WHAT is the BEST SAMPLE RATE?
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- Опубліковано 29 вер 2024
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![96 kHz vs 48 kHz vs 44 kHz - What's (really) the Best Sample Rate for Audio? [2023]](/img/n.gif)
i find 0 Hz produces a really warm tone that you cant replicate with higher sample rates
NICE ANALOG WARMTH
432 is much better. It can potentially heal cancer.
@@owenathanael The sad thing is some people believe that ;)
@@iliatilev true.
#noTroughsHereGang
Food for thought!!!........... Thanks
I record crazy modular synth patches at the highest possible sample rate as the modular synth makes noises well out of the range of human hearing and when I střech the audio and pitch it down I find sounds I wouldn't get at the normal sample rate
Why would you record at 48kHz and not on 44.1kHz?
do you ever have issues converting a client's 44.1 khz project to 48 khz?
I dont know much about those numbers in the exporting box but On FL studio the 'highest' setting of it is the 512 one.
If anyone knows can pro tools do the rendering of files at 192 while the session runs at 48 or is that only reaper?
48k and 32 bit (for not destructives clipping ;)
The highest possible is the best! but then everything must be recorded in high kHz from the beginning. no point in converting 44.1kHz to 192kHz
but remember that cd is 44.1kHz
before, it was often recorded in 48kHz to convert to 44.1kHz
48kHz for me. I have thought about going 96khz, just for the sake of over engineering.
Always been a 48 kHz guy myself
high sample rate is really useful for resampling in sound design tbh, its nice having that high head room when you pitch stuff down many octaves down or really on weird distortion and other audio mangling
This is true, it's very noticeable.
Ahh, I see this is makes sense now
what about sound design? aren´t higher sample rates more interesting there? i mean, in the terms of audio-stretching, slowing things down and stuff like that. (i am talking digital games and scifi movies here). in theory, you should be able to slow audio down without hearing too much artifacts when recorded in higher sample rates, right? (and you are golden when you have enough money to buy those sanken mics that go up to 100kHz)
Yes and no. No to the fact that a band-limited nature of sampled audio it will not help you extrapolate audio because, the signal is fully defined by the samples. However, higher sample for reconstruction and noise reduction higher sample rates help because, clips and pop can easily extend to 50kHz and do build to filters to remove them you need that content to exist.
You are a person in the audio world I really respect and also the last one to say that recording at 48 kHz makes sense, I remember your old video where you said that 192 kHz was the best. To be able to rectify your ideas is praiseworthy. Cheers!
@@trevorclover read shannon
es.m.wikipedia.org/wiki/Teorema_de_muestreo_de_Nyquist-Shannon
I didn't understand a word you just said but it sounds very interesting lmao
Hahaha this was me two years ago! Keep with the hobby
It shounds very intreshting
I used to run at 48KHz, but recently moved to 96KHz since I now have a much more powerful computer.
Pretty much all plugins support it now and any that do not internally oversample will benefit from 96KHz natively due to the reduced aliasing. Wont make any difference to those that do already oversample though.
What format do you convert to for 96
@@WaterRiver777 WAV
@@insanebiscuit1 where do you upload too
@@WaterRiver777 i dont upload, i mix for others
I've worked over the years at Universal, Enterprise, many big studios and the most common preferred bit and sample rate is 48K 24Bit, so i've always worked at these rates plus the plugins function much better unless you have an HD system
Video editing programs call for 48k/24bit and so I have always used that as well for the compatibility. Set it and forget it.
Though I’m not a mix engineer. I’m a film composer and songwriter that does his own tracking, and I am working with a lot of samples that are already 48k/24bit as well. It’s pretty much the standard in my world.
No this is just solely music production
@Lostie DeMonde Absolutely, bigger numbers are always a great selling point.
That makes a lot of sense leaving 192kHz for processing, since you should then have more data points to manipulate, you could have more accurate processing. But definitely still no need for more the 48kHz when listening to a finished product. Excellent video and explanation!
Don't higher sample rates make for better time stretching?
Theoretically yes, but the sample rates that we’re talking about and the ability to over sample during that process render the benefit more or less imperceptible. It’s not like video frame rates, where we’re talking about huge and extremely obvious differences.
@@melissabell585 I see. Thanks
192k yields some unruly file sizes to catalog, really only noticed a difference with a client playing classical music on a 9’ Grand piano with tremendous dynamic range in the music. Also it greatly reduces your access to how many plug ins can be instantiated. I agree-24bit/48k is the ideal tracking paradigm.
Stock plugins are good enough, also manufacturers can always upsample their plugins
How can the dynamic range be affected by the sampling rate? The bit depth controls the dynamic range, by lowering the noise floor. And the hardware, no matter if it’s audiophile consumer hardware, a prosumer audio interface or professional gear, has yet to surpass the limits of a bit depth of 24 bits. And nobody would ever hear a noise floor of even 16 bit recording, unless the volume is at a level that will hurt and degrade your hearing capabilities quickly. So the advantages of 24 bits really only are of any interest in the field of recording and mixing. Or do I forget something here?
192k has nothing to do with the dynamic range, it’s bit depth that is responsible for dynamic range.
>really only noticed a difference
stop lying mxr
My Tascam maxes out at 48/24, so I'll vote for that... Second place would be a Maxell xll-ii running at double speed 🤷♂️ ;)
I run Studio One at 48kHz. The problem is that the rest of the world is typically 44.1kHz. When I leave Studio One and go to say, Soundcloud, I have to remember to change the Apollo to 44.1kHz. Maybe I'm just doing something stupid....it wouldn't be the first time.
44.1 forever and ever
I have had the option of high sample rates for a long time, but the final format for the film scores I worked on was always 48k, 24 bit. My decision was to track at the sample rate of the final (dub stage), so I recorded and mixed everything at 48/24. My experience is that Sample Rate Conversion does much more damage to the sound than any benefit that may be gained using high sample rates. The 3 worst words in digital audio: Sample Rate Conversion.
Your Mileage May Differ :)
Been working at 24bit 96k since 2004. I hear a significant difference switching down to 48k, and only a slight difference moving from 96k>192k. 192k may very well kill my ancient rig.
@Joe Smith In theory, you may very well be correct. If you get a chance...try it on the old RME FF800. We hear the difference over here in the studio.
@Joe Smith most times yes as we track at 24/96. So provided the source files are recorded in high quality then yes. I'm not here to debate as I'm snowed under with radio work and need to prep for tomorrows show. Each to their own
Its a simple enough test..play a high quality Ultra HD song and from RME's Fireface settings switch between the sample rates as the song is playing. You should hear it for yourself (unless I have a defective unit of course)
I'm working in the film industry for over 30 years now. There is still a Myth going around that 192k is the real deal. Which is completely nonsense. 24bit/48k is mostly used and totally enough for EVERYTHING. Even 16bit/44.1k is enough and nobody will ever notice any audible difference between 44.1k and 48k, 96k or 192k. You will of course see some differences in your RTA, but you are not able to hear it. Anyone who claims they can hear higher than 20khz is (sorry) a complete idiot. What you CAN hear is saturation from tape (real tape folks... no plugin mojo which does nothing other than cutting/boosting frequencies and adding fake "harmonics".... but there is no real saturation, it's just marketing). Take care and have a good day.
So, basically 192khz is snake oil.
Your favorite plugin manufacturer Acustica Audio recommends running their plugins @96k :)
The one big reason to use 192khz is if you want to time stretch audio. More sample points, better result.
Yes there are more sample points in general but the region,say,between 20hz-20 khz is represented with the same number of sample points in any scenario.
Silly question, if I use vst instruments that use sampled soundsets such as the Grand 3, aren't these usually recorded at 44.1 khtz? Wouldn't playing these back at 48khtz alter the pitch of those instruments?
I'm very confused by this, sorry if it's a silly question.
Nice to see you talk about this too! I made a video about this way back, and made sound examples with some cheaper converters. Some people got pretty upset and claimed that I did something wrong haha.
Would you mind directing me to that vid?
Remember guys, people don't listen to sample rates, they listen to spotify with their 15$ earphones :P
Spotify is 16bit/44.1k anyway
@@weschilton So are CDs. Streaming services are just lossy-watered down versions of 16/44.1
@@DragonboltBlastter "people"? I listen to 96/24 and 192/24 on Qobuz on Campfire IEMs & on a pretty decent hifi setup. It's really depressing that people are using "lowest denominator" arguments to justify shoddy practices The saving grace is that 99% of the people in these comments are running home studios producing music that no-one will ever hear.
@@domcoke I can't disagree that most music made by home studio owners will see the daylight. Howeber what do you mean with ''shoddy practices''?
@@DragonboltBlastter I count "shoddy practices" as those that are justified by those that think they're "too much hassle". Or that disregard quality over convenience. Or take the position, as the original commenter intimated, that if "people" are only listening to music from Spotify on shit headphones, then why bother maintaining any standards at all... for example: why not just work at even lower sample rates - maybe even mp3 quality if that's all anyone cares about...? My main point being that it's pretty depressing to see people justify the lowering of standards based on the fact that some people don't care. Enthusiasts, of all people, *should* care.
I always mix all my songs through 96khz and it immediately sounds better
I track at 48/24, master is tape and then 96/24. The quality of the converters is more important that the sample rate. Depending on their clock and design, converters can sound better at specific sample rates. Find your sweet spot.
do you ever have issues converting a client's 44.1 khz project to 48 khz?
@@johnthecreative I would keep it at 44.1 while mixing
@@Synth2000 tell that to many pros that do 48.
@@johnthecreative My 44.1 comment was a response to the former poster. I currently work using 88.2 and DSD, and for many years I was on 48/24.
Can you make a video about audio 32-bit float depth vs 24 bit?
Completely agree with the workflow. I do most basic stuff and recording in 48. I even get a basic mix and levels done this way. Once I am happy with most things I set Reaper to 96 and stem ALL the tracks to 96.
Then I do the last tone shaping and compression.
At this point it does not tax the system too much as the main fx like amp sims etc have been applied.
I have noticed that the mix and master is a lot more clean and defined as well as the instruments are more separated. I guess it just gives the daw more points to work with. I also try to make it clear that resampling higher does not add information it just ads more data points.
I have a video about this here if anyone is interested:
ua-cam.com/video/i01zLOH5j9Q/v-deo.html
I run on 88.2k, as it's the easiest for the converters to handle (to 44.1k) , unless there are 2 dedicated circuits for (doubling) 44.1 and 48k. 192k should be converted back to 48k if you want to be sure of matching bits and their calculations. Floating integer calculation at 64-bit will handle the conversion better, but I would stick to doubling/ halving myself.
It seems like it should be easier, but mathematically the conversion from 88.2 to 44.1 is just as complicated as 96. It's not a case of removing bits or samples or anything, it's still fourier transforms and curve fitting.
@@EricOehler01 as far as I have heard dedicated to multiples was the way to go. It seems logical too when dividing 2 by 3 in integers. that's where the 'curve' fitting gets slightly affected, The bits are changed by the tiniest float points or something like that. I saw it in a presentation from Marantz in their high end stuff. It def changes the sound, butm at the moment processing is way improved, so I wouldn't worry that about it that much personally. That said, I record and mix in the same samplerate. Always. If I change the settings on both my interface or DAC there will be differences in the sound. And not always for the better. Steady is the way to go imho.
@@EricOehler01 I thought the calculations are easier with 88.2 to 44.1 and vice versa because no interpolation needs to occur between samples. That vs something like 96KHz to 44.1 which would require some interpolation because of the non-integer division of the sample rate no?
Eric is right here. The “easier calculations” thing is really a myth. I always encourage people to do their own tests to see what actually works and sounds better. So I encourage you to try 96 vs 88.2 and see what you like best. At those rates it’s a slight difference and probably doesn’t matter very much though.
@@Dthebeatsmith it always requires interpolation, because downsampling isn’t strictly a divisive operation. My DSP math is weak and I haven’t done any since college a loooong time ago, but YOURE not just picking the path between samples, you’re refitting a curve to a new periodic rate. Basically saying “this equation described this curve at x sample rate, what equation does it at y”
We’re used to the notion of halving and multiplying sample rates because it makes logical sense, and works for visual mediums and frame rates. But the math is complicated and fussy.
If it were as straightforward as halving things, there wouldn’t be any differences between SRCs and we’d never have to worry about whose DAW did the best downsampling. :)
A very clever idea is that you let the one file run in 192 khz and record this file with 48 khz so there will be no processing of the rate, its more like a recording of the configuration, without changing it on sonical levels. This is one of the best solutions.
www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf I think you were referring to this document. It states that the optimal sample rate is 60kHz.
Another thought. Have you ever been in the situation that the anti-aliasing filter didn't have enough bandwidth at 44.1kHz and caused aliasing? Because when the end product should be 44.1, I'd recommend to just use this sample rate, or use multiples of it like 88.2kHz (better would probably to set project on 44.1 and use that DDMF wrapper vst for oversampling, when a plug-in introduces aliasing). It is easier to calculate that back to 44.1 without errors. In practice, both ways probably give very good results though.
If you got oversampling on the plugins where aliasing is a problem like Saturation, Distortion etc, 48 kHz is enough. You don't need any more.
6:08 it is worth putting reaper on the extreme setting for sample rate conversion. It has the least amount of aliasing according to that sample rate converter comparison website which I don't think I can link to without UA-cam dumping this comment in the spam folder.
Good video, Wytse!
Sadly, the mentioned links don't work any more - I would have loved to try the experiment.
Anyway, saying that all DACs can perfectly reproduce everything audible at 44.1kHz samplerate only tells half the story. For one: While we clearly can't hear standing waves or pitches above 18-21 kHz, we _do_ can discern differences in timing and pitch combined at least an order of magnitude more granular than what the Fourier uncertainty principle sets for linear systems like, say, microphones (phys.org/news/2013-02-human-fourier-uncertainty-principle.html). Adding to that, aside from overtones, it's transients that give instruments their timbre and they can have frquency components higher than would be audible as a sine wave that are important for their timing.
Now, you probably know that virtually all consumer devices use delta sigma DACs and you probably also know that the raw output of a D/S DAC at 44.1 or 48kHz sample rate sounds like absolute crap, which is why every D/S DAC internally oversamples the signal to a much higher sample rate before doing the D/S conversion, so that the tons of filters that need to be applied in the process even have a chance to filter out all this crap. What happens there is that a mathematical algorithm is used to add additional samples. There's many different algorithms which make different trade-offs between keeping true to phase or timing and that are more or less computationally intensive. And this is where it all comes down: Other than crazy expensive DACs that have unique approaches to upsampling and even output topologies, virtually every D/S or "normal" DAC on the market will profit from feeding it the highes sample rate it can accept - solely because it needs to do less work upsampling with the very limited computational resources it has; that is, of course, assuming the high sample rate is original material or being upsampled with in a better way than what the DAC could do by itself.
This is btw. also the argument for DSD as the D/S conversion is already done and a DAC just needs to output the raw data without having to fiddle with it before.
Does any of this matter for most consumers? Absolutely not. Who listens _that_ closely, right? But it does make an audible difference for a setup and listener able and willing to do some critical listening, too. Sorry for nitpicking but it's pretty tiresome to keep hearing that "any dac can _perfectly_ reproduce everything you can hear at 44.1kHz sample rate" because it's simply not true - if just for the fact that our hearing is way more capable than any microphone setup we use for measuring stuff and that perceptional qualities of music like sound stage, instrument separation, instrument timbre, cannot be objectively measured (yet), but can be hugely affected by "unaudible" information... Music rarely is made up of sine waves or simple interference of only a couple of waves. It's utter chaos in which every bit counts. And I wished that sound engineers started looking beyond their nyquist theorem and: a) accepted the fact that the "perfect" math requires infinite reconstruction steps, which aren't available in reality, b) that music isn't sine waves and frequency sweeps, and c) that our hearing works different from microphones and can detect and discern things that our measurement instruments cannot.
After lots of testing and listening, I'm absolutely content with recording and mixing at 44.1kHz. The only plugins I'll run with oversampling enabled are those that produce a *lot* of distortion.
Interesting... 44.1kHz is a difficult setup for me since AVB is not really stable/non existent in 44.1...
Same. Unless I’m working for picture, 44.1khz. What about the debate about whether recording at 48khz but printing your mix to 44.1khz causes audio degradation - theoretically or audibly?
@@Whiteseastudio what’s AVB?
@@JAMStudiosIE
Audio Video Bridging
(I actually just looked that up, and the first definition that came up was “already vaped bud“ LOL)
44.1 is great for music alone. 48 is what is used for film/video production (probably because it synchronizes with a 24fps ... I can be corrected on this).
I usually use 192KHz 24 bit but in some games (Cyberpunk 2077 for instance) the audio clips and you have to use 48KHz (24 bit) to prevent that.
But i'm no producer or musician, only an enthousiast.
The Resampling feature at 6:00 is basically how accurate you want sample stretching to be. When you stretch a sample to be a different length than the original, you might run into these really digitized, computery artifacts (especially if you make the sample slower). The higher the resampling the better the stretching will be, and the less audible the artifacts.
That computery artefact is so nice of crunch to samples. Thats the whole reason i now use arturias emu sampler. 20 years back used the emu 1212 chipset for that.
Stretching is a concept not quite applicable to samples. They are defined as scaled dirac impulses which have a infinitesimally small width and cover a area of exactly 1. The option in reaper fives away all the information: 512pt sinc interpolation. You use a sinc function sin(x)/x where x is chosen according to the desired low pass frequency. A perfect sinc goes from -infinity to infinity though. You approximate it with 512 points here and call it good enough. The 512 points (if one side is meant) lead to 20*log10(1/512) = -96 dB which is the theoretical limit of 16 bit PCM. For each resulting sample you need to do 512*2+1 evaluations of the sine and one division though, both quite expensive operations on CPUs. The sum over all is cheap at the end. The amazing part about this technique is that you can choose any resulting samplerate you want. Fancy some 37kHz ? No problem.
TLDR; samples can't really be stretched, sinc interpolation is great but computationally expensive.
@@Mefistofy I meant an audio file, not individual samples, sorry if it wasn't clear. Stretching something like a guitar to last longer, or be shorter. They are indeed pretty expensive on CPU, and that's why one should save higher point counts for offline rendering.
@@iammodus Did not think of the sample as sound recordings. I might bee a little to much into DSP sometimes. Everything you mentioned is exactly on point.
Higher sample rates on conversion are useful for creative sampling - if you plan to radically slow down or pitch down your samples, its a good idea to convert at higher rates, so you do not lose all the high frequencies.
Actually it does not work like that. If you tune down your sample, your low frequencies get pushed to high frequencies area, so your 100 hz becomes 200 hz, etc. So having frequencies above 20000 hz won't help - they will be translated to the inaudible frequency range anyway.
@@kadavr0s I think you got it backwards. Maybe I've explained it poorly.
@@kadavr0sNah fam, there's definatley a huge difference in resolution if you're slowing down or speeding up high sample rate files. It's almost like recording at a high frame rate with a camera.
I’m confused. I work in logic, and if I record at 48 and then change my project settings to96 or 192 after the fact it changes the speed of the playback.
Plus, I’m not really understanding how you can process at 192 when you recorded at 48. Where is the computer getting the extra data from if it wasn’t recorded in?
Do you remember I was talking about this, that some people were hearing the intermodulation distortion from the convertors, as an improvement in the treble response (One day you might get a plugin to give it that vintage I.D. sound, though I hope not!)
sound recordist use high sample rates with mics that have extended frequency response, there is huge difference when you design sounds from these recordings then, but for anything else i don't think there is a reason to go over 48khz at 24bit.
this sounds useful for recordings that are intended to be slowed down later, as this would bring supersonic frequencies down into the audible range
I write electronic music at 96khz because I can get lower latency on soft synths, at the cost of more CPU. For mixing purposes 48khz with plug-in oversampling like you said it's the way to go.
I have been running projects in 88.2 with my 80s analog console...
I was told that if you plan to sum or do external processing that it’s better to leave digital domain at highest rate possible.
Any truth to that??
96k/24bit is what I use and I like it. I record into my desktop because my pos laptop cant keep up like you said.
Snake oil?
Only time I find use for higher samplerates (96kHz), is when I do field recording-I do walk around with mic and headphones and capture interesting sounds, I can later transform in studio. Samples recorded at higer samplerates respond, sound better, when time stretched (usualy slowed down). And later do bounce them to my working asmplerate-48kHz, that I also use for recording bands.
Would be interesting to record some music at 192Khz and then spectral shit the top part into the audible range (without folding like in aliasing), would be the equivalent of a gamma ray picture.
48Khz scientifically proven.Anything above that is overkill.
Do you even foldback distortion bro?
Article link is redirecting and takes us to some chinese website...not sure if that was what you were going for.
The link is not working (japanese or chinese web site)
OK, holy crap.
I just took an old mixing session I did at 44.1kHz (because all those audio files are 44.1kHz) and just increased the project sample rate (in Reaper) to 192kHz, and holy crap, the dynamics improved like crazy...!
I did nothing else, just increased the project sample rate to max.
So, I did some quick testing.
1. I rendered a fresh 44.1kHz version, with the project at 44.1kHz. (Just to make sure it's this exact mix I'm listening to.)
2. Then I rendered a 192kHz version, simply setting the project to 192kHz.
3. Then I loaded this 192kHz render and put a brick-wall lowpass at around 20kHz (analyzer showed some action above that, so I wanted to manually remove it), and rendered it back to 44.1kHz.
Versions 2 and 3 sound more or less the same, with very much improved dynamics, especially the snare. Simply allowing it to process in 192kHz, even after later downsampling back to 44.1kHz (native resolution of the audio files), made a huge, huge difference.
Obviously, the true 192kHz version has a tiny bit better quality in reverbs and such, but that's not the point, here. The point is that the dynamics improved so much that the mix would need some adjustments to adapt to this change -- it's not a subtle difference at all!
The snare now gives me a headache, as it seems to shoot out from the mix and punch my forehead...
The upper range of the distorted guitars is also more defined in versions 2 and 3, though this is more subtle.
(All renders were still native 24bit, and I used lossless FLAC format. Only the sample rates were changed as stated above.)
imposible ,si una grabacion es a 44.1 siempre sera de esa frecuencia, si lo procesas por un hardware y lo grabas a 192 es otra cosa ,si grabas una guitarra a 192 por ejemplo o 96 etc , si se notara ,aunque yo no noto nada ,ni en numeros ni en oido ,uso analogico , si tienes algo en 44.1 siempre lo sera aunque lo subas ,saludos
@@rafaelgarciamoreno8089 It's about oversampling during processing.
@@antiHUMANDesigns imposible """No hice nada más, simplemente aumenté la frecuencia de muestreo del proyecto al máximo"" eso dices ,si lo procesas con pluginsde 192khz que apenas hay puede que mejore ,te habra pasado otra cosa ,uso ssl the bus+ y fusion hardware ,grabando a 192 y 44.1 el master ,no hay diferencia audible ni medible ,una eq solo llega hasta 20khz y el oido igual ,a no ser que seas un murcielago no notaras nada, seguramente ha cambiado el pannig law a triangular ,entonces sube el volumen ,la dinamica ,es la forma en que se trata el estereo, la gente lo confunde con calidad muchas veces ,saludos
@@rafaelgarciamoreno8089 The point is that I must have had some plugins on that were causing aliasing, for example, which need to be oversampled to remove the aliasing.
I don't even remember, this was an old comment.
@@antiHUMANDesigns ok entendido ahora ,saludos
I preffer to use plugins with oversampling on 44.1/48 not only because my CPU doesn't need to struggle as much, but also because of compatibility.
Also 24/48 in the mastering room and production room here.
I've been recording and mixing at 96k with an I9 20 core 48g Ram setup using reaper as my DAW. I'm currently using Presonus CLARETT for my interface.
I find that large track and plugin counts are giving me lagging issues.
I've streamlined the buffering settings in reaper for my rig but still having issues from time to time.... You mentioned a testing website to see if my converters could handle this... But the only links that work for articles on your page send me to a Japanese website...
Any suggestions?
Thx, Jeff
At the end of the day, its somewhat system dependent - recording 96k sounds a bit better than 48k on my system, but I do not know if that is plug in related and I've not been able to tell the difference on other systems.
Why .... did it sound so much better importing the 192Khz files into the daw? Define 'better'? Could it be that the conversion from 192Khz to 48Khz suffered from 'bad settings' ? And how did it sound once the final project got exported in the 44.1Khz or 48Khz master. Also better? Or did 192Khz treat your ears in such a way neither 44.1 or 48Khz can ever do? Hmm. Questions.
.....i'm a rebel......if someone tells me i can't do something i'll DOIT!
384kHz Sample Rate with a Bit Rate of 64 bit FLOAT!!! for my vocals LETS GOOOOO!!!!!!!
Personally, I record at 96kHz/24bit. Both to be able to release a "geek" version (96/24 FLAC) and to be more "future-proof", I suppose.
I don't see a reason to go all the way up to 192kHz.
After testing various sampling rates....96khz is the best sounding imo,that extra headroom is important as yes we only hear up to 20khz, more 16-18khz, but need the headroom,as i keep telling people if super tweeters are outside the human hearing range why do people use them ? but yet they make a massive audable difference to the top end clarity & definition, & can even make the bass sound better somehow...some can hear the difference but others somehow cannot...96khz is perfect imo & should be an industry standard, used with 32 bit..if you cannot hear the difference between 24 bit & 32 bit your in the wrong job...it is the future & works far superior if used at 96khz min, with 32 bit you could go higher than 96khz...a massive sonic difference...any sampling from 48khz up will work fine, but 96khz for me, but only really at it's best if recorded at 24 bit (32 would be far better & 96khz & kept at that throughout the whole recording & mixing process...upsampling & downsampling are best avoided for the best audio quality...everone is going on about 32 bit float...i agree 32 bit is far superior in audio quality than 24 bit & makes 16 bit sound terrible...lol..32 bit is coming so why not settle on the best sound quality & use higher sampling rates...i cannot hear much difference between 96khz & 195khz at 24 bit as most music is recorded at 48khz then upsampled....but recording at 24 bit (32 bit would be far better) & 96khz & mixed without up or downsampling sounds really good to my ears...all our ears are different & so many variables to take into consideration...32 bit 96khz recorded music sounds amazing...
I went to 48k from 96k about a year ago for your reasons and also to save CPU overhead and drive space. It's been great!!!
Have you tried 96khz ?...it seems quit popular in recording & should give a better sound..bit rate is more important...16 bit is old school & not very good...22 bit, 24 bit & 32 bit are far superior....
Ok
i just put 256 buffer and 192k in hertz in my pc external card and i feel its better in quality in medium and high frequencies.Like it has more information and detail.
REAPER's Resample Mode - it has to do with the amount of samples the DAW uses to re-create a new waveform. I'd like to try to explain. Please correct me, if I'm wrong.
Imagine a file being played at 0.5x speed without pitching it up 2x. So just "slow it down" in the most literal sense.
In the DAW the file's samples are now twice as far apart. But the DAW still needs to play at YOUR OUTPUT's sample rate. So it has to re-sample the audio and create a new waveform on the fly. It has to create samples at your output's sample rate inbetween the original file's samples.
How does it know how to set the amplitude, the height of the sample?
The easiest method is "linear interpolation". The DAW just draws a line from the two surrounding samples and creates a sample at the line's amplitude in the middle of the line.
Sounds fine on paper but in reality it sounds bad, because the waveform between the two samples might contain more complex frequency content, which will be reduced to "whatever is in the middle". It's a bit like aliasing. Lower frequencies get introduced.
So anyway, any higher quality resample mode goes to where it needs to re-create a sample, looks at X samples AROUND that spot and THEN recreates a waveform that better matches the original.
The more points you look at, the higher the CPU cost per new sample.
That being said, I also can not hear any difference between "good" in REAPER's resample mode or anything higher.
However, I had to deal with this problem above in game engines, because in order to run fast, they often use linear interpolation. Imagine a car engine going up in pitch as it's going faster, this is often done by dynamically pitching the file - resampling it. In my example my high frequencies of a bat squeek were lower and lower, weird sounding ones were added, after I put the file from the DAW into the game. So yeah. Cool to know audio nerd stuff. :)
WHA.... We have to read stuff??? I come to UA-cam to avoid reading :)
I've been running my AD/DA at 48kHz for 2 years, after years of 96 and experimentation with 192. My Lexicon reverb's SPDIF only runs at 48kHz, which dictated the sample rate I used for several projects. I found that I honestly couldn't tell a difference. I quit worrying and have not changed since.
With a few soft-synths (NI Reaktor) and distortion plugins (as you've discussed many times) I can hear a difference in very contrived circumstances. But I don't think these would be audible in an actual track.
if you have clients do you ever have issues converting a client's 44.1 khz project to 48 khz?
well in single tracks it’s different, depends on the mix, when i submit the final mix to the record labels i usually go 96khz 1411 kbps 24 bit, if requested 32 bit. the average music consumer doesn’t give a honk on immaculate audio quality, they care about their phones and storage, thus preferring aac or mp3 to lossless formats, live flac, alac or wav, mainly because the highest quality they gonna have is spotify most times, which tops out at 256 kbps. i did an experiment, with several people, making listen various file formats of the same song, through the same equipment, and to the untrained ear, it all sounds the same, slightly more detailed when playing lossless, with high bitrates.. plus bluetooth, is now the main portable headphone output, which tops out at max 320 kbps (LDAC excluded due to sony/lg proprietary software) through aptixhd, so yea
@@datutturugang666 thanks for so much info. I heard a pro say he prefers the sound of 48 to 96 because 96 actually has TOO MUCH headroom to respond to what he is trying to achieve. I have only ever worked in projects in 44 and I need to start up-sampling these to 48 for compatibility with gear I just bought. I wondered if anyone has hurdles or glitches in this conversion process, like plugins and samplers that don't do this reliably, etc. I use logic and I am crossing my fingers hoping it works okay. another issue with 96 is many plugins don't work in it, like older waves plugins.
@@johnthecreative i also use logic mainly, like 80%of the time, i usually have no issue in opening different bitrate files. it’s easier to open 48khz files in a 96khz project, because its exactly double, it gonna sound like a 48 file anyways, but i mean, who aside pros or audio enthusiasts even gonna care about the difference between a 44.1 vs a 48 file, if they enjoy a song they’ll listen the shit out of it even if it’s downloaded on a shitty mp3 downloaded from a sketchy website.. it’s just us visibly mentally unstable people who prefer comically large file sizes for a 5 minutes song lmaoao
@@datutturugang666 well thanks for the info. Like I said I only know about 44 but now i Have to learn about others in order to use new gear requirements of 48. Crossing my fingers that I can easily switch sample rate of on an entire project in Logic Pro.
playing my guitar thru a amp vst with higher sample rate definitely sounds better tho. like you dont even need trained ears to hear that, its day and night
I agree, the ADSR when playing sounds more realistic when playing, especially the attack
Never used Reaper and I don’t understand how this is working. How can Reaper work with different sample rate files in different project-sample-rate without down/up converting files?
When you change the project sample rate does Reaper automatically convert files in background or something?
I use Cubase and as far as I know it’s not possible to do this in Cubase without converting files or files will be played back slowly/fast in incorrect pitch.
Does anybody know how to replicate what he did in Cubase?
I don’t understand. You started by saying that 48khz or 44.1khz is fine, but at the end you said 192khz for processing sounded better for your client’s stems. Can someone explain?
Even a cheap mixer like the Behringer X32 works internally at 96kHz whilst externally at 48kHZ meaning the plugins do not alias in the audio range, they do it for a specific and important reason !! Quite a few UA-camrs have shown scientifically through spectrum analysis why 96kHZ is important as against working at 48khz. That is why some of the better plugins offer over sampling its notr a gimmick its for an important reason, anyone who does not understand why should not be mixing professionally and should go back to school !!
Everything is going 48kHz. Why? Dolby Atmos. The major labels in the US are asking for Atmos for all their big artists. Demand is increasing.
link does not work. anyway i'm trying to run my system at 96khz and it runs ok.
48khz allows for me to use the highest buffersize on my system. And im not able to hear aliasing from the synths like i can at 44.1khz. and if i need higher samplerate somewhere i just use a plugin with oversampling. If i was running a tracking studio i'd probably run 96khz.
someone told me that he uses 48kHz only for low frequencies to 300Hz and 192kHz for everything above, because he said, that 48kHz is having a bigger compression on the frequencies than 192kHz. He also stated, that using 48kHz on everything over 300Hz causes a lack of audio information. Dunno if he is right, wasn't able to check it myself to prove that myth and I think, I'll watch that video to the end and than decide myself.
I have never gone over 48Khz while recording for this same reason + takes less HD space. Many pro producers telling "record highest possible" without actually understanding what they are promoting.
Ironically as a deaf person this is fascinating
Hi.
With all respect, the information in your video is not very accurate.
First of all, the fact that you can´t hear something, is not a assuring that there is not information that affects you. I recommend you research Hiroshi Ohashi papers regarding to how our body deals with frequencies upper than the human hearing range. Also there is a great story With Rupert Neve and Geoff Emerik with a console's channel strip rising amplitude in 54 kHz (Rupert Neve, Massenburg and many others produce equipment flat up to 100 kHz).
Another thing is that you are using Reaper’s sample rate converter, that adds alias frequencies with the conversion (a lot more than other programs as iZotope RX), and that is affecting the sound. I recommend you to check the website src.infinitewave.ca that has tons of DAW sample rate conversion comparisons including information such as alias, phase, impulse response, etc.
ok is a old video but work at 96khz sample rate is best option to avoid aliasing introduced from saturation plug in (especially) that in a full mix in the end will make it sounds so bad in hi frequencies
I m sorry but I ‘m very confused about this video.
First, you explain 48 is better then you end up with the story of your client with a much better sound at 192 ?
Can you, please, resume because I’m a French guy and maybe you talk too fast for me !
Thx for your valuable work by the way
Oversample does matter in any plugin, thats how we can hear that some daws or plugins sound "better", its oversampling (aside aliasing)
Being bedroom DJ i used 44-48kHz 24bit, so my MP3 collection ez to mix live music👍
I use 3 audio interfaces so I don't have to switch between 44.1/48 or greater sample rates. I feel 24bit/48khz is perfect for videos and most applications. When I hire a client for professional vocal audio-work, I'm perfectly happy with receiving studio-grade 24bit/48khz audio files.
hello.please tell me what kind of CPU is needed approximately to work in 192? please advise.
Hey Wytse. thanks for some always great content :) The alternative link you provided for the article seems to lead to a chinese unsecure website. Just so you know...
I had once 96 KHz mixing experience. Everything sounds more natural close to analog domain. Especially I liked how kick and bass sounds in higher discrete rate. All I can say it sounds better actually, but really cpu hungry.
It is obvious that the author of this video either does not under stand or cannot accurately explain what digitisation and or analog to digital/digital to analog conversion is and how it is done. Just as important to the sample rate (the A to D conversion is the capacitive circuitry the does the D to A conversion/recovery.
Is there really much of a difference between 44.1 and 48?
i can mix longer when on 44.1.... sometimes i switch to 48 but i keep getting back to 44.
the links to the article are broken :-(
Much of the music we listen to has poor sound quality that comes from poor mixing and eq adjustment in the studio! So then it does not matter if you use 44.1khz 48khz or 198khz
Domagoj Vida changed profession, from football to producer 😎 P.S. Tnx for video 🙂
If you are working with something like pitch bending sine wave, 192 vs 48khz show massive differences. Since you're no longer limited to what your microphone can record.
By far the most insightfull video I have ever watched upon the topic is the one on the FabFilter UA-cam channel "Samplerates: the higher the better, right?". Very interesting video, very much een aanrader!
4:50 By the way: That "Sandstorm" song is awesome! Love it! Props to Darude!
Dude that’s not Sandstorm by Darude. That’s actually Darude, the song is Sandstorm.
FINALLY! 44 or 48 for recording. Oversampling for mixing.