information from the EQ articles.

One follow up question, please.

I play live with my MOXF supplying MIDI backing tracks and whatever Part I'm playing as well. So my MO for Mixing, Voicing and recording sequences in my rehearsal space is to run everything through my stage Mixer (MG 10XU) but leaving the knobs dead flat and only use the mixer EQ for adjustments at the actual gig.

I mention this as a lead up to a question about the Gain knobs on the MG 10XU. I run my Left and Right MOXF sends to two mono channels (3 and 4) on the mixer. Those channels have 3 knobs of EQ (instead of only two when running through the stereo channel). This also gives me a Lo Cut for each channel.

AND it affords me a Gain knob for both channels.

Should I leave that Gain knob at 0? Or at the little V area around 8 o'clock? Does it matter? There is clearly a lot of volume to be had with the Gain knob and a Line input, but is there anything to be gained (pun intended) from using some or a little or none of the knob? It SOUNDS to my (often unreliable, due to Hi frequency loss) ears that it "improves" things somewhat to set the Gain at the V -- but that could be a total aural illusion.

Here is some advice from me.

Do some web research on General Compression. I'll warn you it is quite a big subject, can be quite technical and there are a multitude of Applications. I recommend understanding Compressors and Compression before using them and expecting to see any benefit.

There is an adage that a compressor is working its best when you can't hear it... until you turn it off and "something good disappeared".

Because of it's subtle nature, people try it, can"t hear any difference and immediately abandon it as a tool, especially if they have to buy one to have one.

TLDR...
I will run through some rudiments to get you going.

A lot of people see Compressors as a "magic" tool for adding sustain.

They don't *add* Sustain... bear with me here. What compressors are actually doing is Boosting the Volume of the decay portion of a note above what you would normally be able to hear. In effect you keep hearing it for longer. But if the instrument itself is not sustaining, a compressor won't help in this respect... because it cannot boost a sound that is not there to begin with.

How does it just boost low level signals? It doesn't, it boosts all of your Signal. That is Attack, Decay, Sustain and "Release". Some instruments continue sounding after you have "released" the note... Stringed instruments are a good example... Piano is a great example.

But If it Boosts ALL the sound, then the already high volume Attack would be deafening or clip the input of the Amp?

No, here is the magic. You set a Threshold Level, and if the Signal Attack Level exceeds that threshold, the Compressor will Trigger (Switch On). The Compressor will reduce the volume of the Signal ONLY above the Threshold you set. You instruct the Compressor to "limit" or reduce the gain of the signal above the Threshold by setting its Ratio (Gain Reduction). The smaller the Ratio, Attack peaks (aka Transient Spikes), will be pulled down in volume only slightly. High Ratio settings will "Clamp Down" and reduce the transient volume drastically. Ratios above 20:1 (or infinity) "Limit" the Signal to the Threshold Level, i.e. the signal level cannot exceed the Threshold.

Compressors with a High Ratio setting are said to be acting as "Limiters".

All Limiters are Compressors, but not all Compressors can be Limiters. Take note.

Remember, all the while, that the lower volume "Decay/Sustain/Release" part of the signal is still being Boosted, but not compressed (assuming it is below the Threshold volume). This is how you get "Sustain" without turning the Amp up to 10. .

But, this lowering of loud, and boosting of quiet, reduces Dynamic Range. Softly Played Notes are heard Louder. Harder Played notes are heard Quieter.

There is a break point where everything you play, soft or strong, appears as the same volume i.e. there no dynamic range at all. So if Dynamic Range is important to the composition (Light and Shade, Soft and Strong) then you need to be careful how you use a Compressor.

The Gain Reduction can also be set with the Threshold Level (Sometimes called Sensitivity, and maybe Input Gain depending on design). But if you set Threshold Low (or Sensitivity High) everything you play will trigger compression which can lead to unwanted distortion or a "dull, lifeless sound".

The trick is to Balance Threshold and Ratio to get the desired effect for how you wish to use the Compressor.

A guitarist may well plug his guitar directly into a compressor (before other effects and Amp), to give some dynamic control, fluidity and usually some punch.

A Record Producer will likely use Compressors "Post Production" on each instrument, to level and balance multiple tracks such than none "spike out" or equally "don't get buried" in the mix. Everything attains a certain consistency (this is addition to EQ and Levelling). A compressor will usually reduce "thin harsh treble" and leave a generally warmer, fuller sound, so in this respect it can act both as a Level and EQ effect.

A Studio Engineer might specifically target an Instrument for effect... For example, Recording Drums with "Parallel Compression" is a technique you will certainly read about (Dry "Blend" controls are used for this).

Attack Control... delays the onset of compression after being triggered, to allow some of the Transient through... restoring Punch, Dynamics and "Percussion".

Release Control... delays the Release of Compression beyond the signal falling below the Threshold, usually to keep it compressing through the "next attack"... for fast "flowing" Legato Phrases for example.

Attack and Release controls usually need to be finely balanced also.

I eventually bought two external compressors (for guitar mainly) which i can vouch for...

Origin Effects - Cali76CD ( based on the UREI 1176)
Boss - CP1X (Multi Dimensional Processing, Multi-Band Compressor).

One follow up question, please.

I play live with my MOXF supplying MIDI backing tracks and whatever Part I'm playing as well. So my MO for Mixing, Voicing and recording sequences in my rehearsal space is to run everything through my stage Mixer (MG 10XU) but leaving the knobs dead flat and only use the mixer EQ for adjustments at the actual gig.

That is exactly the correct usage of the EQ on your mixer — to make adjustments for the actual room you are performing in. If it is a basement club, with rugged walls and floor, the mixer EQ is the place to make this adjustment. And conversely, if the room is too lively, the mixer EQ is where you want to fix this. Hopefully, this does not require a midrange adjustment. (But we’ve all have played gigs in rooms that are totally inappropriate for music).

I mention this as a lead up to a question about the Gain knobs on the MG 10XU. I run my Left and Right MOXF sends to two mono channels (3 and 4) on the mixer. Those channels have 3 knobs of EQ (instead of only two when running through the stereo channel). This also gives me a Lo Cut for each channel.

AND it affords me a Gain knob for both channels.

Setting the GAIN is scientific, there is a right and wrong and you can take the guess work out of this, entirely… your synthesizer is a LINE LEVEL Source… approx. +4dB Output. MIC/LINE is a selection that prepares the channel for *how much* signal is getting ready to enter the system.

Using any of the first four channels is fine (provided you setup for Line level signal), Channels 5/6 and 7/8 are for LINE level signal -10dB and stronger… while channels 1/2/3/4 can be either.

Obviously the actual level of the synth is in motion and constantly changing… but in general your synth has way more output signal than the microphone — in a dynamic mic a tiny amount signal is generated by a permanent magnet mounted on a diaphragm which moves in a coil of copper wire…this can generate a signal as weak as -60dB; your synth which is typically +4dB will need less gain boost than a microphone.

You see two dB scales written around each GAIN knob… they are coded to match the PAD. The Pad adjusts between MIC and LINE level signals — the pad ‘protects’ the electronic circuitry from you sending too much signal through the mixer channel. Notice it applies a -26dB attenuation… two minimum-to-maximum scales are indicated… one for Pad engaged, the other for Pad disengaged.

MIC are weakest of all signals… typical dynamic microphone generates a very small amount of signal and therefore requires no pad, but plenty of boost — Line level devices, like your synth, require less boost, if any. You might need to engage the PAD and then increase the GAIN knob to adjust the right setting. How do you know when it’s right?

How you can know is to use the PEAK light indicator on the channel… it will flash when signal comes within 3dB of the maximum you want to send through this channel. That PEAK indicator is just after the channels EQ and just before the Level knob. (EQ will affect its response).

The ideal Gain is achieved when the PEAK indicator flashes/flickers on and off. A steady light is bad (consider that the channel’s electronics screaming in agony)! . You want that indicator to flash ON only momentarily and only at the very loudest peaks of your musical performance.

When you have your channel’s PEAK light dancing happily you are good… this can be set without even hearing the synth. This is not subjective, this is your scientific setting! When set properly you can then begin to musically mix the sound levels of your channels using the Level knob.

In general, raise the channels Level Knob (this is equivalent to a channel Fader on bigger format mixers) to the chevron (arrow)… this is the nominal amount, this valve is used to pass signal to the Output stage and serves as the *artistic level control*… you adjust slightly above or below that chevron, as necessary, to create the subjective balance (mix) of the different inputs. It is this Subjective balancing of musical levels that feed to the main stereo output.

The sum of all channel Levels feed the mixer output… the theory of operation is deliver to your amplifier *enough signal* for it to work with ...
_Feed it too little, you wind up turning up your amp, making it work unnecessarily hard… result: a noisy mix.
_Feed it too much, you risk clipping and potentially damaging your amp and speakers.
_Feed it the correct amount (as indicated by the Peak lights and the main output meters) and you will have clean, loud signal and a happy system.

Turn the amplifier up enough to fill the venue with clean awesome sounding audio. Do not set the amp level without being in the audience. You cannot tell from the stage how loud it is to the folks out front. This is what soundcheck is supposed to accomplish. Go sit in the audience and playback a sequence or something — or have someone play your rig. (Or better, have a sound person).

Should I leave that Gain knob at 0? Or at the little V area around 8 o'clock? Does it matter? There is clearly a lot of volume to be had with the Gain knob and a Line input, but is there anything to be gained (pun intended) from using some or a little or none of the knob? It SOUNDS to my (often unreliable, due to Hi frequency loss) ears that it "improves" things somewhat to set the Gain at the V -- but that could be a total aural illusion.

It matters! When the device is fed enough signal, it can deliver the right amount to the next device and so on… Gain staging.

If you distort at the input Gain stage of this chain, no matter how much you lower the Channel Level, or the Stereo Out, or your Amplifier, the signal remains distorted.

When teaching audio engineering, I used to use water going through a series of pipes to help student visualize gain staging…a water system with various valves to control the flow through the system.
Deliver too much water to the first set of pipes, you’re in trouble from that point on. If distortion is like a red dye that gets in the water… if it enters at that first INPUT GAIN stage, the water will remain red throughout the entire system… no amount of correction in flow will remove that red dye… turning the flow down later in the system just means the flow is lessened (it’s still red/distorted, it’s just distorting at low volume)!

By generally, setting the channel Level at that suggested ‘v’ mark, you are feeding the overall main stereo Out — this means there is no increase or decrease from the Input Gain stage. When you have several channels active simultaneously, adjustments, up and/or down from that point are available to make an artistically pleasing musical balance… you want to have your MG10 XU stereo Out meter dancing happily - just short of the red (always avoid red, that’s your gear saying “Ouch!”) by providing a good healthy amount of signal from your mixer means your amplification stage will have enough signal coming in to work with without unnecessarily having to strain. When amplifiers strain they are noisy. Feed them and they can perform. Nothing sounds as good as a sound system that can part your hair with LOUD volume, without sounding like it’s struggling to be there. Like a muscle car, it can do 150-160mph without any effort effort. Your underpowered car may get there but it feels like it’s gonna fly apart at any minute.

It is Signal to Noise ratio… maximize Signal (short of clipping)… minimize Noise.
In the pipe system - there is water and air, in our audio system there is signal and noise.
You cannot eliminate all noise, what you want to accomplish is to have enough signal available at each stage so that the noise is negligible.

Negligible is a technical term that means you can ignore it, it is insignificant enough as to not be an issue, at all.

Hope that helps. Thanks for the questions.

————

Edit— by the way, the single knob Compressor you find on the mixer, represents a lot of R&D to make using Compression easier for musicians. As noted in the article, and in Antony’s post, Compressors are difficult to use because they should be felt not heard. If you can hear the compressor you have too much.

That is old or original school thinking… compressors we’re originally used as a utility, not an ‘effect’… but guitarist, as you know don’t hear noise and distortion the same as normal beings (haha)…. They use the fact that by *reducing* the transient peak you can create the illusion it is staying at the same volume longer… it’s an aural illusion, that works.

The one knob compressor on those channels are there principally for vocal microphones. The settings along its travel from minimum-to-maximum are designed to work for a typical incoming vocal mic… it is actually changing several parameters simultaneously — following a curve of known good/working presets; as you turn the knob you can set it to taste (rather than having to know how to setup Input, Gain Reduction, Ratio, Output, etc… the “known curve” delivers a set of known working changes… you turn it until it feels right.

Knowing what to listen for comes with experience. When you realize it is dealing with soft-to-loud as a range. “Soft”, however, does not mean 0 or silence. Not at all.

Sounds strange, but this is at the center of the mystery… soft means the level of signal that corresponds to the softest useable/desirable sound. When you strike a note on any instrument the softest useable sound = Soft. The Loud is easy, every one can listen and set the Loud — it’s in your face, and metering helps. But listening to “the soft” takes knowing to do so. You have to concentrate on whether the soft of the musical input is intelligible - among the other instruments.

Example, a singer during a song might go, literally, from a whisper to a scream. Imagine a 17-piece big band and the singer on a mic… Setting the levels for the scream is easy. Just don’t let the meter peak and you’re good.

But what about that whisper, how can that also be heard with all the horns, and drums etc., whispering is the soft. Left without compression, the loud/scream is many time stronger than the whisper… the whisper will be lost because the Dynamic Range between the Loud and Soft is too big. You would have to raise the gain when the singer whispers, and lower it again immediately afterwards,… “riding the faders” is impossible, as an engineer you simply cannot respond fast enough. Enter your help: Dynamic Level control via the Compressor.

Compressing that soft-to-loud Dynamic Range is the job of the Compressor… as it is applied, it shrinks the distance between Loud and Soft. When applied the Loud hardly increases, as you bring up the Soft.

So you want to listen for “the soft” coming up in level. The Soft will get closer to the Loud… as you *compress* the dynamic range.
The entire signal initially lowers, but that is what OUTPUT parameter is at the output of the compressor — to restore “unity gain” (the level of the original Input). You compress the range, then restore its overall level. It will be ‘tighter’, somewhat ‘punchier’, more ‘present’, phatter, etc. Comparing original with your settings as you make then is the way to proceed. You want to hold the loud signal back from having to be raised in order to hear that softest useable sound level.

So you must train your brain to listen for the Soft getting louder… sounds strange to say, but that is what compression is about… bringing the soft sounds up so that they are more present in the mix. So that they are not so far away from the Loud.
OverCompression make soft sounds sound bizarrely out of balance. Breathing sounds uncommonly loud, lip smacks sound like rimshots…

I have written below some anecdotal side notes which may be of interest. You will need some rudimentary "electronics" knowledge. If that is beyond you, don't read any further.

Recording is all about turning natural sound (a voice, a plucked string, a blown pipe) into an electronic "audio" signal that can be recorded via electronic means. That conversion is usually done with a microphone, or magnetic pickups in an electric guitar... they are said to be transducers.

But the source could be electronic to begin with, e.g. a played back "Tape" or a Synthesiser.

The "loudness" of an electronic signal is effectively measured in Volts/Voltages. But its all relative, so rather than absolutes (Volts), Decibels are used... which is a different topic.

But, on a single source, a higher voltage represents a louder signal which has the ability to "drive" a "device connected to the output" harder, and ultimately make a louder sound.

A good and very obvious example of an output device is a Speaker Cone. But it could also be an Amplifier, amongst many other examples.

When we think of Amplifiers we typically think of a black box with an Input jack and either an output jack or an actual speaker.

But in reality, that Black Box (usually) contains multiple Amplifiers. These amplifiers are staged (or cascaded) with each "stage" ratcheting up the Volume (voltage) a little more, such that the final voltage is high enough to "drive" big magnetic speaker cones (a PA for example).

A good example is an Electric Guitar Amplifier. The Voltage generated by the guitar pickups is tiny, yet the sound coming out of the speakers can be literally deafening.

The tiny input voltage, is amplified to a higher voltage by the "Pre Amplifier", such that its output voltage is sufficiently large that it can "drive" the Power Amplifier. The Power Amplifier is typically more "efficient" and can significantly boost its source input voltage to a level that can "drive" a Speaker Cone. It is said to have a larger "Linear" operating range, than a typical "Pre-Amp".

We need to take a look at that word "Linear". It refers to a graph of Input Voltage vs Output Voltage. The line on the graph is "Straight" i.e. Linear.

You often hear the term Amplifier Gain. Gain is simply the multiplication factor of an Amplifier. If an input of 1 Volt results in an output of 5 Volts, the Gain is 5 (or x5). The term Gain is misused these days and commonly understood to mean "amount of distortion".

That is because, very often "cranking" the Gain Control results in distortion. In truth, turning the Gain Control above its "recommended" level, causes the amplifier to stop operating "linearly".

Example with 1 volt input

Gain 2 = 2v out (linear)
Gain 3 = 3v out (linear)
Gain 4 = 4v linear
Gain 5 = 5v linear
Gain 6 = 5.5v non-linear*
Gain 7 = 5.6v non-linear*

* The amp is starting to "compress".

Distortion literally means that the output waveform is "not identical" to the input waveform.

When an Amp is operating in its Linear "zone" , the output IS identical to the input, except it is bigger (louder, higher Amplitude).

Pre-Amps, Power Amps... more black boxes. But each of those may also consist of multiple Amplifier Stages.

The whole point of "Staging" Amplifiers is such that each Stage can operate "linearly" (not be overstressed) So each stage can "lift" the signal a little (think of a Stairway) and hand it off to the next Stage. The aim is to have an Amplifier that is High Gain (very loud) and is still Linear.

But, reality creeps in.

Each Stage is extra cost and price. So fewer, "mostly linear" stages are employed.

Also it was historically difficult to make "truly" linear amplifier components, such as Vacuum Tubes, and later, Transistors.

So Amplifier design became more a case of acceptable tolerances than true linear operation.

Even to this day you will often see Amplifier quoted "Nominal" Input and Output specs. These are "Normal", "Expected" or "Recommended" operating levels.

As well as "cranking the Gain" you can also force an amplifier to behave non-linearly by feeding it a "Too High" Input (Red line VU Meters are used to check this).

An Amplifier actually has fixed "available" voltage supplies at both its Input and Output. Lower Voltage on the Input, Higher Voltage on the output.

It cannot go above these, they are fixed , or all that is available.

So an Input that is "expecting" no more than 3volts, is fed a signal that peaks at 4volts. The Input "clips". That means the Peaks cannot actually reach 4 volts, so instead are "limited" to 3volts. But in reality, the Amplifier "tries" to find 4 volts by pulling down (borrowing) voltage from the supply, which takes it away from the output (sometimes known as Amplifier Sag).

So the Amplifier has been handed a signal which is "too high", which it tries to accomodate. But it now has to Amplify that already Too High signal which will now likely also clip the output. Lets say the Output Supply (Fixed) Voltage was 12volts, the max Gain was x4 (3v x4 = 12v). So you have amplified 4v x 4 = 16v... 16v Peak "requested" at the output... it tries, fails and clips. It managed to find another 2.3v but the output clipped at 14.3v.

At this point the Amp has been "requested" to operate outside of it's recommended Linear Operation.

It is now "Soft Clipping" and COMPRESSING the signal.

Earlier we talked about multi-stage amp design with the output of each each stage driving the input of the next stage.

If the Gain on any given stage is turned up so that its output is now "too loud" for the Input of the next stage it is said to be OVER Driving the next stage. Overdrive... you've heard of that. Overdrive can cause a chain reaction. Mild overdrive (in valve amplifiers) results in soft clipping warm compression. Pushed further it will result in hard clipping "crunch" and ultimately complete "square wave" distortion.

So compressors, if you like, are poorly designed "linear" amplifiers. Except of course, they are very cleverly designed to give you full control over the phenomena of compression, witnessed in non-linear amps.

Final point. As guitarists we all love what we call "Natural Amp Compression", and in part this is also the reason we still love the "old" Valve Amplifiers. Although the intent of Valve Amplifiers was to be "Loud and Linear", their design was full of great sounding imperfections (Leo Fender could never understand this guitarist mentality).

The problem is, to get to those non-linear "compression" imperfections, you had to turn the Amp volume up to deafening levels. Remember, compression also allows for more "sustain" which is an added benefit.

These days, to preserve our hearing we "fake" it with a Compressor Pedal.

And on top of that you get your "Overdrive", "Distortion" and "Fuzz" pedals... all fakes. All trying to mimic the sounds of Tube Amplifiers going into ever increasing meltdown.

The (Synth) onboard Yamaha VCM 376 Compressor, as far as I can tell, is modelled after the UREI 1176 Studio Compressor (Limiting Amplifier) which is a very highly regarded weapon of choice among guitarists (including me). There are number of Guitar Effect Pedals also modelled on the UREI 1176... they usually cost a lot of money.

So you Yamaha Synth owners can consider yourself fortunate.

🙂