#24

Yes, however in most cases the computer works harder servicing the higher sample rate than it does if you use smaller buffers at a lower rate to get the same latency figure. For most general purpose audio computing systems you get the best performance by choosing the smallest practical buffer size at the lowest sample rate. Increasing the sample rate from there usually requires increasing the buffer size as well. Either way, since the system has to compute more data overall, higher sample rates usually push it over the edge of instability.

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#25

Makes sense, thanks for the calrification

#26

FWIW 24bit 48kHz has always been the standard for film/TV post production for a very long time - whether for sound design or score/music… Accordingly the default for high rez recording is multiples of ie 96kHz and 192kHz, so that half and quarter speed playback are easily implemented…

#27

in short yes. 48kHz has higher Nyquist Frequency, so it allows more complex signals in the higher parts of the spectrum. It sounds all the same to my ears but in a no-oversampling DSP chain that tends to accumulate “errors”, at 44.1 there are more inaccuracies at the start and at every stage.

#28

Everything upsampled to 96k for mastering here, if not already there. For my own recording projects, everything at 96k too, from start to finish. Easy to do a final SRC down and dither at the end for any required delivery formats.

I did lots of listening tests and my Crookwood converters seemed to like 96k the most. You’d have to do your own tests on your own gear to find what sounds best, but it seems to be true that some converters sound better at some rates (and not necessarily higher ones).

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#29

Thanks a lot everybody for your opinions and experiences regarding this!

@mzero thanks a lot for those tables! These are really useful, since I do like to work with single cycle waveforms and create my own!

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#30

maybe you are interested in this AES paper:
Pras & Guastavino - Sampling rate discrimination: 44.1 kHz vs. 88.2 kHz [pdf] (1.4 MB)

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#31

Thanks for posting, this was great and confirms some of my own findings.

#32

Also, using higher sampling rates reduces the quantization noise in the audible frequency range. See: http://electronotes.netfirms.com/AN345.PDF

#33

My naive assumption about 48kHz was always that it related to 24fps cinema film.

This Wikipedia page has a nice table of sample rates and where they came from:

48,000 Hz
The standard audio sampling rate used by professional digital video equipment such as tape recorders, video servers, vision mixers and so on. This rate was chosen because it could reconstruct frequencies up to 22 kHz and work with 29.97 frames per second NTSC video – as well as 25 frame/s, 30 frame/s and 24 frame/s systems. With 29.97 frame/s systems it is necessary to handle 1601.6 audio samples per frame delivering an integer number of audio samples only every fifth video frame.[9] Also used for sound with consumer video formats like DV, digital TV, DVD, and films. The professional Serial Digital Interface (SDI) and High-definition Serial Digital Interface (HD-SDI) used to connect broadcast television equipment together uses this audio sampling frequency. Most professional audio gear uses 48 kHz sampling, including mixing consoles, and digital recording devices.

If you do the calculation neither of the look that good… but maybe if you go back to the actually scanline calculation for NTSC it looks a bit better. :man_shrugging:t5:

> 44100 / 29.97

  44100 / 29.97 = approx. 1471.4715

> 48000 / 29.97

  48000 / 29.97 = approx. 1601.6016
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#34

Don’t believe anyone else answered this yet - as others mentioned 48k is the default for working with the film. The number was chosen because a) it is high enough to avoid the Nyquist frequency issues as previously discussed and b) it’s because film is typically shot at 24 frames per second, so a number was desired that was a multiple of 24. In this case there are 2000 samples per frame of audio.

Edit: @sam beat me too it by a few seconds (also my name is Sam too…)

One other difference between working at 44.1/48k vs higher SRs like 96k or 192k - if you are doing operations in the frequency domain (timestretching, pitchshifting etc - stuff that depends on FFT), having finer resolution data is beneficial.

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#35

I wonder how much the downsampling difference mentioned in the paper depends on the actual algorythm. Does anybody have any data on that?

Any my takeaway from this paper is that it’s better to work at 88.2 or 96Khz if you end up doing something like vinyl (assuming they don’t downsample your audio to make the vinyl) and your music benefits from the extra resolution. If you are making CDs just use 44.1K and if you’re doing a digital only release you could also do everything at 48K, but if the benefits of 96K are marginal over 48K I don’t think it’s worth it.

#36

Unless the focus is infra/ultrasound (where you might actually be a bit surprised what you get since it’s stuff you couldn’t hear on the source recording), my understanding was that bit depth would matter more than sample rate in restituting a somewhat accurate image for timestretching / pitch operations, and that the timestretch/pitch algorhythm is the real factor more than khz.

The only thing that’s clear to me from all these talks is 24 bits all the time. The rest seems like really secondary consideration unless doing some very extreme treatment at the limit of the audible frequencies where aliazing might occure.

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#37

It totally depends upon the quality of the SRC, they mention it at the end of the paper. They used the same one for everything, but say more tests need to be done with different SRCs. I use FinalCD and RX7 SRC and am very happy with both.

#38

This paper is interesting but utterly inconclusive about anything really dealing with sample theory. It’s interesting to note that in their most statistically significant result the result was that nearly everybody got it wrong… also the listeners were all highly experienced, trained sound professionals who rated the difficulty as a 9 / 10 collectively and who stated that they often doubted their own perceptions. The results which are solidly grouped around 50%, corroborate this.

I don’t believe the data supports the authors’ conclusions definitively, it hints that there might be something there, but that’s all. It also is very important to point out that this does not cover audio that’s been UPsampled to 88.2kHz, which I think would have been an essential cross-test as the output conversion of their hardware may simply be better at 88.2kHz.

It’s also very important to note that frequency domain operations will get the same benefits in terms of frequency resolution using upsampling alone - you do not need the source to be sampled at the higher resolution at all, if you’re concerned with frequencies in the audio band only.

In fact, in some other studies (I don’t have references handy, I studied this back in uni) the addition of ultrasonic frequencies from higher sample rates actually made the recorded audio perceptibly worse because they affected the amplification and reproduction equipment unfavourably (stealing energy in the high frequencies from audio content, in some cases). Higher sample frequencies also suffer more unfavourably from clock jitter, which means that, again, they can end up sounding worse than music recorded at a lower sample rate (and clock jitter damage is irrecoverable).

Again, the evidence strongly suggests that the only benefit to running at 88.2kHz and above is if your particular equipment for some reason performs better at that setting (e.g it has poor antialiasing filters at lower sample rates) SoundOnSound discusses this in some detail (mentioning both upsampling and poor antialiasing filtration), or if you need content in the ultrasonic region (e.g. for restoration work where the ultrasonics can be useful frequencies for identifying and filtering out unwanted noise in the audio band). This can be a good reason to run at these rates, but it’s important to note that even in the best case situation only pros seem to be able to tell the difference, and even then it’s quite difficult and requires special listening conditions (and even then they seem to get it wrong more strongly than right!). So if you want to spend the CPU power and storage processing high bandwidth audio, go for it, just realize what you’re (likely not) getting in return.

This is definitely correct. 16 bits is fine for final master output (with proper dithering), but up to then the extra bits make a vastly more significant difference in the final quality than sample rate.

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#39

That’s why I get the extra little oomph by using 25 bits.

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#40

Also key point - many plugins sound better at higher sample rates. It is not subtle.

That and hardware inserts I notice the difference more easily, rather than recorded sound with the same quality mic set up. The last I’m not sure I could tell difference, but there’s no reason not to use 96khz if possible for important projects when you know you’re going to be doing some processing.

I’d say 48khz is the minimum.

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#41

Wow, I’m not THAT professionnal.

#42

When I edited the spoken word audio of Alan Moore playing the part of FUCKUP (a talking computer) in the Robert Anton Wilson - Cosmic Trigger play a few years back, all the audio was delivered at 23 bits. :wink:

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#43

!!! (apologies for the derail) how was it? (reading high weirdness (erik davis) now)

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