I'll PM you on my Vibroplex racer. Unless pp is low on your end
It was bound to happen eventually...
PC Win7-64-24G i7-4790k/Cubase 9 Pro 64-bit/2-Steinberg UR824's/ADAM A7x/Event TR8/SS Trigger Plat Deluxe/Melodyne 4 Studio/Other things that don't mean anything if a client shows up not knowing what it wants.
Resolution of digital audio goes down as the signal level goes down.
This has absolutely nothing to do with gain staging or where your faders are set or how loud something should be. Line level is a happy place for recording and mixing. Especially tracking. Track too hot and things will run out of headroom and get crunchy.
What happens in PCM audio formats with low bit depth (8 bit or 16 bit) is that as the audio approaches the low end of the scale, there is the potential for something called quantization error.
In basic theory: an analog signal that passes through a D/A converter has to be sampled and quantized. The sample rate of the target format, eg. 44.1 kHz means that there will be 44,100 snapshots of your signal taken per second. It mostly has to do with the upper limit of frequencies that can be captured. Quantization of the signal is digitized information about how much power the sample has. (volume) When the incoming signal cannot be represented with reasonable accuracy, it gets rounded. The rounding errors, usually called truncation, result in something called "digital noise".
"Digital noise" is a bit misleading. Noise in general usually refers to a sound other than your signal. White noise, pink noise, etc. Static. Hiss. "Digital noise" or "quantization noise" is actually distortion. It generates harmonics. A lot of harmonics. A veritable spew of nasty sounding harmonic distortion. If you want to hear what it sounds like up close and personal, there are several videos on youtube that demonstrate this. If you have a signal with real noise and the signal stops, you can still hear the noise. If you have a signal with quantization noise and it stops, you get nothing. The noise stops with the signal. Digital noise is correlated to the signal - it IS the signal.
A sound wave will oscilate between positive and negative values. In order to get from one side to the other, it has to cross something called zero. This is called the zero crossing.
So anyway, if you're in 16 bit audio or something, the distortion from quantization error only really happens at very low signal levels. 16 bit has a range of 96 decibels, so the minimum signal you can have is -96 dBfs. Things will only really start to distort at around -80 or so. So if you have a sample that lands close to the zero crossing of a waveform, it's a candidate for quantization error. This can happen hundreds of times per second, at any signal level, since all waveforms have to cross zero with rapid frequency.
Getting closer to the top doesn't help the problem at all, because it happens at the bottom.
The solution is dither. Dither is a device that controls the LSB of the audio. So the very first "bit" in 16 bit audio or whatever, is now being controlled by some kind of random probability generator instead of the signal itself. It creates a sound much like broadband noise or hiss at a really low level. The quantization errors are still there, but dither traps the harmonics generated within the noise it creates. Essentially it decorrelates (removes) the distortion from your signal. The result is your actual signal. At the correct level, which in theory doesn't exist. Our ears are not really drawn much to broadband noise at infantissimally small levels. They are drawn to harmonic distortion that sings the very same song.
As an example, if you were to record a 1 kHz sine wave at a level of -100 dBfs in 16 bit audio without dither, you would get nothing. Digital black. No signal. Range is only -96, so you're on the outside. Once the converters start dithering, you'd hear white noise at a very low level. And somewhere underneath that, a 1 kHz sine wave.
Dither should be applied any time the signal gets quantized. It's mostly automatic (unless you're running plugins with bad code or something), so the only time you should really have to pay attention to it is when you mix down and render a project. Some workstations will have it as an option when you render a mix. Others might require you to put a dither plugin on the master buss.
All that is well and good...but the reality is that most every converter and DAW app currently runs at 24 bits...and internally, most DAW apps are running at 32bits.
So lowering the faders in your DAW mixer will not degrade the audio in any way.
Only skimmed through the thread....
I'm pretty sure most DAWs use 32 bit floating point calculations which, I think, eliminates the loss of resolution of low signal levels. I always kind of thought that was the whole purpose of it.
To the OPs original question, I remember reading in the manual for my synth where they recommend leaving the volume knob all the way up because lower volume would have less resolution. In fact, I found the excerpt...
"Turn the QuadraSynth Plus’s master [VOLUME] control to maximum. The best signal-to-noise ratio is achieved when [VOLUME] is set to maximum. This is a digital volume control, and lower settings have lower resolution." They go on to suggest adjusting volume with a mixer.
BUT.... this is a really old keyboard, for sure newer models adjust an amp output.
"I'm pretty sure most DAWs use 32 bit floating point calculations" At the very least these day Mr C!
'Resolution' is a dirty word in many digital audio circles! As was said at the start, one bit is (approx) 6dB and so as bits get fewer, things get noisier but RESOLUTION is always the same.
Mind you, I always want an ANALOGUE level control on the final feed to monitors but that has nothing to do with 'resolution'.
Resolution can be thought of as the number of bits. Since each bit represents 6.02 dB, every time you attenuate by that much you're reducing the word length by 1 bit of resolution.
It can also be thought of as the difference between 16 bit and 24 bit audio. 16 bit can sound amazing. What it can't do is handle a lot of processing. It should be okay for a while before things fall apart but 24 is way less fragile.
I've only seen it suggested to leave the faders at zero and use a gain plugin once. As general methodology it sounds like a myth. In this case it was to run the Airwindows plugins "Console" and "Channel" on your DAW to bypass the 32 bit floating point mix engine (or whatever the system has) and run on the Airwindows 80 bit fixed point summing. It supposedly sounds more like outboard summing.
Whole different kettle of fish. The only reason to do it would be to get those specific plugins to work properly.
I haven't tried them yet so I use the faders.
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