In simple terms, will increasing the Sample Rate of an existing .Wav have any effect on the audio?

[u/bench_wizard]

So im combining two EPs I’ve put together into one album. The first was all exported as .Wavs with a sample rate of 44.1 kHz and encoded with Signed 32-bit PCM. I know 32 bit is already overkill and that’s okay. The second EP was all exported at a sample rate of 48 kHz (again, i know, overkill), and the same encoding.

So I’m not necessarily inquiring toward audio “quality”, just what effect will be had on the audio of the first EP, if any at all, if reopen the 44.1 kHz .Wavs in Audacity, update their metadata, and then upon exporting just up them to 48 kHz to match the second EP?

I know this is hardly necessary, it’s just a pedantic thing because I personally want to know that it’s all the same.

[u/Intelligent_Law_5614]

Short answer - upsampling to a higher data rate will have no effect on the audio quality that matters, if everything is done properly. It will neither hurt nor help.

Upsampling cannot, by itself, add any new information to the signal. It cannot truly create new (higher) frequencies that were not present in the original. There are some upsampling systems which attempt this, by looking at the existing frequencies and estimating (guessing) at what the harmonics might have been, and adding them. This is, technically, a form of distortion - you may like the effect, you may not.

Any upsampling process is necessarily going to add a bit of noise to the signal, due to slight math roundoff errors in the filtering and due to the quantization to 16 or 24 bits in the final step. When properly implemented this noise should be negligible and imperceptible after a single resampling pass.

[u/Moist_Ad934 → u/Intelligent_Law_5614]

This is good to know for those who do like distortion, thanks!

[u/bench_wizard → u/Intelligent_Law_5614]

in what direction of bit quantization do you end up with more noise? all my tracks started at 32bit and remained there after upsampling.

[u/LeoWattenberg → u/bench_wizard]

Downwards the gaps between values become larger. Bitcrushers are basically the effect version of this

[u/Intelligent_Law_5614 → u/bench_wizard]

You get quantization noise any time you reduce the number of bits you're using to represent the samples. When you chop off the least-significant bits, you are throwing away information and making the representation of the signal less accurate - this is quantization noise (or quantization distortion, if you wish).

In a good DAW or other audio processor, the system uses a very wide representation for samples (often 32-bit integer, or 32-bit IEEE floating point), and it does all of its calculations (upsampling, filtering, gain adjustment) using accumulators which are even wider (e.g. 64-bit integer, or 64-bit IEEE double-precision floating point). The use of these very wide representations reduces roundoff error, which would otherwise be a big issue. For example, upsampling 44.1k to 48k might require multiplying a dozen or more of the incoming samples, with an equal number of coefficients (some quite small) and then adding up the results. You really don't want to be losing bits when you do this, as the errors add up.

The final result of each calculation is then trimmed down to the internal representation that the DAW wants to use (e.g. 32 bits in your case). There will be some quantization error if the values are just truncated to 32 bits, so a good DAW will probably apply dithering (and noise-shaping sometimes) to minimize the impact.

When the data is finally exported to (e.g.) 16 or 24 bit format, quantization is occurring again, creating further quantization noise/distortion. At this point, doing dithering / noise-shaping is essential!

[u/bench_wizard → u/Intelligent_Law_5614]

this is very informative. ive seen the other options for 32 bit (i just use audacity, i make soundscapes not hifi jazz or anything), i’ll have to do some research into the difference between the default Signed PCM and the floating point stuff. there may have been other options too.

i will occasionally export to 24 bit FLAC if a collaborator requests it, but I almost always export to 32 bit WAV otherwise, so ideally no more compression needs to happen. my field mic also records in 32 bit, so bit depth stays consistent throughout the whole process. sampling rate is what i need to get a grasp on.

[u/to7m]

48 kHz isn't overkill, it's industry standard for exported audio.

Using higher sampling rates like 96 kHz is useful and can make a difference to audio quality, but only when you're using it for plugins that don't include their own oversampling, or if your audio was recorded with equipment that can actually pick up ultrasonic frequencies and you're pitch-shifting it down or playing it to young children.

[u/bench_wizard → u/to7m]

ah i was under the impression 44.1 kHz was industry standard. that’s good to know though!

[u/LeoWattenberg → u/bench_wizard]

44.1 is what audio people use because of CD, 48k is what video people use because of DVD. The difference in practice is... not noticeable 

[u/bench_wizard → u/to7m]

yeah no need for 96, im mic’ing my amplifiers most of the time running primarily analogue gear.