How do you do this, Kim?

[Kim Lajoie]

1) A filter "squashes" a sound signal horizontally by simply eliminating a range of frequencies, whereas a compressor or limiter "squashes" a sound signal vertically by attenuating the entire frequency spectrum of the sound equally (when it reaches the threshhold), correct?

As much as I butcher metaphors (and it's almost impossible to talk about sound without using metaphors), I try to avoid thining in terms of "horizontal" or "vertical". These terms have can have different meanings, even within the same conversions. For example - "vertical" could refer to stacking/layering of sounds, frequency spectrum or amplitude. Likewise, "horizontal" could refer to time domain or frequency domain.

Having said that, I don't think it's right to say a filter "squashes" part of sound. When a compressor or limiter squashes a louder sound, it changes (or moves/shifts it) into a quieter sound. A high pass filter doesn't do any moving or shifting - it simply (theoretically) removes that part of the sound. To squash the frequency spectrum would require a pitch shifter or mroe advanced FFT processes (which I won't go into here).

2) And by using a filter, you can reduce the physical level of a sound while not noticeably reducing the audible level of that sound, right?

A filter can do this if you use it to remove a part of the sound that's contributing a lot to "physical" level but not to "audible" level. Most commonly, this is the case of using a highpass filter to remove rumble from a recording.

3) Saturation is a purely analog device that's similar in effect to a limiter except that it introduces distortion at peak levels instead of attenuating the sound signal at those peaks, correct?

For the purposes of this discussion - correct, except for the "purely analog device" part. There's certainly a lot of digital saturation that can be useful (hard clipping being an extreme example). I work mainly in the digital domain, and I use a few saturation processes that are modelled on or inspired by analogue processes. That doesn't mean it's analogue saturation - it's still digital saturation. It's still mathematical algorithms.

Also cutting hairs, saturation is not a device, but a technique or process.

Hope that makes sense.

-Kim.

[Uchdryd]

When a compressor or limiter squashes a louder sound, it changes (or moves/shifts it) into a quieter sound.

Right; a compressor works just like a person with his hand on the volume knob--except that a compressor is able to look ahead and react milliseconds before the peak.

A high pass filter doesn't do any moving or shifting - it simply (theoretically) removes that part of the sound.

Is there any difference, in theory, between using a high-pass filter and using an EQ with the lows cut out? In practice?

A filter can do this if you use it to remove a part of the sound that's contributing a lot to "physical" level but not to "audible" level.

If you were to record a sound too high- or low-pitched to be audible (below 20 Hz or above 20,000 Hz, if I remember correctly from my physics classes), would this "physical level" register on an EQ or in an audio editor?

That doesn't mean it's analogue saturation - it's still digital saturation. It's still mathematical algorithms.

Hmm. As you've explained, analog saturation has the effect of working like a limiter in that it lets you raise the overall loudness of a recording (it just adds some distortion), but I wonder if digital saturation, which is only emulating analog saturation, also has this effect or if it just simply adds distortion, leaving the peaks too high and subject to digital clipping.

Which makes me wonder, is there a difference between saturation and analog clipping or is it the same thing?


Uchdryd

[Kim Lajoie]

A high pass filter doesn't do any moving or shifting - it simply (theoretically) removes that part of the sound.

Is there any difference, in theory, between using a high-pass filter and using an EQ with the lows cut out? In practice?

It depends what you mean by "the lows cut out". Many EQs have a high pass filter available (mathematically, EQs are really just a bank of filters).

A filter can do this if you use it to remove a part of the sound that's contributing a lot to "physical" level but not to "audible" level.

If you were to record a sound too high- or low-pitched to be audible (below 20 Hz or above 20,000 Hz, if I remember correctly from my physics classes), would this "physical level" register on an EQ or in an audio editor?

Yes.

Generally, you shouldn't worry about signal above 20KHz because it's usually quiet and beneficial to the sound (unless there's something horribly wrong), and if you're working at 44.1KHz you didn't record it anyway. It's the subbass signal you need to be careful about.

That doesn't mean it's analogue saturation - it's still digital saturation. It's still mathematical algorithms.

Hmm. As you've explained, analog saturation has the effect of working like a limiter in that it lets you raise the overall loudness of a recording (it just adds some distortion), but I wonder if digital saturation, which is only emulating analog saturation, also has this effect or if it just simply adds distortion, leaving the peaks too high and subject to digital clipping.

First - not all digital saturation emulates analogue saturation. Hard clipping and wrapping are examples of "native" digital saturation - the inherent behaviour or digital systems without any additional modelling or emulation.

Digital saturation that emulates analogue saturation will generally behave in a similar way to "true" analogue saturation (just how similar is a matter of mathematics and endless debate). Whether the peaks are too high depends on the algorithm and your artistic judgement.

Which makes me wonder, is there a difference between saturation and analog clipping or is it the same thing?

They're not quite the same thing, but they are related. Saturation is a process - the process of overloading equipment (or the emulation of overloading equipment). Analogue clipping is one of the possible end results of saturation. Digital clipping or wrapping are other possible end results. Intermodulation distortionm, altered spectral balance and reduced dynamic range are yet more end results.

-Kim.

[Uchdryd]

Generally, you shouldn't worry about signal above 20KHz because it's usually quiet and beneficial to the sound (unless there's something horribly wrong), and if you're working at 44.1KHz you didn't record it anyway. It's the subbass signal you need to be careful about.

I just today discovered this as I was able to "strip out" the inaudibly low part of a kick drum track using a multi-band compressor.

I tried out what I've been learning here by carefully using compression and saturation on the drum tracks (the only tracks that needed compression, as it turned out). The end result is a louder, warmer-sounding song that has no asymmetric peak issues (hooray!), thanks to a high-pass filter. I simply reclaimed headroom that was being stolen, if you will, by frequencies which were doing nothing beneficial and probably were barely even audible. The dynamics of the song are largely unchanged. It simply sounds better.

Thank you very much, Kim, for the wealth of advice and information you've presented here. Much appreciated.


Uchdryd

[Kim Lajoie]

Cool! Glad it was useful.

-Kim.

[ouroboros]

Just a quick note that a lot of Kim's useful discussions are organized and linked from his homepage - if you haven't already kvrmarked them thanks, Kim.

[gnu23]

Wow, Kim. Thanks for the lesson and open-eyed critique of your own work. I haven't even listened to the track, but after your post I don't need to. You laid it all out perfectly in words. I am bookmarking this thread because it is that good and I learned a ton from it.

[Arglebargle]

-12 RMS isn't overly loud imo. It's pretty easy to get those levels with a balanced mix. TLs Pocket Limiter is the best free limiter I've come across, you can really pump the volume before hearing audible distortion.

[dark_virus]

Uchdryd, the quickest way to kill a DC offset is a highpass filter (DC offset is equivalent to signal at zero hertz). I usually highpass every track in the mix except for kick and bass.

Exactly. I was about to sugest this. But I rarely have DC problems nowadays, due to this very same procedure. (put a hipass filter on every track, but sometimes even on the kick. On bass, with a little resonance)

[eduardo_b]

3) Read everything you can.

2) Learn to listen.

1) Practice.

I'd actually reverse the order of importance, because reading provides the knowledge upon which learning to listen and practice will be most effective. Being able to think through what needs to be done and how to do it starts with knowing the basics. I really like how you've simplified the tool side of the equation and made application the most important aspect of the process. Too many people spend more time looking for and comparing tools than choosing a few tools and learning to use them well.