Audio Process 7 (Time Flow)
In audio Process 7 I am explaining difference between digital and analog compressor.
You can call the analog compressor the Time lead, because this effect only cuts time of your mix/sample/sequence. The main problem with this type of compressor is that your sample is bend passing the time when your note is triggered, and triggers the note in different time creating new and, better or worse ,resolution of your sound. The way it triggers the different time creates is something that is called a sampling flow.
Sampling flow is usually based on zero rates of note velocity, which means that, if we follow logic, this compressor can only be
Part of multiple compressions. You actually need more of them to get the things done.
So how to use this compression tool? Best way to use it to use it with filters. It is especially effective with low pass filter because this way it can act as the part of digital compression.
So you need to set the filter value to set a compression value.
Dynamic filtering is the winner here, but you will have to boost your audio to max first and that's not easy in any way.
Since this is a analog compression, you can set the loudness of your sound first by using the limiter, because the limiter can simulate loudness rates.
The way I use this fact is that I am actually tuning both of them in the master section every time I add something new to the mixer.
So for example, if I start my song with a kick, and hat. I need to change limiter and compression setting before I add a snare or a new hat.
This is very tricky to understand because you would have to change your set up for all your samples and notes every time you add something new.
It can also create an ultra loud sound that can couse a bug in which the app you use is not able to produce sound.
You can find this bug most of time on low end devices where, for some reason some your notes have stopped to produce dynamic/ harmonic audio, or if your device is powerful enough those notes will overdrive and start to play in distortion effect.
This distortion, however, will not be recognized easily because human ear can only hear distortion in the human voice, but for example if you add this type of song in the real sampler or, in to real, mastering studio the guy will tell the you didn't tune your notes right, and the sound is not audible enough.
This is kinda sad and inspiring in the same time.
You can call this effect a time evolving tool.
Digital compression works little bit like the analog compression, but this time you work with numbers rather then nods.
You can say that digital compression allows you to simultaneously set the time when your mix will start playing.
So essentialy it reads your audio loudness in reverse. When the effect finds the loudest part it start to determine which part of that audio has the highest resolution and your compression starts to work.
So how to use digital compression?
Well it's hard to determine which is the best way. So just imagine that you need to reverse the loudness of your mix, and the ending result should equaly perceived.
Your compression is fully dynamic if the ratio of your mix is equal to the numbers on the out signal.
So this gives a good hint how you can improve rest of your loudness with the limiter.
In caustic you can turn the loudest part of your mix in reverse by cutting the loudest part in a form of loop. The loudest part is where most notes are played. Then add it in to a sampler.
Then lower that part on your mixer to zero. When you do that add the limiter in the fx section. Add the compressor in the masterfx part.
Connect those two with sidechain. And start from there if you get things right in your mix, this should work really good.
Many people I think ans give to much attention to the compression in mixing. But remember that you must know why you are doing it, because that's fundamental to get the sound on a right tracking level.