A useful thread like this pops up every now and then. A lot of great stuff has been said, so people can learn something.

For me it is quite simple since years ago. I had read all kinds of papers about digital audio such as this one by Dan Lavry, and there's also plethora of great info on the subject at Bob Katz's website, so I decided to record everything at 96/24, mix everything at 96/24, and master everything at 96/24. Before mastering I archive the track in 96/24 FLAC, and I can convert this HQ file at any time to any other format that is needed easily later and master it.

I use lots of dynamic and saturation effects, so it is even more important that I do everything @96/24. If you use only analogue hardware [I use both], you won't notice that much difference, but if you mix itb with lots of compressors, saturators, and distortions, you absolutely will. Even when I record something @ 48/24, and I do, I resample it to 96/24 with R8Brain before mixing @96/24 because all the digital artifacts tend to accumulate with the number of plugins, and it just sounds better that way.

I noticed these days that 48/24 is taking over 44.1/16. Personally, I almost hate 44.1/16 with passion... but for loads of music that has been released in last 20 years 44.1/16 is like a well deserved sentence, as it wouldn't sound any better at any higher SR or bit-depth anyway.

 

Last edited: Dec 3, 2019

Well, yes of course. But the point was what has (a much) greater impact on sound reproduction. And definitely it is bit-depth, by far.

 

Last edited: Dec 3, 2019

A lot of people say that. But it makes me wonder what is the purpose of oversampling feature in CD players and computer audio players (like foobar2000, Winamp, etc.)? And it 'works' (changing-sound-wise)...

 

48kHz / 24bit and forget the rest, on the net u can read alot about no sense going over the 48 kHz

 

After some quick search I found this:
An oversampling CD player might result in better D/A playback, this is a function of the converters, not the data on the CD.
But also:
- 8x oversampling filters can only cut frequencies between 22.05kHz and 330kHz. Everything above 330kHz remains untouched, etc. for other x times oversampling.
- with an average jitter environment, oversampling even decreases accuracy within the work environment. In short, oversampling does not preserve the original 16-bit precision data.

 

For me it doesn't matter as I make music for myself only. Right now my DAW is set to 48 khz. If I was a professional I would probably care more about that sort of stuff.

 

Correct, because artefacts can also occur during the conversion from digital to analogue, which falsifies the reproduced material.

 

The conclusion on players' oversampling would be oversampling is good for bad chips [the ones we all have]:
The origin of oversampling he described in a report: "We at Philips argued that it was impossible to redesign the finished 14-bit converter to 16 bits in a short time. But my colleague Karel Dijkmans said: No problem, I know a little trick to turn our 14-bit converter into a 16-bit converter. The trick is called oversampling. Marketing will then make the virtue out of necessity.

The more building density grew in the further development of the chips, the more functions were implemented, the more the chances of correcting the wrong direction had disappeared. Today, there are hardly any converter chips that do without built-in oversampling and complete output stages of the simplest design. Although these converters make it possible to cheaply produce millions of small cell phones and MP3 players. From the ideal lifelike rendering they increasingly remove. Thus, they have nothing to look for in a high-quality music playback device. The dilemma of developers today is that they can no longer rely on other chips.
So whenever we talk about oversampling we must consider three stages (and their dependencies)
1. DSP, 2. DAC, 3. Analog circuits.

 

Last edited: Dec 3, 2019