96kHz - I don`t hear a difference to 48kHz in the audio quality. But with some vst effects, I (think) can hear the fold back aliasing if I go below 96kHz.

 

i know, only during mixing/mastering it stays in the 48/24 format, the final product is always 44khz/16bit wave/mp3

 

Personally, I found that I really can't hear much difference in working with rates higher than 44.1, EXCEPT to say that some plugins seem to sound very different at different rates, and in those cases, I seem to find that I prefer 44.1, which seems to make sense as it is a pretty standard rate.

 

I tried to change the rate in my DAW (FL Studio) and went back to work on some other material. Some of the instruments were completely out of tune until I changed it back. Anybody have an answer for this?

 

Not that strange.

Latency is calculated in samples, not ms, that's the 'trick'.

Let's say your song has a latency of 44100 samples. At 44,1kHz sampling rate this results in a whole second. But at 88,2kHz sampling rate this latency only needs 0,5sec, that's all.

Of course this also has a drawback, higher CPU consumption.

 

Since several members mentioned bit resolution: 16bit have dynamic range of 96dB. If you need more, which you most probably don't, switch to 24bit (remember, the old hardware samplers and some synths had 12bit resolution and no one complained about it).
32bit FP (floating point) has in fact a 24bit resolution with the advantage that digital clipping doesn't occur (it moves with the level). Means you can push your tracks above 0dB FS as long as the inserted plugins work with 32bit FP too (same goes for 64bit FP, ofc).

The thing with the sampling rate is this: only freqs up to the half of the SR can be produced, higher freqs are thrown/mirrored/reflected back into the spectrum below. This is called aliasing.
So, for any freq below SR/2 this doesn't matter at all, for higher freqs it could matter.
The problem is not only the artificially raised levels of those freqs, it could also lead to audible phase issues.
Whereas this is very well measurable, it's a question of source, monitors/HP and ears if you can really hear it.

 

Bit depth has much greater impact on recorded sound quality than sample rate.
For instance 24 bit against 16 bit has 28 [256x] more possible amplitude positions per sample. 44.1 kHz against 88.2 kHz has just 21 [2x] more samples per second [and the resulting file is necessarily also twice as big]. So sound quality difference between 16/44.1 and 24/44.1 is much greater than between 16/44.1 and 16/88.2.
But many people don't care even for the difference between 24 bit and 16 bit, because audibly it is just slightly greater than the difference between a mp3@320 and a 16/44.1 WAV. It is greatly appreciated among Hi-Fi enthusiasts though, and clearly audible on expensive [3.000+$] Hi-Fi gear.

 

Last edited: Dec 2, 2019

Well, that can lead to misunderstandings. 24bit does not have a higher resolution within the upper 96dB, it just adds 48dB of dynamic range below the 96dB.

 

http://productionadvice.co.uk/high-sample-rates-make-your-music-sound-worse/

 

The gist of this is that it is the final audio standard that matters for auditioning [where you meet the analog components]. But for processing purposes higher bit depths and sample rates [divisible by 44.1, for reasons I mentioned in my earlier post] are better.
I personally, for the last three years, always end up with 24/44.1, which is, in fact, in accordance with what is recommended in this article.

 

If you work at 96, all recorded stuff should be recorded as 96k, samples which are 44.1k should resampled to 96k with good SRC or it will be autoconverted by your DAW (DAWs usually have not so good converters). Saracon, r8brain 2 pro, dBpoweramp, sox, ableton live latest versions, izotope rx's SRC. Always sharp filter at 22k if samples are 44.1, linear phase. Samples before src should be quiter than 0dB, otherwise clipping afte src may appear.
32 bit float or 64 bit float DAW engine.
Downsample to 44.1 before Limiter. After limiter dither to 16 bit for cd or 24 bit for hires. No dither for mp3. Dithering: ozone or rx are fine.

Some short correct info.

 

Make 8Khz great again

 

48kHz, 96kHz and 192kHz make no sense. These bitrates are made movies or games because they fit into the fps range of a movie.
If you render down from 96kHz to 44,1kHz you will probably lose quality, since it's hard to split the bits. You better use 88,2kHz, so the bits are divided by 2.

 

Hey, I love my Mirage ;)

 

It doesn't correlate to fps. Where do you get this info? These sample rates are also for work, reduce aliasing, headroom for frequency etc.
1. Some converters convert too bad from 88.2 to 44.1, a lot of aliasing, IMD, noose etc.
And vice versa, a lot of converters does the best from 96 to 44.1 without any noise, IMD, aliasing. Check 96to44.1 tests at src.infinitewave.ca/
2. Some simple maths and you can divide 96 to 44.1)
And i dont understand, about your bits, bitrates... We speak about sample rate here)

 

Fun fact about Ableton. Watch for a few minutes for the demo.
When sound designing I work at the highest sample-rate possible. When it's pure audio, as in, samples only, I set the sample buffer as high as possible.

 

Last edited: Dec 2, 2019

No, it doesn't. The only difference bit depth makes is the signal to noise ratio - with 16 bits it's already 96dB, and with 24 bits it gets to 144dB. It doesn't matter all that once you realize most music has way less dynamic range than that - and it's deliberately reduced throughout mixing and mastering. Even orchestral recordings don't have dynamic ranges that extreme, since you'd have to turn the volume up during quiet passages and then blow your ears once the music gets louder. 96dB is equivalent to the difference between a whisper and a jet taking off.

Also, by comparing the noise floor you could say the most high-end reel-to-reel tape machines with fancy noise reduction trickery could barely hit 13 bits.

See https://wiki.xiph.org/Videos/Digital_Show_and_Tell for more information.

^ this

Pretty much every DAW operates on 32-bit floating point internally, this is nothing new - everybody has been doing this for like 10 years already

 

This Fun Fact has every DAW which has 32/64 bit float engine, export etc. Float preserves info above 0dB.

 

I'm kind of an old guy. Why you gotta spoil my fun?

 

Well this quote:
"The actual reason for the 48KHz multiple has to do with optimal synchronizing to video. So it makes sense to have sound tracks from movies recorded in a 48KHz multiple, such as the 24-bit 96KHz format embedded into 7.1 channel audio on DVDs and Blu-Rays. But since over 90% of all music recordings are sold in a 44.1KHz for Red Book CD or DSD64 SACD it is rather ridiculous to offer any HD music in 96KHz or 192KHz as opposed to the optimal 88.2KHz and 176.4KHz HD formats. But because ignorant consumers demand 192Khz falsely believing it is better than 176.4KHz, that is what record companies market."
stems from this source:
https://www.mojo-audio.com/blog/dsd-vs-pcm-myth-vs-truth/