# Why is 44.1kHz still the standard for music ?



## Fredeke

I really hesitated about which forum to post this question in... I considered Mixing and Mastering, and then I even considered Drama Zone since I expect the discussion to be controversial. You know, with all the double frequency debate... However my question is not so much about why not use 96KHz, as it is about why not use 48KHz ?

I seem to understand that one reason you might want to increase the sampling frenquency (besides conveying ultrasounds, if you're into that), is that the converter's inner digital low-pass filter, designed to avoid nyquist-related artefacts, can work with a gentler slope, hence do less collateral damage within the audible range.

To me, 48KHz seems to offer the best potential quality gain over 44.1KHz for the least increase in filesize and cpu load.

Moreover, it is a standard in the film industry, along with its close relative 96KHz. Isn't sample rate conversion supposed to sound better when converting between whole multiples ? (Does anybody seriously use 88.2 KHz ?)

44.1 kHz is the audio CD's frequency. But who still makes (or buys) CDs ? Not me.

I hope not to start a flame war. I would just like to hear arguments in favour of 44.1 kHz, other than "because everyone else uses it". (Or arguments to why doing like everyone else is important.)


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## Divico

Imo thats mostly a tradition thing. CDs being 44.1kHZ/16bit. Keep in mind though that music being released in this format was certainly not produced this way. Reducing bitrate and sample rate is common as a last step.


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## storyteller

I think (as @Divico stated) that 44.1k/16 bit is probably a byproduct of CDs. They likely wouldn't have been able to put a whole album on disk at 24/48... kind of like DVDs and Blu-rays were 8bit color rather than 10bit due to the data size. We can all agree that the newer 4k/10bit standard is much better with more vivid color... and there is room to grow from there.

The 24/48 standard is based on math calculations that are relative to our existence. In essence, a hertz is a "cycle per second" which is relative to time, the rotation of the earth, the sun, etc. Without going too philosophical here, there is more than just "it sounds better so let's make it a standard." The same goes for film at 24fps. It sounds _best_ and looks _best _(and multiples of 12/24/48 thereof) because they are calculations in *synchronicity and relative ratio* to our existence.

If you research "hertz" and learn what it actually is by definition and then research why time is divided into multiples of 12, that will be a good starting place.


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## Divico

storyteller said:


> for film at 24fps


Thats something irritating me :D For my taste it could be wayhigher


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## robgb

Because it's enough.


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## Polkasound

Fredeke said:


> But who still makes (or buys) CDs ?



99% of polka music consumers still buy CDs, so it makes sense to produce a polka CD at 44.1k from start to finish... unless the music is going to be digitally distributed as well.

I record at 48k/24bit because if you upload 44.1k/16bit files to sites like SoundCloud, the compression algorithms they use can result in audio that's torturous to the ears. On the other hand, any sonic losses incurred by the sample rate conversion from 48k/24bit to 44.1k/16bit are too insignificant to matter to CD consumers.


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## storyteller

Divico said:


> Thats something irritating me :D For my taste it could be wayhigher


It is an interesting concept that directors keep toying with. Cameron might be doing the Avatar sequels in 48fps, but that wasn't very well received with Peter Jackson's Hobbit experiment. 48 might work well for Cameron's 3D goals though. We will see.

What fps do you like? 30/60 has always seemed a bit off to me. I'm always curious to learn about other's tastes. 

Regarding audio, I find that if I am working at 16/44.1k I feel the same "off" feeling. Not as much with 24/44.1k, but 24/48k "feels" the best. Can't really explain it in words though. Sonically, without a doubt 24/48k sounds superior.


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## Divico

storyteller said:


> It is an interesting concept that directors keep toying with. Cameron might be doing the Avatar sequels in 48fps, but that wasn't very well received with Peter Jackson's Hobbit experiment. 48 might work well for Cameron's 3D goals though. We will see.
> 
> What fps do you like? 30/60 has always seemed a bit off to me. I'm always curious to learn about other's tastes.
> 
> Regarding audio, I find that if I am working at 16/44.1k I feel the same "off" feeling. Not as much with 24/44.1k, but 24/48k "feels" the best. Can't really explain it in words though. Sonically, without a doubt 24/48k sounds superior.


Not sure. As I had a childhood with lots of video games I prefer higher FPS. Often movements in the movies make me dizzie.


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## ironbut

As others have said, 44.1k is the standard for cd and for most film, 48k is standard. There are plenty of other standards too. 
The "best practices" for sound archives when digitizing analog sources is 96k. 
https://www.archives.gov/preservation/formats/audio-video-resources

If you see a setting for something like 88.2k or 176.4, believe me that these are also standards for something or the folks who design and sell these clocks wouldn't take the time or money to make these available.

Personally, I feel "cheated" by 16/44.1k. And it would be great if enough consumers asked for better sound. But, I'm not holding my breath.


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## Living Fossil

Fredeke said:


> To me, 48KHz seems to offer the best potential quality gain over 44.1KHz for the least increase in filesize and cpu load.



For film, 48kHz is the standard. With the cease of CDs there in fact nothing wrong working with 48 kHz.

When it comes to "quality" i'll have yet to meet somebody who can tell a difference between 48 kHz and 44.1 kHz and/or 96 kHz.
Dogs and bats probably can.
People who pretend they can, usually even fail in blind tests where it's 32 KHz versus 96 kHz.
Because you need to have really bad ears not to realize that you can't hear ultrasound. 

However, i'm not saying that it can't be good to work with 96 kHz... some plug ins deal better with aliasing issues etc. in 96 kHz.


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## Fredeke

Living Fossil said:


> However, i'm not saying that it can't be good to work with 96 kHz... some plug ins sound deal better with aliasing issues etc. in 96 kHz.


Interesting. I didn't know that.
I use 96k for sampling, to retain full spectrum when pitching down.


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## Fredeke

Ok, it seems I'm the oldest fart here (yes that's an old photo).
I was actually there when the CD format hit the market. So here's how it went, historically : CDs had a sampling frequency of 44.1kHz because you needed a flat response up to 20kHz, and then a very steep analog filter before digitizing. The steeper they could make at the time was a filter that went from 0dB to -96dB (I think), which is the dynamic range of 16bits, between 20kHz and 22.5kHz. Which was quite a feat if you ask me. Coming from tape and vynil, some people complained the CD sounded "cold" or "metallic", and I wonder to which extent this was due to the first generation of these extreme low-pass filters. At the time, everything was recorded and mastered at 44.1/16, there was no difference between professional and consumer resolutions. It was just the best that could be done (remember the first digital multitrack recorders used tape ! So data rate was a real issue.) Then the DAT came (a stereo digital audio tape shaped like a miniature vhs), and with the professional needs in mind, it was decided to set its frequency at 48kHz, to allow for about 10% of varispeed without impacting the audible range. Like CD, DAT was 16bits linear. It could also record at 44.1kHz, and there was an LP mode at 32kHz/12bits exponential aimed at radio stations that allowed the tape to run twice slower (hence last twice longer), but never really caught on, as far as I know, though it still sounded pretty decent.
Now the situation is different: AD converters work at a frequency of about 1MHz, then downsample internally. This allows for a much gentler analog filter (going from 0dB to -144dB, the 24bits dynamic range, over the several octaves between 20kHz and 500kHz), then a steep digital filter is applied before downsampling, because digital filters handle steepness more transparently than analog ones. (Also, some converter manufacturers today choose to use a less steep filter when working at 96kHz, for a usable bandwidth of only 30kHz instead of 40kHz, but with even less phase rotation below 20kHz than at 44.1 or 48.)

I know all that. My question was: are there still good reasons to use 44.1kHz today, given that everything else is a multiple of 48 ?
(Wow, that was a number salad !)

Thank you all for your inputs. I hope to read more.

[EDIT: I overlooked that 44100Hz was also picked for easier synchronisation with NTSC video. (There'd be more to say about this, but the number salad would become indigestible. Suffices to say there were complications from that color NTSC has a slightly different frame rate than black and white NTSC.) Anyway, isn't NTSC even more obsolete than CDs?]


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## JamieLang

44.1 never been the standard for production of music recordings.


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## Fredeke

JamieLang said:


> 44.1 never been the standard for production of music recordings.


Are you sure about this ?
http://artsites.ucsc.edu/ems/music/equipment/digital_recorders/Digital_Recorders.html
https://en.wikipedia.org/wiki/Digital_Audio_Stationary_Head

At which frequency do you produce music ? (Me, about an album per year )


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## Dietz

In this context, the link to the famous white-paper from The Man Himself is always due:

-> http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf


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## Fredeke

Dietz said:


> In this context, the link to the famous white-paper from The Man Himself is always due:
> 
> -> http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf


Thanks ! I knew some of these arguments, but not all of them.
A detractor of high sampling frequencies (whose name I can't recall) summerized it a such: the money spent at conveying ultrasonics would be better spent at conveying the audible spectrum more accurately. I love that .
But this guy here advocates the use of slightly too fast sampling rates (emphasize _slightly_), but accepts that since there's no such standard, we should use 96k, but certainly not more, in a compelling and educational argument.

But anyway, that wasn't really my question. I see nobody actually coming in defense of poor old 44.1 kHz ! At least not vs. 48k.

It surprises me because I thought most music studios record at 44.1k and provide 44.1k masters (when they don't go for 96 or 192k). And that most sample libraries aimed at music producers are either 96 or 44.1 kHz, but rarely 48 (or 64, or what else ever).
Am I wrong?

If I released a commercial sample lib at 48 kHz, would it attract less buyers than 44.1k ?
If I provided a music client with a 48 kHz master, would I look bad for it ?

It seems to me every industry works with 48k or its multiples, except the music industry.


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## chrisr

Surprised no has trotted out the old Beethoven's 9th theory. https://www.snopes.com/fact-check/roll-over-beethoven/


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## Dietz

Fredeke said:


> I see nobody actually coming in defense of poor old 44.1 kHz !


Well - here I am (... sort of  ...). Under _ceteris paribus_-conditions I would opt for producing in 96 kHz, of course. But as an old-timer who started to work in this business just when commercial digital audio was in its infancy, I have yet to stumble a across a situation where 44.1 is the _actual_ problem of the production at hand.


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## Fredeke

Dietz said:


> Well - here I am (... sort of  ...). Under _ceteris paribus_-conditions I would opt for producing in 96 kHz, of course. But as an old-timer who started to work in this business just when commercial digital audio was in its infancy, I have yet to stumble a across a situation where 44.1 is the _actual_ problem of the production at hand.


So, habit, and no compelling reason to change ?

Me, I once heard artefact production due to D-A-D conversion from 44 to 48k, but it was with prosumer equipment of the early 90s, and it happened only once. Not even sure I indentified the cause right.

But I'd be open to arguments other than sound quality as well. 
(Practicality, nostalgia, image... whatever ; theoretical, practical, silly... anything)

Btw, thanks for teaching me some new latin.


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## Dietz

Fredeke said:


> Btw, thanks for teaching me some new latin.


:-D ... that's about as good as it gets.



Fredeke said:


> But I'd be open to arguments other than sound quality as well.
> (Practicality, nostalgia, image... whatever ; theoretical, practical, silly... anything)



From a sampling POV things are quite obvious: The lower the sampling rate, the smaller the samples themselves will be, thus less CPU-, RAM- and storage-consumption. OTOH, you will be able to achieve lower latencies when working with higher sampling rates (which is of little importance for me as a mixing engineer, though). There's a certain "impress you clients" aspect when working with higher SRs, no doubt, but luckily that's nothing I have to deal with very often. 

... in a nutshell: When I have to decide which SR to choose for an upcoming production, I'll tell them "48 kHz, but 44.1 won't be a showstopper either".

_Sidenote: 24 bit AD/DA conversion (at least) and 32 bit FP DAW environment is mandatory, of course._


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## Saxer

In my youth the first investment of teenagers after owning a bicycle was a good hifi stereo system. It was common to work in holidays to afford one. This times are gone.

Today the kids listen even to bass heavy hip hop on their phones. All of them.

I often make playbacks for live singers, balletts, travesty show guys etc. When I send them playbacks 95% of them listen to it exclusively on laptop speakers, iPads or phones. They don't own something else. Same for "pro's" in advertising agencies and even film production companies. They never hear anything below 100Hz and then discuss about the mix. People performing music on stage and the best sound system they own is in their car.

For most 'music lovers' today the Windows 3.0 media player would be enough.


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## Fredeke

Dietz said:


> _Sidenote: 24 bit AD/DA conversion (at least) and 32 bit FP DAW environment is mandatory, of course._


Of course. 



Saxer said:


> Today the kids listen even to bass heavy hip hop on their phones. All of them.


I know... Today's youth is so perverted !


Saxer said:


> People performing music on stage and the best sound system they own is in their car.


And car stereos are so built-in that they've become difficult to customize. So, ok, the basic built-in setup is better than what you got in the 80s or 90s, but nobody has a really good custom car stereo anymore. Everything is average.


Saxer said:


> For most 'music lovers' today the Windows 3.0 media player would be enough.


You're being too kind.

In conclusion: We mix for posterity. 

(Note: my GF makes me let go of my sonic demands in everyday life, and in return I teach her to not tolerate mediocre sound. So we meet somewhere in the middle. Yesterday it was her who asked me to plug the laptop into at least a Bose. I take it as a small victory )


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## JohnG

Kids today...

Ok, it's hackneyed to have a go at young people, but my anecdotal investigation suggests that they do listen to music through the most appalling playback systems. Half the time it's through the iPhone speaker itself. Not some external thing, the actual built-in iPhone speaker that can't be larger than, what, a dime?


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## Fredeke

JohnG said:


> Ok, it's hackneyed to have a go at young people, but my anecdotal investigation suggests that they do listen to music through the most appalling playback systems. Half the time it's through the iPhone speaker itself. Not some external thing, the actual built-in iPhone.


I know. Everything sounds like mosquito buzz. When I go to friends' place who listen to music like that, I say either you plug in at least small speakers, or you stop the music, or I go. Fortunately all my friends aren't this tasteless, even among the younger ones ! Some like real sound. (I wouldn't say good, but... real, at least.)

They certainly wouldn't care about sampling rate anyway


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## JohnG

Fredeke said:


> They certainly wouldn't care about sampling rate anyway



exactly. 

The advantages of this or that sample rate -- at least as a practical matter -- is wildly exaggerated.


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## germancomponist

@Saxer Great post! I agree 100%


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## Fredeke

Well... If the answers I got here are any indication... Sampling rates don't matter so much.

I always work at 48k (when I'm not sampling for 96k libraries), but all my friends record at 44.1, so it can be a minor problem when sharing sessions - or no problem at all depending on the software they use. But when sharing back and forth that's a lot of conversions. I tried to convince them to go for 48k but apparently I'm the one who should yield to the majority.

I just wanted to try and determine "who's right" ... But apparently the answer is "who cares?". I was somehow expecting the topic to be more controversial. But ok, I can work with _who cares_


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## JamieLang

Fredeke said:


> Are you sure about this ?
> http://artsites.ucsc.edu/ems/music/equipment/digital_recorders/Digital_Recorders.html
> https://en.wikipedia.org/wiki/Digital_Audio_Stationary_Head
> 
> At which frequency do you produce music ? (Me, about an album per year )



Yes, I am quite sure, having been there. I could write a novel, but it won't matter. This I've learned about the internet Tower of Babel. 

To answer your question directly, I have used 24/88.2 for a number of years....until recently, a local mastering engineer pointed out that while 96khz provides no direct value in the conversation process, that it DOES in delivery--because consumer playback always halves (where it can't play it natively)....and as anyone who's done this for any time will tell you: 48 is "more better" than 44 than 88 is to 48. 

It's not, IME, exaggerated. It's misunderstood. People who expect HD to be some revelatory "now it sounds amazing", will find it exaggerated. People who value the fidelity to their ANALOG input, will find it indispensable.


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## JamieLang

But, to clarify--that's is not "the standard", what I use. 48khz was the standard my whole "coming up" in the 90s. The argument between "HD or not" references 48 vs HD in PRODUCTION....and 44 vs HD in delivery--different jobs and different weights of "need". Don't conflate them. 

CD was ALWAYS a lossy format.


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## Dietz

JamieLang said:


> CD was ALWAYS a lossy format.


Not in the sense of data reduction. 44.1 might not be "hi end" by any means, but what went in there stays in, because linear PCM is not lossy like mp3, for example. 

_/nit-picking _


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## Dietz

Fredeke said:


> But when sharing back and forth that's a lot of conversions.


This I don't get. I work in any SR a production has been started with - 44.1, 48, 96, you name it. No resampling involved. Why would you do that ...? (... I'm asking out of real interest, not because I think that I'm right and you're wrong.)


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## JohnG

Fredeke said:


> I always work at 48k (when I'm not sampling for 96k libraries), but all my friends record at 44.1



*So What Should You Use?*

The delivery format for a lot of media music is 48k, so a lot work at that rate.* However, many use 96k to record live players, even if the delivery is 48k. Even if the producer/sound guys on smaller budget projects are willing to accept 96k, one never knows what they do on the dub stage....the minute your back is turned...... 

That said, a lot of sample libraries are recorded at 44.1 or (sometimes) 48, so then you have to ask, "is there a benefit to recording lower-sample-rate samples at 96k?" Most knowledgeable people argue that recording 44.1k samples at 96k simply takes up more disk space and adds nothing except, depending on your specific rig, some potential benefit for some plugins and effects.

If you are recording a live orchestra (or anything live), that's another matter. The standard recording sample rate for live players is (generally) 96k, irrespective of delivery specs, but not always and not for every project. Some people use 192k but I've seen arguments that 96 is sufficient for human ears.

*Does Anyone Hear on Their Crappy Speakers?*

As far as your engineer's faith in 96k, @JamieLang , I think the answer is "it depends." If you're in a mastering or other high-end playback room that's well constructed, with good gear, sample rate is important.

However, plenty of consumer gear plays back at 44.1 and besides, in many/most cases it doesn't matter because their speakers/connectors/D/A converters and speakers furnish such feeble audio quality to start with.

I am stunned at how many music consumers -- avid, enthusiastic ones, who play music constantly -- do so through horrible speakers / headphones / bluetooth devices.

Accordingly, because of the shockingly inferior equipment most use to consume music, while acknowledging that there is an important debate about all this, not many can hear the difference even between 96 and 44.1. Far fewer -- vanishingly few, actually -- can hear the difference on their systems between 48 and 44.1.

-----------
* One of my rather successful pals (pretty well known) uses 44.1 to compose if he's using samples because the majority of his libraries are recorded at 44.1. He thinks it's a mistake to work at 48k. Just for what it's worth.


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## Fredeke

@JohnG : Thanks. That was clear and well laid out.


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## Fredeke

Dietz said:


> This I don't get. I work in any SR a production has been started with - 44.1, 48, 96, you name it. No resampling involved. Why would you do that ...? (... I'm asking out of real interest, not because I think that I'm right and you're wrong.)


You are right. I got carried away. Conversions only happen at export/render/bounce or freezes.


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## dgburns

I agree with everything @JohnG wrote.

I work at 48 locked and loaded. That said, I’d love to bump that up to 96k, since the few times I did, I hear the benefits when the final is crunched down to mp3.

But maybe this is all just personal

Edit

Unless you write in the key of ‘D’, in which case, it’s gonna just sound better at 44.1  especially if it’s the LSO


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## Fredeke

dgburns said:


> Unless you write in the key of ‘D’, in which case, it’s gonna just sound better at 44.1  especially if it’s the LSO


I know explaining jokes can make them fall flat, but would you please indulge me, for my education?


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## dgburns

Fredeke said:


> I know explaining jokes can make them fall flat, but would you please indulge me, for my education?



***sigh***

Bad joke on my part. 

So I just made the assumption that John was referring to a certain composer who writes regularly in the key of D. This certain individual sampled his own orchestra while sampling was in it’s infancy, hence the 44.1 (and no real good reason to change that)

carry on >>>


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## brenneisen

Hans's samples were recorded at 88.2


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## dgburns

brenneisen said:


> Hans's samples were recorded at 88.2



maybe you can read-

https://www.soundonsound.com/techniques/scoring-pirates-caribbean-iii

original samples recorded to a DA-88 , which were at 44.1 16 bit.

*** again sigh ***

the freakin’ notes matter, not much else .....IMNSFHO


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## JohnG

dude you were just wrong. no big deal


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## brenneisen

nah, not a good point.

we're talking about options, not limits. DA88 couldn't record higher than 48 and S760 couldn't play higher than that also. 

so it's not like he chose to keep it low.


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## Nick Batzdorf

As Dietz says (paraphrasing what I recall from his post), it's unlikely that you'll ever be able to blame a bad recording on 44.1.

And the quality of the converters made a big difference in the DA-88 days.

Well, it still does of course, but at this point we're probably in the golden age of digital recording. Everyone uses the same converter chips, and - while I'm not an electronics *ngai* - I'll go out on the limb and say that design engineers have figured out the analog circuitry surrounding those chips by now.


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## NoamL

JohnG said:


> I am stunned at how many music consumers -- avid, enthusiastic ones, who play music constantly -- do so through horrible speakers / headphones / bluetooth devices.



Guilty as charged John. I listen to 90% of music through these






They're easy to take on and off, very light on the head and ears (I wear glasses so anything that presses on the frame gets tiring/painful after a while), super cheap to replace when they break every 2 years, and most perversely of all, I've become really familiar with their response so I can even mix on them. I've written entire tracks only listening through these or at least alternating with my ATH-M50's. I'll never be a great mixer so why not listen on a system that's convenient. In many cases the clients are listening on systems this bad or worse. Once I had a callback from a short film director who was listening to my track - THE FIRST LISTEN - in his car... on the 405.... through an iPhone.


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## Jeremy Gillam

I always make sure my mixes bang on my iPhone speaker.


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## Fredeke

Nick Batzdorf said:


> it's unlikely that you'll ever be able to blame a bad recording on 44.1.


That's for sure.
If we must go down that road, even some recordings from the 1940s are still great to listen today. The limited bandwidth of 1940's equipment didn't prevent good engineers from doing a good job.

But where's the fun in being an engineer, if we can't indulge our obsessive compulsions a little bit ? 
I started this thread for the sake splitting hair. (Don't we all like our hair well split ?)



NoamL said:


> Guilty as charged John. I listen to 90% of music through these


Well, another key aspect of any monitoring system, aside from it being accurate and all, is for us to know how it is supposed to sound. And your must know your Sony headphones intimately by now !

(That's why when coming to a studio we don't know, it's a good idea to first listen to songs we know, to get a feel of its monitoring - not that I expect to be teaching anybody anything)



NoamL said:


> I'll never be a great mixer so why not listen on a system that's convenient. In many cases the clients are listening on systems this bad or worse.


And yet you work professionally. That's a relaxing thought 

Let's see if we can send clients 96k mp3 over email and get away with it 
(and I mean 96kb/s, not 96kHz !)



Jeremy Gillam said:


> I always make sure my mixes bang on my iPhone speaker.


That has become a necessary evil.

Me, I've settled for a Bose Soundlink Mini II - the best of crappy speakers :
https://www.bose.com/en_us/products/speakers/portable_speakers/soundlink_mini_ii.html
I can't use any worse. I know I should, but I just can't.
I would rather quit than inflict that on my ears (besides, I only have a dumbphone)

The 17-inch MacBook's built-in speakers were great, but since Apple has reduced the size of their laptops, their sound has become thinner too. In 2002-2003, I had a Mac tower, and the built-in 1'' speaker was a real joy ! It sounded so smooth... It was actually nice to listen to music on it (while all a PC's built-in speaker can do is beep). I don't know how today's trashcan macs sound. I know Harman-Kardon made Apple's speakers in the 2000s... Is it still the case ?


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## Jeremy Gillam

Fredeke said:


> That has become a necessary evil.



I find it weirdly satisfying to get a good racket out of that thing. When I was a kid I had this piece of shit Fischer Price tape recorder with a microphone attached which I used to make my first productions, around 4-5 years old. My dad has remarked that he was able to pick up handclap patterns etc. in Beatles songs coming out of that mono speaker that he never noticed on the HiFi (44.1kHz CD) system in the house. As much as I wish everyone would listen to our music on great systems, it's nice to know that kids everywhere are still getting some enjoyment out of their little mono speakers


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## Fredeke

Jeremy Gillam said:


> I find it weirdly satisfying to get a good racket out of that thing. When I was a kid I had this piece of shit Fischer Price tape recorder with a microphone attached which I used to make my first productions, around 4-5 years old. My dad has remarked that he was able to pick up handclap patterns etc. in Beatles songs coming out of that mono speaker that he never noticed on the HiFi (44.1kHz CD) system in the house. As much as I wish everyone would listen to our music on great systems, it's nice to know that kids everywhere are still getting some enjoyment out of their little mono speakers


Indeed. When rewatching a movie on the laptop, of course I loose everything that's happening in the bass, but sometimes I notice some new things in the mid and high range that otherwise were masked by the bass.


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## Nick Batzdorf

Fredeke said:


> That's for sure.
> If we must go down that road, even some recordings from the 1940s are still great to listen today. The limited bandwidth of 1940's equipment didn't prevent good engineers from doing a good job.
> 
> But where's the fun in being an engineer, if we can't indulge our obsessive compulsions a little bit ?
> I started this thread for the sake splitting hair. (Don't we all like our hair well split ?)



Oh, no question - I'm a total fellow nerd.

The thing is, the argument for higher sampler rates (or the only one I believe) *isn't* the increased bandwidth, it's that the brick wall filter's ringing gets moved out of the audible bandwidth.

Sorry if that's baby talk to you!


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## Fredeke

Nick Batzdorf said:


> The thing is, the argument for higher sampler rates (or the only one I believe) *isn't* the increased bandwidth, it's that the brick wall filter's ringing gets moved out of the audible bandwidth.


Exactly my point.

In the article @Dietz posted a link to ( http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf ), there is a companion argument: since a 20kHz bandwidth means there's a -3dB drop at 20kHz, and since there's AD and DA, and some other steps it mentions, that's several cumulative -3dB drops. Personally I never noticed this, even in my youth when I could hear up to 20kHz. In fact the argument seems mainly theoretical, and I'd like to see its practical validity discussed... Don't modern converter manufacturers have stricter standards for measuring bandwidth (like +/- 1dB, or even less) ?

Nevertheless, I can still see an advantage to increased bandwidth, in the particular case of sampling, when you intend to play a sample across the keyboard: then it can be pitched down without sounding duller, because the ultrasonic harmonics get within the audible range to repopulate the "brightness band" of the spectrum.
I am making a theoretical argument too, here. In practice it's not a magical solution, probably because the highest harmonics are also (usually) the quietest, but it helps a little bit.


----------



## Nick Batzdorf

Lavry is *intense* about this stuff! He also has a famous (among nerds) article about why higher sample rates just screw things up because of all the extra data gagging processors - at least that was my takeaway from copious amounts of detail that I sort of skimmed a few years ago.


----------



## Dietz

Nick Batzdorf said:


> He also has a famous (among nerds) article about why higher sample rates just screw things up because of all the extra data gagging processors


... that would be this one, I assume:

-> http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf



> *Conclusion: *
> 
> There is an inescapable tradeoff between faster sampling on one hand and a loss of accuracy, increased data size and much additional processing requirement on the other hand.
> 
> AD converter designers can not generate 20 bits at MHz speeds, yet they often utilize a circuit yielding a few bits at MHz speeds as a step towards making many bits at lower speeds.
> 
> The compromise between speed and accuracy is a permanent engineering and scientific reality.
> 
> Sampling audio signals at 192KHz is about 3 times faster than the optimal rate. It compromises the accuracy which ends up as audio distortions.
> 
> While there is no up side to operation at excessive speeds, there are further disadvantages:
> 
> 
> The increased speed causes larger amount of data (impacting data storage and data
> transmission speed requirements).
> 
> 
> Operating at 192KHz causes a very significant increase in the required processing
> power, resulting in very costly gear and/or further compromise in audio quality.
> 
> The optimal sample rate should be largely based on the required signal bandwidth. Audio industry salesman have been promoting faster than optimal rates. The promotion of such ideas is based on the fallacy that faster rates yield more accuracy and/or more detail. Weather motivated by profit or ignorance, the promoters, leading the industry in the wrong direction, are stating the opposite of what is true.


----------



## Nick Batzdorf

Dietz said:


> ... that would be this one, I assume:
> 
> -> http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf



Yup!


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## Fredeke

What I take from both articles, is that the ideal rate would be around 60kHz. Some converters offer 64kHz and 128kHz, as multiples of rather obsolete standard 32kHz... Too bad this isn't more common !

(Without knowing it could be pertinent, I've sometimes considered releasing my albums at 64kHz, just to poke fun at the whole 44 vs. 96 debate...)


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## Fredeke

Now that I think of it, another argument in favor of higher sample rates, again in the case of sampling: they are a way of setting loop points more precisely. It often doesn't make a difference, but it definitely does sometimes. Especially for high-pitched sounds with too few harmonics to mask the looping click. Granted, this might be an expensive trick (in terms of file wieght), and there are other ways, but still, it's easy and works well.


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## Dietz

Fredeke said:


> Now that I think of it, another argument in favor of higher sample rates, again in the case of sampling: they are a way of setting loop points more precisely. It often doesn't make a difference, but it definitely does sometimes. Especially for high-pitched sounds with too few harmonics to mask to looping click. Granted, this might be an expensive trick (in terms of file wieght), and there are other ways, but still, it's easy and works well.



Are you sure that this is the real reason for what you're hearing? The sampling theorem says that below Nyquist the wave gets reconstructed perfectly, so the SR shouldn't make a difference, in that respect - at least not within the range of human perception.

... but I'll happily learn something new!


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## Fredeke

Dietz said:


> Are you sure that this is the real reason for what you're hearing? The sampling theorem says that below Nyquist the wave gets reconstructed perfectly, so the SR shouldn't make a difference, in that respect - at least not within the range of human perception.
> 
> ... but I'll happily learn something new!


Sampling theorem is right, but I don't think it applies here. The effect would be the same if we were looping analog signals, as long as the looping points are constrained to happen at discrete timings.

Sampling theory says the signal's period doesn't need to last a whole number of samples (aka words, in this case). But samplers require that a loop be a whole number of words long. So chances are the loop's length won't match the sampled note's period exactly.

But suppose that's what you want to do. Then, reduce the duration of a word (say from 1/48000 sec. to 1/96000 sec.) and you'll increase the precision to which the loop's duration will match the signal's period.

This is a good trick even if you want to loop more than just one cycle of the waveform.
I'll try to illustrate...

Say you sample a note at 48KHz. Now you want to loop it. You'll need a loop length that is a multiple of one period (the duration of a wave cycle) to avoid any clicking, as well as to keep perfect pitch. But is the wave's period a perfect multiple of 1/48000 seconds ? It is most likely not. Yet the loop's length must necessarily be. Hence, the loop will not exactly match the period. The error in length is small enough as to not matter for most of an instrument's range... But for the highest-pitched notes, which have very short periods, that error becomes a significant portion of the note's period. So the loop becomes significantly faulty.

There are several ways around this :
- you can loop several periods, and find a loop length that is both a multiple of the wave's period and close enough to a multiple of a word's length (1/48000 sec. in this example). I do this all the time, and it get rids of the tuning issue very quickly. But you might still be left with the clicking issue, if you have to also sync the loop to other periods, like a vibrato or a tremolo, a LFO, or unison beat... Then it could take a very long sample before all periods coincide, and could possibly get lost in trying to find identical points that are so very far apart. (Also, you can't really use this solution if you're looping single cycles, say for use in a digital oscillator.)
- you can crossfade around loop points, and in the case of a low-harmonics signal, it would probably work well. But if the waveform has many harmonics, then the slight fault in timing would create a comb filter effect. Since the combing effect would affect mostly frequencies above nyquist, this was a problem only at low sampling rates, on old hardware samplers. You can still have a problem though, if you're looping more than one period and the harmonic content is unstable, because in a jagged waveform, a hard cut sometimes sounds much better than a crossfade.

- last solution (that I know of) : increase the sampling rate (even by upsampling an already recorded sound). If a word's duration is twice shorter (1/96000 sec. instead of 1/48000 sec.), then the error in timing is reduced by about as much. Quadruple the rate and you get four times the precision, etc.

In practice, by combining all those tricks, I never need to go higher than 96KHz. But this added precision often helps.

Conversely, when converting looped samples to a lower rate, some loops get clicky and/or slighty out of tune, because the loop points' positions get rounded. Again, this mostly affects the highest pitched tones.

I hope I was clear. This is challenging to put into words.

(Note that I sample analog synths. I don't know if sampling acoustic instruments would confront me to these issues as much. I have little experience in that.)


----------



## Dietz

Fredeke said:


> [...]
> - last solution (that I know of) : increase the sampling rate (even by upsampling an already recorded sound). If a word's duration is twice shorter (1/96000 sec. instead of 1/48000 sec.), then the error in timing is reduced by about as much. Quadruple the rate and you get four times the precision, etc. [...]



Interesting, thanks a lot!

I think I understand better now what you're doing. I have to admit that the last time I looped single wave cycles was on my Ensoniq Mirage, back in the 80ies. It might very well be that I never ran into that issue therefore, working on 8 bits samples by editing paramter values in hexadecimal. 8-)


----------



## Fredeke

robgb said:


> Because it's enough.


Best answer ever


----------



## bill5

For newbies going "holy #### just tell me what rate I should record at" the answer is 44.1 or 48, with the diff being negligible about oh 99%ish of the time. 96 on up is overkill and buys you nothing...in fact if anything, is worse.


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## WaveRider

bill5 said:


> 96 on up is overkill and buys you nothing...in fact if anything, is worse.



But when I've attempted to set my DAW to 96 and compose/record everything at this setting, to my ears everything sounds noticeably better. Even 88 sounds much better than 48 to my ears. The only reason I CAN'T record higher than 48 is because of computer/speed limitations.

Kind of reminds of SD vs HD in the video world. Everything was 720x480 up until around 2006. Now it's 1920x1080 and soon standard will be 4k. 8k is on the horizon someday.

But why should audio fidelity be stuck in the 1980's?


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## bill5

WaveRider said:


> But when I've attempted to set my DAW to 96 and compose/record everything at this setting, to my ears everything sounds noticeably better. Even 88 sounds much better than 48 to my ears.


Confirmation bias is a powerful thing.


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## WaveRider

bill5 said:


> Confirmation bias is a powerful thing.



Like the guy in 2006 who said wow, HD looks much better.


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## Divico

WaveRider said:


> Like the guy in 2006 who said wow, HD looks much better.


The thing is most libs are 44/48. So the only benefit from going higher with samples is mixing plugins that dont oversample.


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## Fredeke

WaveRider said:


> But when I've attempted to set my DAW to 96 and compose/record everything at this setting, to my ears everything sounds noticeably better. Even 88 sounds much better than 48 to my ears. The only reason I CAN'T record higher than 48 is because of computer/speed limitations.



As I understand it (but never heard any of it first hand), there may be reasons 96k sounds better than 44.1k in your system, that have nothing to do with the actual increased bandwidth. I'm not saying it's confirmation bias - I'll trust you can trust your ears, but I'd be surprised the increased bandwitdth is the reason you hear a difference. There could be other factors, like:

- Even in the "mere" audible range, your interface might sound better at higher rates, for esoteric reasons having to do with the particulars of its circuitry. Or maybe it's actually worse, but you happen to like it better - which would be fine of course.

- Maybe it does sound better, because the brickwall low-pass filter applied at the nyquist frequency (half the sampling rate) creates ripples in the octave below it - which at 88.2/96k is still above, hence outside of, the audible range. (I personally think I heard that difference maybe once in my life, and I'm not even sure about it.)

- Some software work better at higher rates, especially the modeling ones, because whatever is modeled can be modeled with higher precision (even though the ultrasounds themselves don't matter) - according to some developers.

- Maybe your amp or speakers can't render ultrasounds, and produce intermodulation distortion instead. And maybe you like that effect. In this case though, the "gain" would be utterly futile, since it would only be an artefact of your monitoring system.



WaveRider said:


> Kind of reminds of SD vs HD in the video world. Everything was 720x480 up until around 2006. Now it's 1920x1080 and soon standard will be 4k. 8k is on the horizon someday.
> 
> But why should audio fidelity be stuck in the 1980's?


I don't think that's a fair parallel.

The objective improvement in audio fidelity (related to sample rate) is very slight at best (did I mention this was a nitpicking thread?) - nothing to be compared to the tremendous improvements video has gone through over the last decades !


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## bill5

WaveRider said:


> Like the guy in 2006 who said wow, HD looks much better.


More like the guy in 2006 who said wow, 1080 looks so much better than 720. In fact I knew such a guy and won a free lunch when I proved he couldn't reliably tell the diff.  Hey you're convinced it's better, knock yourself out, there is something to be said for things that make you feel better about your music regardless of how much diff they actually do or don't make. I think we have all been there at one time or other.


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## WaveRider

Fredeke said:


> As I understand it (but never heard any of it first hand), there may be reasons 96k sounds better than 44.1k in your system, that have nothing to do with the actual increased bandwidth. I'm not saying it's confirmation bias - I'll trust you can trust your ears, but I'd be surprised the increased bandwitdth is the reason you hear a difference.



Thanks for the reply. You bring up some really interesting points here that I never considered. I'm using an old MOTU 1296 interface that records in 44, 48, 88, and 96. If you sat here with me in my studio you would definitely agree that 88 and 96 sound better than 44 and 48 (assuming your ears are still in good shape). This whole time I just assumed that 88 and 96 are 'better' because on my system, there's an audible difference. But my interface is pretty old (purchased in 1999) so the audible difference could very well be system independent. 





Fredeke said:


> I don't think that's a fair parallel. The objective improvement in audio fidelity (related to sample rate) is very slight at best (did I mention this was a nitpicking thread?) - nothing to be compared to the tremendous improvements video has gone through over the last decades !



Yeah that was definitely an exaggeration. But again, I was speaking from my own personal experience which I now realize could differ across studios and equipment.


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## Fredeke

WaveRider said:


> Thanks for the reply. You bring up some really interesting points here that I never considered. I'm using an old MOTU 1296 interface that records in 44, 48, 88, and 96. If you sat here with me in my studio you would definitely agree that 88 and 96 sound better than 44 and 48 (assuming your ears are still in good shape). This whole time I just assumed that 88 and 96 are 'better' because on my system, there's an audible difference. But my interface is pretty old (purchased in 1999) so the audible difference could very well be system independent.


If your interface dates from 1999, there might be yet another factor (not that modern interfaces never exhibit this, but I assume it was more pronounced in the days) :
- If the nyquist lowpass filter is set at 20KHz, that would normally mean -3dB at 20KHz. Point is, it might not be flat up to 20KHz. Assuming -3dB, that would be -3dB at recording, and -3dB at playback (because of the anti-alias filter at the output of the converter - though even by 1999, oversampling would have permitted less drastic anti-aliasing, but you never know). Total -6dB. Of course at double frequencies, that non-linearity is moved way up where it doesn't matter.
If your ears are still young, maybe you can ear this ?

It's all a matter of taste. Me, when I could still hear up to 20KHz, I often lowpassed my masters at 16KHz, because I found it sounded cleaner. Can't tell the difference anymore, but still use the lowpass anyhow. On the other hand, I suppose most engineers like their bandwidth to go all the way up to 20k.

Anyway, you might like following that thread, which is kind of a continuation to this one : https://vi-control.net/community/threads/when-and-why-not-to-use-high-sampling-rates.81386/


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## SupremeFist

I just noticed that I had a project accidentally set to 96/24, and when I changed the sample rate back to 44.1 it sounded notably worse (less warm and rounded, more "tinny"), which I didn't expect at all. So maybe this is a thing?


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## storyteller

SupremeFist said:


> I just noticed that I had a project accidentally set to 96/24, and when I changed the sample rate back to 44.1 it sounded notably worse (less warm and rounded, more "tinny"), which I didn't expect at all. So maybe this is a thing?


Haha. Yep! This IS a thing! Something else that tends to be audibly noticeable is that renders from a project can often sound noticeably more warm and rounded than when played back live. Specifically, you can hear this in reverb. The quality of offline rendering versus live playback is a real thing too (depending on how your DAW is likely setup since realtime playback tends to reduce CPU cycles when possible)!


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## OleJoergensen

SupremeFist said:


> I just noticed that I had a project accidentally set to 96/24, and when I changed the sample rate back to 44.1 it sounded notably worse (less warm and rounded, more "tinny"), which I didn't expect at all. So maybe this is a thing?


I recently experienced the same. Especially external hardware reverb sounds much better at 96 khz...


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## John Longley

As stated above:
1. Sony/Phillips made it that way with Red Book forty years ago. Hard to change.
2. Manageable file sizes
3. Less CPU intense
4. Humans can't reliably tell the difference in blind testing.
5. Because very very few people understand sampling theory and those that are evangelists for high SR often spew forth nonsense (see DSD evangelists).


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## Dewdman42

96k is causing climate change!

On a more serious note, I record at 48k and see no point to 44.1k anymore now that CD's are toast. Most people do not need to record at 96k and are simply incurring CPU overhead and consuming more disk space for something that nobody will hear in the in final mix, even *IF* you can hear some difference in your studio while you're tracking the parts at a higher sample rate. There is probably some very minute difference in the final mix that someone can measure and show and maybe even claim to hear, but most people will not hear it....and you're just waiting resources, using more electricity!


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## SupremeFist

Dewdman42 said:


> 96k is causing climate change!
> 
> On a more serious note, I record at 48k and see no point to 44.1k anymore now that CD's are toast. Most people do not need to record at 96k and are simply incurring CPU overhead and consuming more disk space for something that nobody will hear in the in final mix, even *IF* you can hear some difference in your studio while you're tracking the parts at a higher sample rate. There is probably some very minute difference in the final mix that someone can measure and show and maybe even claim to hear, but most people will not hear it....and you're just waiting resources, using more electricity!


I mean, I was actually shocked at the difference and I don't have great monitors and a treated room, this was just listening through HD650s.


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## Dewdman42

sure while you're tracking. I doubt your final produced 44 or 48k mix will show any noticeable difference, certainly not enough to justify all the bandwidth your sucking through your system. IMHO.


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## Dewdman42

someone also pointed something else out..which is that when you're using your high end 96k device...it could very well be that the device itself is optimized in some ways such that you get less jitter or other things related to the device, it could are any number of things that might run smoother at the higher sample rate in some way, while you monitor what you're doing... Still doesn't mean you will see any effective difference in the final produced and delivered mix.


----------



## John Longley

Dewdman42 said:


> someone also pointed something else out..which is that when you're using your high end 96k device...it could very well be that the device itself is optimized in some ways such that you get less jitter or other things related to the device, it could are any number of things that might run smoother at the higher sample rate in some way, while you monitor what you're doing... Still doesn't mean you will see any effective difference in the final produced and delivered mix.


Some DACs auto downsample from their native internal sample rate. So this is a thing, but I have not tested it because I stopped caring years ago. I own a couple of sets of very serious three and four way mains and I cannot reliably tell the difference blind consistently in testing, even if I can tell there is a difference. Im not alone, once it comes to actually testing people tend to stop talking lol

I think there are numerous merits for high SR and high bit depth during production (dsp, headroom internally), but for program and for "sound"..... Meh


----------



## Dewdman42

Here's what Craig Anderton had to say about this long standing controversial debate: https://www.keyboardmag.com/lessons/should-you-record-at-96khz


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## John Longley

Dewdman42 said:


> Here's what Craig Anderton had to say about this long standing controversial debate: https://www.keyboardmag.com/lessons/should-you-record-at-96khz


I would also recommend all of the old Paul Frindle posts on gearslutz from years ago (I avoid there in recent years) as well as the white papers from Benchmark.


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## SupremeFist

Dewdman42 said:


> Here's what Craig Anderton had to say about this long standing controversial debate: https://www.keyboardmag.com/lessons/should-you-record-at-96khz


That's a really nice explanation, thanks!


----------



## Nathanael Iversen

Thing #1. It's your DAW, you can do whatever you want for whatever reason you choose.

Thing #2. The Recording Academy's (the Grammy's) Producers and Engineers Wing has published technical guidance for recording music. Really! Read the list of signatory engineers at the opening of the documents and you can't help but recognize the names. It is a who's who in the industry for sure. They lay out their reasons and recommendations at length.

https://www.grammy.com/sites/com/files/recommendations_for_hires_music_production_09_28_18_0.pdf (Here is the document for recording audio)

https://www.grammy.com/sites/com/files/delivery_recommendations_for_recorded_music_projects_final_09_27_18_0.pdf (Here is the document for delivery formats for recorded music)

https://www.grammy.com/sites/com/files/surroundrecommendations.pdf (Here is the article for surround sound productions)

Every time this topic comes up on any forum, everyone immediately activates the echo chamber of personal opinion, psuedoscience, actual science, and hearsay. The threads all look the same in minutes. In this one, no one posted the Monty video on how digital audio works yet, though, and we don't have any personal attacks, so that is a plus....

I appreciate the thoughtful way this is handled in the above documents. You can agree or disagree (see thing #1 above). I think it is useful for industry veterans and leaders to agree where possible on technical standards of excellence and professionalism. The best practices is part of the guidance the senior practitioners can offer the rest of us. It turns out they have. And they wrote it down in case anyone else wanted to know. But, it doesn't stop thing #1 from still being true. 

When working with samples, mine are almost all 48kHz, so that is what I use. When recording my band, I follow the Academy guidelines and record at 96kHz for the reasons listed in the document above. I have plugins that sound better at 96Khz for certain. It is also certain that in my mid-40's I can't perceive sine wave tones above 16kHz let alone 20Khz or 40Khz. SSD space is practically free where audio is concerned, and DSP and CPU are plentiful in my studio. Whether it matters or not, I've chosen to just accept the wisdom of those far more experienced than me - I have no reason to think I have a better opinion and no lack of resources to make it true.


----------



## Nick Batzdorf

Dietz said:


> Interesting, thanks a lot!
> 
> I think I understand better now what you're doing. I have to admit that the last time I looped single wave cycles was on my Ensoniq Mirage, back in the 80ies. It might very well be that I never ran into that issue therefore, working on 8 bits samples by editing paramter values in hexadecimal. 8-)



I just saw this from almost a year ago.

Re: the post just above this one that Dietz is talking about - if I understand right, if you're not going to xfade the loop (which I suspect that you're going to do anyway), you're going to loop at a zero crossing. Ergo it shouldn't matter what the SR is in that case.

Or are you saying you can use the end of the recording at a more exact loop point at a higher SR? I'd have to hear an example of that making a difference, but I still don't understand how it would - because a sine wave below Nyquist is a sine wave below Nyquist.

Someone mansplain me what I'm missing.


----------



## sumskilz

With well-designed modern converters, 44.1 kHz objectively provides a more accurate reproduction of the source material than 96 kHz does. With only about one millisecond of extra latency compared to a 96 kHz sample rate, decent modern converters working at a 44.1 kHz sample rate can reduce aliasing induced distortion to below -100 dB at 20 kHz, whereas as the ultrasonic frequencies present in material sampled at 96 kHz produce intermodulation distortion throughout the audible range which can easily be as high as -30 dB.

Here is a detailed illustration of why that is the case: https://www.gearslutz.com/board/mas...ot-high-resolution-quot-audio-processing.html

It should be obvious that -100 db of inharmonic distortion limited to frequencies that most people can't even hear would be preferable to inharmonic distortion throughout the audible range at amplitudes that can be as much as 70 dB higher. Why then does 96 kHz not sound terrible compared to 44.1 kHz? Keep in mind, I said 44.1 kHz is objectively more accurate than 96 kHz (within the audible range). People may nevertheless prefer the sound of the extra noise floor produced across the audible spectrum by 96 kHz. The more likely reason is that listeners can't actually hear the differences despite the fact that said differences are objectively measurable. The reason is that ultrasonic content tends to be much lower in amplitude than the content within the audible spectrum, so the intermodulation distortion produced by ultrasonic frequencies is largely masked by the intermodulation distortion produced by frequencies within the audible range.

In a 2010 AES study on sample rate discrimination, expert listeners in blind tests could not discriminate between material recorded at 88.2 kHz and 44.1 kHz in most cases. When they could consistently discriminate with some accuracy, they believed 44.1 kHz to be the higher sample rate, presumably because they thought it sounded better.

The moral of the story is that there isn't much to worry about either way. There are reasons to use high sample rates if you're pitching down, if you need a millisecond less latency, or if you're using non-linear plugins that lack good anti-aliasing filters and/or oversampling. In that case, you may benefit from using an ultrasonic lowpass filter before and after every nonlinear process. In my opinion, considering the tradeoffs and the current state of technology, 44.1 kHz and 48 kHz are superior to higher sample rates in most circumstances, but it doesn't really make much of difference to what people can actually hear. So it's probably best to just work at whatever rate you need to deliver at.


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## SupremeFist

sumskilz said:


> With only about one millisecond of extra latency compared to a 96 kHz sample rate, decent modern converters working at a 44.1 kHz sample rate can reduce aliasing induced distortion to below -100 dB at 20 kHz, whereas as the ultrasonic frequencies present in material sampled at 96 kHz produce intermodulation distortion throughout the audible range which can easily be as high as -30 dB.



That's really interesting. So essentially I might be preferring the 96Khz playback on my particular project even though it's actually _more_ distorted, in a similar way to how some people prefer the sound of vinyl to digital?

Alternatively maybe one of my plugins does not work well (re oversampling) at 44.1. This particular project is very simple, containing a handful of sample libraries (all recorded at 44.1) plus the following FX:

NI Replika delay
Logic Graphic Equalizer (from the Vintage EQs collection)
Logic Channel EQ
Logic Chromaverb reverb
Waves Puigchild 670
Waves Abbey Road Mastering Chain
Boz Digital Labs The Wall limiter

Is one of those likely to be the "culprit"?


----------



## sumskilz

SupremeFist said:


> That's really interesting. So essentially I might be preferring the 96Khz playback on my particular project even though it's actually _more_ distorted, in a similar way to how some people prefer the sound of vinyl to digital?


Yeah, it could definitely be that and you wouldn't be alone. If there was already recorded audio that had to be downsampled to match the new rate, there could have been some degradation. Although Logic X's sample rate conversion is really good. Earlier versions of Logic weren't (nor were many other DAWs back then). Downsampling has the potential to cause more artifacts than simply recording at the lower sample rate in the first place, but with more recent software the process is usually pretty transparent.



SupremeFist said:


> Alternatively maybe one of my plugins does not work well (re oversampling) at 44.1. This particular project is very simple, containing a handful of sample libraries (all recorded at 44.1) plus the following FX:
> 
> NI Replika delay
> Logic Graphic Equalizer (from the Vintage EQs collection)
> Logic Channel EQ
> Logic Chromaverb reverb
> Waves Puigchild 670
> Waves Abbey Road Mastering Chain
> Boz Digital Labs The Wall limiter
> 
> Is one of those likely to be the "culprit"?


I don't know about the Logic plugins, but the non-linear Waves plugins I've tested have a bit of aliasing that improves at a higher sample rate. Depending on the material and settings, it may or may not be audible in any particular situation.

The website tells me that limiter has optional 8x oversampling, so it would be easy see if it improves by engaging it.


----------



## SupremeFist

sumskilz said:


> Yeah, it could definitely be that and you wouldn't be alone. If there was already recorded audio that had to be downsampled to match the new rate, there could have been some degradation. Although Logic X's sample rate conversion is really good. Earlier versions of Logic weren't (nor were many other DAWs back then). Downsampling has the potential to cause more artifacts than simply recording at the lower sample rate in the first place, but with more recent software the process is usually pretty transparent.
> 
> I don't know about the Logic plugins, but the non-linear Waves plugins I've tested have a bit of aliasing that improves at a higher sample rate. Depending on the material and settings, it may or may not be audible in any particular situation.
> 
> The website tells me that limiter has optional 8x oversampling, so it would be easy see if it improves by engaging it.


Yep, I already have the oversampling engaged on the limiter, so maybe it's the Waves plugins!


----------



## Nick Batzdorf

sumskilz said:


> the ultrasonic frequencies present in material sampled at 96 kHz produce intermodulation distortion throughout the audible range which can easily be as high as -30 dB.



Shouldn't those freqs - including difference signals - get filtered out before they're recorded?

Don't get me wrong, 96kHz can kiss my ass, to quote a friend who's a forum member here. But I don't quite understand how intermodulation products generated way up there would make it into the human range.

This is Socratic questioning - I'm not saying I'm right you're wrong nah nah nah, I'm trying to understand the argument.


----------



## sumskilz

Nick Batzdorf said:


> Shouldn't those freqs - including difference signals - get filtered out before they're recorded?
> 
> Don't get me wrong, 96kHz can kiss my ass, to quote a friend who's a forum member here. But I don't quite understand how intermodulation products generated way up there would make it into the human range.
> 
> This is Socratic questioning - I'm not saying I'm right you're wrong nah nah nah, I'm trying to understand the argument.


New intermodulation distortion appears every time there is a nonlinear process such as saturation or compression. So all plugins that emulate analog gear create intermodulation distortion. Even if you don’t use any dynamics or saturation plugins, it is unavoidable because a lot of it is created by speakers/headphones when your recorded signal is turned back into sound. New frequencies are produced at both the sum and difference between every single frequency in your recorded material. For example, if you have 4 kHz and 10 kHz, they will create distortion at 6 kHz (10 kHz - 4 kHz) and 14 kHz (4 kHz + 10 kHz). Now add 23 kHz as a single ultrasonic frequency and you will also get distortion at 13 kHz (23 kHz - 10 kHz), 19 kHz (23 kHz - 4 kHz), 27 kHz (23 kHz + 4 kHz), and 33 kHz (23 kHz + 10 kHz). So in this very simplified example, we get the following distortions at each sample rate:

44.1 kHz sample rate: distortion at 6 kHz and 14 kHz
48 kHz sample rate: distortion at 6 kHz and 14 kHz, very slight distortion at 13 kHz and 19 kHz
88.2 kHz sample rate: distortion at 6 kHz, 13 kHz, 14 kHz, 19 kHz, 27 kHz, and 33 kHz
96 kHz sample rate: distortion at 6 kHz, 13 kHz, 14 kHz, 19 kHz, 27 kHz, and 33 kHz
The reason the extra distortion at the 48 kHz sample rate is only very slight is because the ultrasonic frequencies are already rolled off most of the way and so can’t interact much with the audible frequencies. Anyway, this is a very simple example, because in reality, no one's music is made up of just three sine waves, there will be frequencies across the entire sampled bandwidth interacting with every other frequency across the bandwidth. Every ultrasonic frequency multiplies the amount of distortion. And the above example is also the result of a very light nonlinear process. Normally this will repeat multiple times, so that every new frequency created by a sum and difference becomes a new frequency in the next round creating yet more new frequencies at the sum and difference between every other preexisting and new frequency. Because the ultrasonic frequencies are so low at the 48 kHz sample rate, the difference between it and 44.1 kHz is probably rarely if ever audible, and may be offset by 48 kHz having a more room for it’s antialiasing filter. 

You may think some frequencies that appear in the above example like 27 kHz and 33 kHz aren’t relevant since you can’t hear them, until you realize that at the next round of intermodulation distortion 33 kHz - 27 kHz produces extra distortion at 6 kHz. Oversampling plugins minimize all this intermodulation distortion buildup because they lowpass out ultrasonic frequencies after each process. The buildup at higher sampling rates may not be very significant if you aren’t hitting saturation and dynamics plugins very hard. And as I hypothesized earlier, some people might actually like the build up because a lot of it broadband at very low amplitude may sound more like a cushiony noise floor. At higher amplitudes though it can sound like harsh/brittle dissonant distortion. 

Here are some examples of what obvious intermodulation distortion sounds like:


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## Nick Batzdorf

Thanks. Yes, I do know what IM distortion is, but that still doesn't clear up what I'm asking.

My question was about recording ("Shouldn't those freqs - *including difference signals* - get filtered out before they're recorded?"). But the same thing applies to 96k processing, and you mentioned oversampling filters taking care of that.

At this point I'm becoming less Socratic and more Pyrrhonist.


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## sumskilz

Nick Batzdorf said:


> Thanks. Yes, I do know what IM distortion is, but that still doesn't clear up what I'm asking.
> 
> My question was about recording ("Shouldn't those freqs - *including difference signals* - get filtered out before they're recorded?"). But the same thing applies to 96k processing, and you mentioned oversampling filters taking care of that.
> 
> At this point I'm becoming less Socratic and more Pyrrhonist.


With present technology, there is no way to selectively filter out only IMD (to my knowledge). You can only band limit to prevent unnecessary frequencies from multiplying the amount of IMD that occurs. More IMD occurs at every stage of nonlinear processing, including playback. You could band limit your converter at the recording stage so that it rolls off everything above the audible spectrum, but then it would still accumulate with processing. At that point you may as well be recording at 44.1k anyway (in most cases). But say you have some plugins you really like other than the fact that they aren't optimized for lower sample rates, you could work at 96k and simply place a plugin that filters out ultrasonic frequencies before and after every nonlinear process, and then you would achieve IMD on par with the lower sample rates. Tokio Dawn Labs actually created a free plugin for this purpose: https://vladgsound.wordpress.com/2014/12/21/tdr-ultrasonic-filter-alpha-version/

You could additionally (and maybe this is along the lines of what you were thinking), place an analog ultrasonic filter prior to conversion. Seems more trouble than it's worth. Obviously plenty of people work at 88.2k and 96k and get good results without worrying about extra IMD (if they are even aware of it). The empirical evidence suggests even expert listeners with high end monitoring equipment usually can't hear the difference. Although if you use a lot of dynamics processing and/or saturation, you're more likely to have IMD build up to the point its audible. Really fast attack times on compressors and limiters produce a lot of it. That's probably why the earlier analog models of 1176s could never seem to compress as fast as the real ones, I assume that developers just hadn't figured out how to do it yet without it getting ugly.

It may also be worth mentioning at this point that some of what I've said wasn't always the case. For example, it used to be that many converters performed better at higher sample rates simply because it was easier to design them that way at a reasonable price point. The technology has advanced to the point that today prosumer (maybe even budget) converters can usually do any common sample rate well.


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## Nick Batzdorf

sumskilz said:


> You can only band limit to prevent unnecessary frequencies from multiplying the amount of IMD that occurs



Right, and I've always assumed that's what happens with decent plug-ins, pretty much the same as with oversampling filters.

That might be what I'm missing.


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## sumskilz

Nick Batzdorf said:


> Right, and I've always assumed that's what happens with decent plug-ins, pretty much the same as with oversampling filters.


Ah yeah, I know for certain that Universal Audio does: https://www.soundonsound.com/sound-...strict-processing-bandwith-their-uad-plug-ins


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## Dewdman42

I understood the situation to be that ultrasonic frequencies can cause more Inter modulation distortion to occur, some of it occurring in lower audible frequencies. So you are adding audible Im distortion in order to process inaudible ultrasonic high end. once it’s in the audible range how do you filter it out?

you could band limit the audio before hitting the non linear plugin but then you might as well have been recording at a lower sample rate to begin with.

there will always be some IM present, but using a higher sample rate just multiplies how much of it, and some of it will be audible.


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## Nick Batzdorf

Dewdman42 said:


> I understood the situation to be that ultrasonic frequencies can cause more Inter modulation distortion to occur, some of it occurring in lower audible frequencies.



I think you get this, but if you combine - intermodulate - two freqs, you get two more: one at the sum and one at the difference. So two freqs above human hearing can combine and produce difference freqs within the audible range.

The word "distortion" is a little confusing, because it makes it sound like something you never want, when in fact the combination is often want you do want. Normally the difference freqs in the audible range are in the sound already - i.e. they're recorded.

sumskilz says that sum signals from two sounds in the audible range can combine into the supersonic range and produce difference freqs in the audible range that are -30 (minus 30 below what?), but the amount of power in supersonic freqs is really low to start with.

Anyway, good audio algorithm designers know what they're doing, and I doubt this is an issue to lose sleep over.


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## Dewdman42

I agree but I’m also using 48k for the mere fact I don’t want all the overhead for marginal if any difference. 

There are so many other ways to mangle the sound in good and bad ways, for me the sample rate is the least of my concerns. I choose 48k because anything I do is likely to either be in video at 48k or more likely encoded into lossy compression anyway at this point before anyone other then me ever hears it. I will almost certainly not be producing a cd at 44.1k. So 48 it is, zero concerns


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## Fredeke

Nick Batzdorf said:


> Or are you saying you can use the end of the recording at a more exact loop point at a higher SR? I'd have to hear an example of that making a difference, but I still don't understand how it would - because a sine wave below Nyquist is a sine wave below Nyquist.
> 
> Someone mansplain me what I'm missing.



I'll try to explain this again, to the best of my manly abilities.

First, the situations where this makes a difference for me are when I try to loop the sustain part of a sample.
Two situations come to mind:

1- Long loops (several wave cycles to several minutes) :

If the looping creates a "step" in an otherwise smooth waveform, you will hear a click. Nyquist theory doesn't matter, audible range vs ultrasonics don't matter - it is not about those. It is about a click (aka "step") in the wave, and a click is a click, period. Sometimes crossfading doesn't do it for me, at least not as well as having loop end and start sample values match exactly to begin with, which you'll get more chances of getting right if you split the wave into finer slices.

2- Short loops (one to a couple of wave cycles) :

Loop length is obviously quantized to the sample length (like 1/48000 sec. at 48KHz). With high-pitched notes, sometimes none of two consecutive possible loop lengths will sound in tune: that means the loop length is not close enough to a multiple of the note's period. No amount of crossfading will correct that, since crossfading doesn't affects the loop's length. Some sampler give you the ability to fine tune the loop relatvely to the non-looped part, and that works fine. In other cases, a higher sample rate will help, because it quantizes loop length more finely (you get steps of, say, 1/192000 sec. instead of 1/48000 - that's 4x more precise). Here again, it is not a matter of Nyquist theorem or ultrasonics magic - in fact, it is not a matter of harmonics: it is a matter of precise tuning of the fundamental, to which our ears are very sensitive, especially near the middle of the audible spectrum, where the fundamentals of high-pitched notes fall.

Maybe the reason I run into these issues more that the other guy, is that I mainly sample synthesizers. They can (arguably) generate a purer, more stable sound than acoustic instruments, making this kind of imperfections stand out more.

Now, of course it can be debated whether the increased memory takeup is worth this solution.

But you don't need to run your DAW and converters to a high sampling rate in order to play high rate samples, in any case. All this happens whithin the pitching algo of your sampler, which by definition adapts the sample's rate to the DAW/converters' rate anyway (so it's not even more work for it).

And BTW, if we stick to the issues I mention here, you don't even need to sample at a high rate. Upsampled samples would work just as well. Because as I said, it's not about ultrasonic content.


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## barteredbride

So... when companies advertise they recorded their violins at 96k...it's all just a load of marketing rubbish and the samples might actually sound worse than if recorded at 48k ??

So what about 192k ?? Why do people even use this for?? Making an album of waltzes for bats??


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## JohnG

Interesting debate. For sure, one thing: if you have a bunch of 48k or 44.1k samples, there is vanishingly small benefit to running at 96k -- I say 'small' only to acknowledge those who argue that you may have processing taking place at a higher sampling rate. I think it's a total waste of overhead, storage, processing, brain damage, etc. for samples-only productions.

What puzzles me about the posts above that argue 96k is worse, is that nearly every 'A level' film score and album that records live players is running at 96k.

How come?



barteredbride said:


> So what about 192k ?? Why do people even use this for?? Making an album of waltzes for bats??



Dan Lavry argues that above 96k you not only don't get any benefit, it's actually 'worse.' Check out lavryengineering dot-com


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## barteredbride

JohnG said:


> What puzzles me about the posts above that argue 96k is worse, is that nearly every 'A level' film score and album that records live players is running at 96k.
> 
> How come?



Yes its a strange one! Maybe because the sound systems of modern cinemas might benefit from this??


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## JohnG

barteredbride said:


> Yes its a strange one! Maybe because the sound systems of modern cinemas might benefit from this??



IDK. The engineers say you record and mix at 96 and it's better, but I've recorded plenty of material that went on the air and in theatres at 48. Just don't tell anyone....


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## Nick Batzdorf

Fredeke said:


> If the looping creates a "step" in an otherwise smooth waveform, you will hear a click



That's not a sample rate issue. You can easily get clicks at zero crossings.



JohnG said:


> Dan Lavry argues that above 96k you not only don't get any benefit, it's actually 'worse.' Check out lavryengineering dot-com



And now his son has taken up the argument for him. 

Not directed at John: there aren't "more points" to represent the waveform. It's all sine waves.

The technical argument for higher sample rates is only that it puts the brick wall filter ringing out of the audible spectrum.

Remember, a speaker can only go backward or forward.


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## Dietz

barteredbride said:


> Yes its a strange one! Maybe because the sound systems of modern cinemas might benefit from this??


Good point! Following discussions of Dan Lavry with people Bob Katz, Bob Ohlsen, Glenn Meadows et.al. on the legacy pro.audio mailing-list it's plausible that _converters_ might sound better at higher SR due to an interplay of technical parameters. It not necessary to run the production at that frequency*), though - it's enough to upsample the audio stream before replay (... assuming that the SRC process isn't flawed).

_*) ... like pointed out above, there's evidence that sample rates above roughly 60 kHz do more harm to PCM audio than they do good. This, and higher CPU-load, higher RAM consumption, higher storage size ..._


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## sumskilz

barteredbride said:


> So... when companies advertise they recorded their violins at 96k...it's all just a load of marketing rubbish and *the samples might actually sound worse than if recorded at 48k* ??
> 
> So what about 192k ?? Why do people even use this for?? Making an album of waltzes for bats??


Violin samples won't sound worse because there is no intermodulation distortion without harmony.



JohnG said:


> What puzzles me about the posts above that argue 96k is worse, is that nearly every 'A level' film score and album that records live players is running at 96k.
> 
> How come?


Objectively, 96k has more inharmonic distortion within the audible range due to fact that it has greater IMD, but there is also a trade off in that non-linear processing at 96k produces less aliasing. There is widespread ignorance regarding the former fact among professionals in my experience, and the latter used to be much more of an issue than it is today. Over the period that 96k became the new high end standard, anti-aliasing filters were less effective and oversampling was rare, so the balance between the trade-offs was different than it is today.

Many say 96k sounds better to them, but the blind tests in the study referenced earlier suggest the difference isn't always audible even to audio professionals under the best listening circumstances. When it was audible, most believed the lower sample rate was the higher one. It's fair to conclude that the difference is so negligible under most circumstances, that it doesn't really matter whether you use 48k or 96k, so I would argue it best to work at the whatever sample rate you need to deliver at.

One other factor, 96k gives you about one millisecond lower latency.


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## barteredbride

Hi @Dietz !

I bow to your amazing knowledge about sampling! 

On the Synchron Stage website, it states: "The network is able to operate at 192kHz throughout, with the ability to incorporate lower sample rates as required".

So in what situations would you use 96k and 192k sampling? Is this purely because you can have lower latency for musicians headphones?

I know @Rctec recorded the Netflix series The Crown and BBC´s Blue Planet at the Synchron Stage, I wonder if networks insist upon recording at super-high sample rates?


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## Dietz

barteredbride said:


> Hi @Dietz !
> 
> I bow to your amazing knowledge about sampling!



Huh! *blush* That's really nice to hear, but like I wrote before: I'm just referring to the info shared by The Man Himself, Mr. Dan Lavry, who is regarded as one of the most knowledgeable persons in this realm. 



barteredbride said:


> On the Synchron Stage website, it states: "The network is able to operate at 192kHz throughout, with the ability to incorporate lower sample rates as required".
> 
> So in what situations would you use 96k and 192k sampling? Is this purely because you can have lower latency for musicians headphones?
> 
> I know @Rctec recorded the Netflix series The Crown there and BBC´s Blue Planet, I wonder if networks insist upon recording at super-high sample rates?



I'm not involved a lot with the work that goes on at Synchron Stage Vienna, so I can't give you first-hand information in that respect. 192 kHz are offered "because they can", but personally I would never record at that rate. 8-)

What I can say for sure is that Vienna Instruments are recorded, edited and stored in 96 kHz and downsampled to 44.1 just before the mapping takes place - using a proprietary in-house SRC tool that offers several nifty tricks and clever solutions.


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## barteredbride

Dietz said:


> What I can say for sure is that Vienna Instruments are recorded, edited and stored in 96 kHz and downsampled to 44.1 just before the mapping takes place - using a proprietary in-house SRC tool that offers several nifty tricks and clever solutions.



Is the downsampling done to save on diskspace for users? (plus other things mentioned in this thread?)

VSL has come a long way since the Horizon/Gigastudio days! Do you think as computers become yet more powerful and with bigger diskspace, developers will release 96k sample libraries? 

I know Spitfire record at 96k also.


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## Fredeke

barteredbride said:


> So... when companies advertise they recorded their violins at 96k...it's all just a load of marketing rubbish and the samples might actually sound worse than if recorded at 48k ??
> 
> So what about 192k ?? Why do people even use this for?? Making an album of waltzes for bats??


I seem to understand the degradation happens at playback, if the sound system can't deal with ultrasounds and distrorts them into the audible range. That's why it's not a great idea to provide 96k to the consumer.

Hopefully, when pros record at 96k/192k, their whole signal chain can manage the ultrasounds. Not sure it's useful in all cases, but at least it doesn't hurt (except the wallet when buying more SSDs)



Nick Batzdorf said:


> That's not a sample rate issue. You can easily get clicks at zero crossings.



I wasn't making an exclusive argument. I'm just saying sometimes high sample rates help.
And yeah, I know, some other times there's nothing to do. Or a crossfade might be preferable.
Increasing the sample rate is just one more trick in my bag.


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## Dietz

barteredbride said:


> Is the downsampling done to save on diskspace for users? (plus other things mentioned in this thread?)



Disk space, project loading time, RAM and CPU consumption, download bandwidth and other distribution-related issues ... 96 kHz has little benefit to offer, especially when talking about samples. If you need that SR (or any other), eg. to achieve shorter latency delays, then just upsample in your DAW.



> VSL has come a long way since the Horizon/Gigastudio days! Do you think as computers become yet more powerful and with bigger diskspace, developers will release 96k sample libraries?
> 
> I know Spitfire record at 96k also.


I'm an audio guy, and I have little insight when it comes to marketing decisions. Maybe "the market" demands features like these, like many others that weren't suggested by engineers but by people who know how nto sell stuff. 

Like mentioned before, I have yet to stumble across a session which sounds bad _only_ due to (say) 48 kHz SR, and I can assure you that the differences I achieve by changing the gain of a _single_ EQ band in a mix are fundamentally more noticeable than the differences between 48 kHz and 192. ... but then, that's just the opinion of one old dog with 35 years of audio engineering in his CV.


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## ceemusic

You can record at any bit rate, 96k is becoming more the norm these days.


It's delivery with music distributors that mainly request 44.1khz ( & of course if you're releasing CD's it's mandatory.)


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## Nick Batzdorf

Dietz said:


> Disk space, project loading time, RAM and CPU consumption, download bandwidth and other distribution-related issues ... 96 kHz has little benefit to offer, especially when talking about samples.



Put another way, your computer resources are almost halved! And for what?



> If you need that SR (or any other), eg. to achieve shorter latency delays, then just upsample in your DAW.



If that little latency is noticeable to you - and there are people who actually do notice the 3ms path through converters, not including me - then I'd say you need to use an analog direct monitoring path, in which the latency is on the Planck scale.

(Actually I think it's more than that, but so what.  )


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