finished, mastered -14 integrated lufs (Db max ceiling boosted for bandcamp)

The most profound difference between digital and analog audio is, digital audio's reliance on sampling rate. Sampling rate is inseparably bound by hard math. Which has higher resolution? Digital or analog? Let do the math.

First off, sampling rate is not sampling range. Sample rate is the set number of samples taken per second. It stays the same, no matter what frequency of sound being recorded/rendered. Higher sampling rates aren’t for recording higher frequencies (like ones you can’t hear). They are for recording/rendering any audible frequency in HIGHER RESOLUTION. I reiterate, sampling rate is the SPEED at which a computer samples audio. Not the frequency range it samples! The higher the sampling rate, the higher the resolution (sound quality).


The real math is simple: Divide the sampling rate, 44.1kHz (most common/standard), by the frequency of the sound being recorded/rendered, say 60Hz (bass kick), and you get the number of samples per cycle for that particular sound (735 samples per wave cycle in this case). That means, when you record/render that 60Hz sound, the computer has no more and no less than 735 steps to estimate the shape of the waveform’s fundamental cycle. Like if you laid out 735 square blocks in the shape of a wave. The blocks can be what ever size you want, it's the number of them that's important. Here's an example of a single wave cycle (the shape you would be trying to draw with your blocks): 1.bp.blogspot.com/-Zer0DTeKD8o/Us0Qc4TIE-I/AAAAAAAABqY/3hoUpAh81To/s1600/sinewave.png As you can see, the more samples you have per cycle, the better you can emulate the curves of real life sound waves. It’s EXACTLY like the number of pixels in a picture. The more pixels/samples, the higher the resolution of the sound/picture.


The higher the frequency of the sound being sampled/rendered, the more cycles that particular sound has per second, thus the fewer samples will be taken of each cycle. THIS IS BECAUSE SAMPLING RATE FUNCTIONS AS A CONSTANT/PER SECOND. A 6Khz sound has 100 times more cycles per second than 60Hz. So out of your 44,100 samples taken per second, only 1/100th of them will be made for one cycle in the 6Khz sound. Compared to the 60Hz sound, sampling a 6Khz sound at 44.1Khz sampling rate, gives you only 7.35 samples per cycle to construct your waveform. Try drawing any kind of convincing curve with only 7 blocks or a coherent picture with 7 x 7 pixels. It can’t be done.

It is established science that the universe is fractal by nature. MOST SOUND IN THE REAL WORLD IS MADE UP OF COMPLEX WAVEFORMS THAT INCLUDE SMOOTH CURVES. These waves contain fractal harmonics extending well into the high frequency range. The fractal nature of the physical world is a proven fact. To say sound is the exception, is wilful ignorance. It is also well known, detail, definition and clarity in audio reside in the high end. Go into your daw and try listening to music with a -48db/oct high cut filter at 6Khz and see just how "little" important program material you are missing. So, at 44.1Khz sampling rate, how many samples would you get if you recorded a 16Khz sound? Answer: 2 or 3. That’s it. 2 or 3 blocks. You literally can do NOTHING but draw a simple square wave with that few. Just 2 giant blocks. One up, one down. There's your wave cycle. A simple square wave.

So in the audio bands most important to clarity and detail, digital audio has the WORST resolution. Worse than an 8 bit 1980s Atari console. Yet, SOMEHOW digital audio is toted as "superior" and "more advanced" than analog. This counter intuitive attitude couldn't possibly stem from digital audio's endless editing capabilities could it? Allowing executive controlled musically talent-less people to be transformed into top charting pop stars? Oh wait, THAT'S EXACTLY WHERE IT STEMS FROM. Auto-tune, time correction, quantization, piecemeal production, midi control, algorithmic processing, per-constructed loop libraries, AI created patterning, instruments programed to play themselves. It is a technology most suited to fain musical talent. This effectively renders real talent "financially undesirable" as executive producers would rather hire a "face" and one engineer to fake the music, instead of hiring real artists/musicians to give real performances with original ideas.

The application of the math is easily proven by entering your desired frequency and downloading the sound file from a site like onlinetonegenerator.com Sine wave shape should be the easiest to work with and shows the degradation that happens as you sample higher and higher frequencies using 44.1Khz sampling rate. Open the file in audacity. Zoom in to find exactly ONE wave cycle (as is shown in the linked sine waveform above), then count the number of dots. Each dot in audacity is a sample. This is how digital audio works. Because of the math sampling functions under, the resolution of digital audio goes exponentially down the higher the frequency of the sound you are recording/rendering/reproducing. Digital resolution is very high in the bass bands but practically non existent in the higher bands. To get high resolution across the entire audible spectrum, you’d need sampling rates in the megahertz range. Not kilohertz. 8Mhz sampling rate would be adequate. That’s 8,000,000Hz not 44,100Hz. Bit of a difference there. With modern technology, it should be possible but since everyone’s convinced 44.1Khz is fine, I’m not holding my breath for it to change.


This is why engineers who know how digital audio works, facepalm when people say analog has inferior resolution. Any who claim otherwise are usually saving face in front of their corporate sponsors. Digital has better editing capabilities by far, but dreadful high frequency sound resolution. Modern digital sampling can’t even accurately record/render a 6Khz complex waveform, let alone a 16Khz one. Analog gear can though. Because it doesn’t rely on sampling. In analog, recording is a straight shot of electro/mechanical physics. A good reel to reel or vinyl has MORE resolution than any digital audio to date. All those hard edges on the digitally estimated waves also produce distortion (mostly odd order). Just play with a subtractive synth for a minute and listen to how different a square wave or saw tooth sounds compared to a sine wave. Edges matter. Wave shape matters. Accuracy matters. It all adds up.


Most people don’t seem to know these things. Nor do they want to. Usually subscribing to scientifically unfounded techno babble they heard on youtube, like: "there aren't any curves or sine waves above 6Khz in music". Since computers are the primary tool these people would use to study sound waves, and, as proven by the math, a computer can't adequately detect curves/sine waves above 6Khz while sampling at 44.1Khz, how would they know?

HOW WOULD THEY KNOW?

Did they tirelessly sample all sounds of music in the world at 8Mhz in any attempt to find out? No. NO THEY MOST LIKELY DIDN'T.

They ignore the fractal nature of physics. To boot, they have no real reference for their claims, so it’s not all that surprising. Thanks to digital audio being made the standard, most analog gear is too expensive to buy, too old to function properly, or too hard for noobs to handle. Almost all people hear these days, is digital or analog that has been digitized for the internet. Digital decoders have a few tricks to mask the poor high frequency resolution and noise. However, pretending decoders are perfect (which they aren’t), even if the decoder could completely make up for noise and wave warping, you’d still be missing what the computer didn’t record/render in the first place. That being all the details and curves of those higher frequency sounds, as they get mangled or left out entirely. There’s no way around it, you loose definition/detail when you digitize.