48000Hz vs 192000Hz aka What The Heck Is Going On?

[TFHT]

It looks like your tests are pretty meaningless as your soundcard can't handle more than a 48kHz sampling rate.

Is a test really meaningless if it is educational? Could you please expand on this response, and perhaps answer the other questions I asked you? Judging from your signature, you seem like you are the guy to ask.

SOMETHING is happening hear, and I don't know what it is. If my soundcard cant read 192000Hz, shouldn't it just NOT play that file at all? Is it dropping the Sample Rate to 48000 automatically?

I'm having a hard time finding out exactly what Realtek soundcard model I have, but when I check the Realtek HD Audio Manager, select the "Speakers" tabs, and click on "Default Format," I can choose from 16 Bit, 44100Hz, to 24 Bit, 192000HZ, however, on "Stereo Mix" I only have two options - 16 Bit, 44100HZ and 16 Bit, 48000Hz. Since my Acer Aspire 5553g didn't come with any other cables or external hardware for audio, it SOUNDS like I SHOULD be able to properly render in 192000Hz. But, I'm the newb, so you guys tell me.

Can I turn off aliasing?

AND, since I used Dblue Glitch in the test song to purposely lower the sound quality to something like a video game, might that be a culprit as well? One of the options in the VST is the ability to reduce Bit Depth...

[hollowsun]

It's my opinion that you over-thinking this and taking it all a bit too seriously.

Your Realtek soundcard seems, from what I know, a consumer thing for average audio listening (i.e. MP3 listening). It is by no means a 'professional' audio I/O. To think superlative results are possible is, I think, a little naive ... but I appreciate that you're new to all this and yes, it CAN be damned confusing sometimes.

The Acer you have is a fine little thing for general computing, gaming, music listening, video/Netflix/YouTube viewing but I'd be loathe to use it for any serious recording, especially without some quality external audio I/O interface.

I hate to bring the news to you but having a laptop does not necessarily give you a recording studio. And sorry, 192kHz recording is total arsewash - 44.1/16-bit is perfectly adequate. At a push, 44.1/24-bit is (arguably) better but this will have to be dithered down to 16-bit for CD release and if your target market is an MP3 release, 192kHz really is overkill ... or, to use a technical term, horsesh!t with more than a hint of "The Emperor's New Clothes" about it.

Sorry to be blunt

[Richie Royale]

It should be noted that the main difference he appears to notice is with a plug in instrument, which desmond has already explained about; the discussion about the soundcard seems to be a bit of a red herring as I can't see any reference to recording the guitar at various rates or any improvement regarding it.

there was a HUGE difference between 48000 and 192000 Hz. Zebra 2 sounded amazing at 192000Hz - very full and sounded like it was resonating. The rest of the song sounded even in volume, but slightly muffled / less loud as a whole. Zebra 2 sounded louder at 48000Hz, but sounded brittle compared to 192000Hz. The rest of the song at 48000Hz sounded louder than at 192000, but it was very uneven mixing.

[James Perrett]

Is it dropping the Sample Rate to 48000 automatically?

That's the problem with many consumer audio chips - they do all kinds of conversions without telling the end user. It all started with Creative Labs Audigy range around 20 years ago which resampled everything to 48kHz and, unfortunately, resampling at that time wasn't particularly good. Other consumer audio manufacturers followed this lead so it would appear that resampling is pretty much the norm in these chips. A decent audio interface will either not resample at all or, if it does, it will use a much higher quality algorithm.

Did you follow the link to Dan Lavry's paper that I posted? He also has an introduction to digital audio on his site which would be worth reading - as would anything by Hugh Robjohns on this site. Unfortunately Hugh is on holiday this week, otherwise I'm sure he would have answered your questions much more fully than most of us can.

[Tony O'Shea]

Hi,
Just to second James' recommendation of Dan Lavry's paper. It is well worth reading particularly in the context of this debate.
Best,
Tony

edit - opps, apostrophe in the wrong place.

[Goddard]

It's my opinion that you over-thinking this and taking it all a bit too seriously.

Your Realtek soundcard seems, from what I know, a consumer thing for average audio listening (i.e. MP3 listening). It is by no means a 'professional' audio I/O. To think superlative results are possible is, I think, a little naive ... but I appreciate that you're new to all this and yes, it CAN be damned confusing sometimes.

The Acer you have is a fine little thing for general computing, gaming, music listening, video/Netflix/YouTube viewing but I'd be loathe to use it for any serious recording, especially without some quality external audio I/O interface.

I hate to bring the news to you but having a laptop does not necessarily give you a recording studio. And sorry, 192kHz recording is total arsewash - 44.1/16-bit is perfectly adequate. At a push, 44.1/24-bit is (arguably) better but this will have to be dithered down to 16-bit for CD release and if your target market is an MP3 release, 192kHz really is overkill ... or, to use a technical term, horsesh!t with more than a hint of "The Emperor's New Clothes" about it.

Sorry to be blunt

Just for the record, support for multi-channel 192kHz/24-bit audio is in fact required for playback of a number of defined Blu-Ray primary audio stream formats (and hence, for the recording and production of same). Which probably explains why it was also made part of Intel's High Def Audio spec (and thus, why onboard HDA codec chips like Realtek's include it) as well as why it was added in the USB audio class type 2 spec.

[largon203]

dual Mono Block BAT amps and a high end NAD DAC set to 192

I can maybe see why they're called "bat" amps.

BAT is an acronym for Balanced Audio Technology. Google it...

-Dave.

[largon203]

When you get a chance to listen to a proper demo on an Audiophile system that is tuned properly costing upwards around 1/4+ million USD you let me know what you experienced.

Dave - I'm afraid that audiophile systems don't have a good reputation around here. I've heard expensive audiophile systems that sound worse than my 80 quid kitchen stereo. In my experience audiophiles seem to like "nice" sound rather than accurate sound and audiophile systems often include deliberately introduced distortion to create this "nice" sound.

And anyone advocating 192kHz sampling should read this white paper from Dan Lavry before going any further.

Apparently the "Audiophile" systems you listened to were not properly setup or just a poorly matched set of equipment. My father specializes in the setup of such systems he has a process that includes several mods that remove/lower the noise floor by 60+ or should I say - dec. I'm sure your experience comes from listening to systems setup by wannabe audiophiles. The gentlemen I know and respect, recognize that All audio equipment has color and they generally go for the least intrusive. FYI even your studio monitors have color / distortion... It's unavoidable. We just look for equipment that we can work with. It took me about 6 years figure out my Hafler TRMs and an additional 5 to get good with them. I'm just not at about 14 years with them starting to make some really great mixes. They are very colored but I know how they translate. I LOVE my fathers Beauhorns that incorporate an ultra efficient single louther driver and a wooden knob mounted in it to disperse the high frequencies properly in a folding horn designed box. Lots of geek audiophile gear that most of us mortals simply can't justify spending the $$££ on. I assure you his system is WAY more accurate than your whatever quid kitchen stereo. LMAO.... No disrespect intended.

-Dave

[ernzo]

-What about the Nyquist theorem?

i'm not an expert by any means, but it seems that samplerate could be a bit like framerate..

If higher samplerates (192khz) mean more frames of captured audio per second, not only could it mean being able to record higher frequencyes, but maybe also record "normal" or all frequencies with more detail?

more precision in time and more samples of it..

as music made by real instruments is of very complex and dynamic harmonic nature, the faster and bigger number of samples can only be good.. and even better being able to record a wider spectrum of sound freqs..

anyone can elaborate on this?

-I never saw the need to go higher than 88.2khz tho (wich resamples better to 44.1khz than 192khz), and most of what I do is made at 44.1 from the start... only for acoustic stuff I might try 48khz or 88.2 if it was a super production

-Whats much much more noticeable to me is the change from 32 to 16 bit... it's like night and day, I woud never mix at any less than 32bit

[largon203]

And anyone advocating 192kHz sampling should read this white paper from Dan Lavry before going any further.

Not saying I advocate 192 but I read that whole article and he does a great job of maki g a mountain out of a molehill. I tell you I almost disregarded the whole article after busting out laughing when I got to the end of the very first paragraph. I did Finnish the article for entertainment value though. While this very articulate scientifically thought out observation seems sound it is filled with a load of bunk. To say a higher sampling rate is going to capture too much info is like saying vinyl record players provided too much info. C'mon man! If there is no sound up at 180k and recording up to 192 you are recording nothing. This guy implies you are recording BAD frequencies. He has clearly way over thought this. Ha ha ha. Makes for a good read though if you can stomach his blathering. Sorry if that offends mate but I couldn't disagree more. I'll end with I never record higher than 44.1 NOT because its better than 48 - 192 but because when you compress heavily as I sometimes do it becomes more difficult to control what happens to the sound IMO. That is not to imply that Bad sound gets recorded at higher frequencies. It's important sound IF your going for natural with very little if any compression. Such as a classical piano or a stereo mic'ed symphony.

-Dave

[largon203]

I've been hearing a lot of things about Sample Rate and Bit Depth, and the controversy that ensues when they are talked about.

So, I made a little test song to try out different Bit Depths and Sample Rates and see how they affect my music.

I made a little test track in FL Studio. ASIO4ALL was selected, I recorded an acoustic guitar part with my Microphone on channel 10. Turned off the mic, sent the audio file to that mixer channel, added reverb, chorus, an instance of Dblue Glitch, and an instance of mda Tube Screamer. I also sent the stock FL Studio snare and hat to that channel, kept the kick on channel one, added a compressor to the master channel. I also had 3XOSC on bass, not sent to the mixer, and a many octave spanning C chord in an instance of Zebra 2, also not assigned to a mixer track.

Basically the massive chord plays first, fading into the drums, bass, and guitar immediately.

I made 6 renders, .wavs, 3 in 32 Bit, and 3 in 24 Bit. Each bit depth file had a different sample rate - 22050, 48000, and 192,000 in that order.

For some reason, the 2 wav files at 22050 Hz didn't even play in Foobar or Winamp. The others played fine, and I found that while I could not discern an immediate difference between the two Bit Depths with my headphones, there was a HUGE difference between 48000 and 192000 Hz. Zebra 2 sounded amazing at 192000Hz - very full and sounded like it was resonating. The rest of the song sounded even in volume, but slightly muffled / less loud as a whole. Zebra 2 sounded louder at 48000Hz, but sounded brittle compared to 192000Hz. The rest of the song at 48000Hz sounded louder than at 192000, but it was very uneven mixing.

WHAT IS GOING ON HERE?

Are there parts that simply sound better with one sample rate than the other, and vice verse? Is this a mixing issue? Do some sounds and effects lack and ability to be properly rendered in certain formats?

Considering the test song, I want the loudness of the guitar and drums from the render at 48000Hz, but I want that awesome resonating effect from Zebra at 192000Hz? How can I get the best of both worlds?

I, like many of you, want the best possible sound quality for my finished products, so that those with the ability to hear them in the best light actually can. Considering this, if my music is eventually put on a CD, does it even matter if the song is 192000Hz, when the Sample Rate is reduced once it goes to CD?

Thank you all in advance for your help!

Although this has sparked a pretty big side debate. I agree with most of the others co fear ing your situation. So long as your recording g at 24 it or 32 bit-rate the sampling g frequency should be kept at 44k to 48k range arguably easier for you to mix and fir your computers CPU to handle as well at 44.1 for what it's worth.

-Dave

[Richie Royale]

Dave 'largon' I think Mr Lavry knows his subject very well with over 30 years of expereince and 20 years in the digital world.

http://www.lavryengineering.com/about.html

[zenguitar]

I just wish I could see Hugh's face when he returns from his holidays, catches up with all the activity on the forums, and reads this thread.

Andy

[James Perrett]

I assure you his system is WAY more accurate than your whatever quid kitchen stereo.

I would hope so too - but sadly many audiophile sound systems have taken huge leaps backwards in the time since I was last involved in that field.

I don't think you do your argument any favours by dismissing Mr Lavry's article either. I saw the stringent review process that it went through before it was published and if there were any inaccuracies in that article they would certainly have been picked up.

[muzines]

i'm not an expert by any means, but it seems that samplerate could be a bit like framerate..

If higher samplerates (192khz) mean more frames of captured audio per second, not only could it mean being able to record higher frequencyes, but maybe also record "normal" or all frequencies with more detail?

more precision in time and more samples of it..

No, this is a common misconception due to not understanding how audio works. It doesn't work like that.

Try this for a nice explanation:
http://createdigitalmusic.com/2013/07/video-explains-why-dif...

In short, 44.1KHz is perfectly capable of completely reproducing all frequencies up to Nyquist (half the sample rate) - more samples does not equal "more detail".

[dmills]

Next we will be seeing the 'R' word being applied to audio.... Video is sampled at way below the Nyquest limit, hence wagon wheels in old movies spinning backwards, plaid shirts looking awful on analogue telly and such, audio is not sampled this way and higher sample rates gain you nothing within the original passband.

As long as you limit the bandwidth before the sampler to strictly less then 1/2 the sample rate you capture everything that is there, no ifs, no buts.

Now there are issues with that statement, particularly with the required filtering and its resulting phase response, but that is more a small matter of engineering then a fundamental limitation.

And yes, having more bandwidth in a system then you need to obtain a flat group delay over the band of interest is a bad thing, it makes your electronics more susceptible to noise pickup, and potentially more prone to take off somewhere up above the audio band which will do your tweeters a world of good, it also wastes headroom amplifying stuff that is out of band (Usually more of a problem with excessive LF extension).
It also potentially passes energy above the useful band to transducers which are usually not particularly linear up there, this can give rise to IMD products within the audio band even if the original signals are above the audible range, DC to daylight is not a figure of merit in an audio reproduction chain.
This can actually be a problem with microphones, some that make a feature out of a HF response to many tens of kHz can actually produce difference frequency tones in the output when driven by a reasonable level of two tone ultrasonic power, what this does to a close miced recording of a violin I will leave as an exercise for the reader......

I couldn't find any proper documentation on those BAT amps (Think schematics), but I would regard a 250kHz bandwidth as excessive and as being an excuse to end up amplifying every smpsu in the area. I am also somewhat suspicious of 'zero NFB' as a claim, they might have no overall negative feedback, but there will be at least the intrinsic mosfet source resistance (And probably an external degeneration resistor) providing local feedback, then you have the Miller capacitance.....

Regards, Dan.

[Elephone]

I just wish I could see Hugh's face when he returns from his holidays, catches up with all the activity on the forums, and reads this thread.

Andy

Ha-ha! I was scanning for a response from Hugh when I read your comment.

I can see the purpose of much higher sample rates for capturing ultra high frequencies (e.g. tiny insects) with a 50kHz mic, but only for analysis or maybe to slow them down to human hearing. Otherwise, if there is anything audible with 192KHz, I can only think it's to do with the way digital processors are dealing with it...?

[dmills]

Yup, aliasing in the digital domain is a common issue in poorly written soft synths and dynamics plugins, also half band decimators in delta sigma converters can be an issue if not handled correctly.

When you do it wrong, it does not work properly, no surprise there!

Regards, Dan.

[TFHT]

It's my opinion that you over-thinking this and taking it all a bit too seriously.

Your Realtek soundcard seems, from what I know, a consumer thing for average audio listening (i.e. MP3 listening). It is by no means a 'professional' audio I/O. To think superlative results are possible is, I think, a little naive ... but I appreciate that you're new to all this and yes, it CAN be damned confusing sometimes.

The Acer you have is a fine little thing for general computing, gaming, music listening, video/Netflix/YouTube viewing but I'd be loathe to use it for any serious recording, especially without some quality external audio I/O interface.

I hate to bring the news to you but having a laptop does not necessarily give you a recording studio. And sorry, 192kHz recording is total arsewash - 44.1/16-bit is perfectly adequate. At a push, 44.1/24-bit is (arguably) better but this will have to be dithered down to 16-bit for CD release and if your target market is an MP3 release, 192kHz really is overkill ... or, to use a technical term, horsesh!t with more than a hint of "The Emperor's New Clothes" about it.

Sorry to be blunt

Well, I may not be able to afford a better computer, but I can afford an external soundcard / audio interface, given that even the cheap ones now seems loads better than my current soundcard.

But, I never expected superlative results from my gear, nor do I think I have a recording studio. I just want to see where my gear's limits peak, and to map out my capabilities so I know what to immediately upgrade and what not to. This thread is doing exactly that.

If I, or anyone waited until they had superlative gear to start making music, I think there would be far less music in general, and far less quality music. You gotta start somewhere. If Eric Clapton waited until he had a $10,000 guitar to learn how to play, instead of playing on a crap cigar-box guitar, we wouldn't know who Mr.Clapton was because he would have probably starting playing at 30 instead of 10. I'm no Clapton, and I don't expect to appease the Claptons of the world just yet. I'd rather push my crap gear to its limits and make good music than have the best gear in the world and not know how to use it.

Gotta start somewhere.

[ernzo]

In short, 44.1KHz is perfectly capable of completely reproducing all frequencies up to Nyquist (half the sample rate) - more samples does not equal "more detail".

Great explanation thanks!

I would like to see the same done with real complex audio tho..

Is this true regardles of source complexity?

I mean what if you have a super beautifully complex sounding guitar, violin and piano, playing complex harmonies alla Piazzolla?

Will all those harmonics, supra/sub harmonics and the harmonic relationships of and between every instrument, go thru completely clear regardless of samplerate? (up to nyquist limit, and considering ideal recording conditions/method..)

[dmills]

Yes, **ANY** signal that is band limited to a bandwidth strictly less then half the sample rate can be captured entirely by sampling at that rate.

Consider that any signal can be broken down into a (generally infinite) set of summed sine waves (Not going to demonstrate this without math rendering, but it is VERY well established fact), and that as any sine wave can be fully characterized by at least 3 points over the course of a cycle, then it follows that any sum of sine waves can be fully characterized by measuring the instantaneous value of the sum at a rate greater then twice the bandwidth (Hence ensuring the three points thing for the highest frequency). Addition is after all a linear operation.

44.1k is entirely sufficient for **ANY** signal band limited to DC-(just under) 22.05khz.
Practically of course the filters at both ends are not perfect and leaving a little space for a transition band does not hurt, but in principle....

Music is magic, audio is engineering, and should be treated as such.

Regards, Dan.

[Exalted Wombat]

In short, 44.1KHz is perfectly capable of completely reproducing all frequencies up to Nyquist (half the sample rate) - more samples does not equal "more detail".

Great explanation thanks!

I would like to see the same done with real complex audio tho..

Is this true regardles of source complexity?

I mean what if you have a super beautifully complex sounding guitar, violin and piano, playing complex harmonies alla Piazzolla?

Will all those harmonics, supra/sub harmonics and the harmonic relationships of and between every instrument, go thru completely clear regardless of samplerate? (up to nyquist limit, and considering ideal recording conditions/method..)

Yes, all music - by definition - adds up to a mixture of audible frequencies (do we need to argue over how you "hear" particularly low frequencies?). Sample at twice the highest your ear can make out (44.1KHz is plenty) and you're good.

This is not particularly intuitive. It's easy to convince yourself that complex music must contain "more frequencies than you can hear". But think about it!

The system relies on effecient filtering at half the sample rate, free of side-effects. We're pretty good at doing that now.

[Goddard]

Sampling rate discussions like this thread are so fun. Really takes me back... to about 15 years ago.

[Folderol]

Well that was a fun read, and I think just underlines that when someone says they can hear/see/smell/feel something, you are wasting your breath trying to explain why they can't.

[Exalted Wombat]

Very true however on those high end systems I'm referring to your not hearing higher frequencies as much as your now hearing things more linearly. What's going on in those stratosphere frequencies is pretty important. To be a little poetic... It's the sonic glue that holds the whole thing (on a really interesting level) together. After you've heard 192 that was recorded at 192 on a high end audiophile system and then you hear the same track at 44.1 it becomes obvious what your missing. The 44.1 track sounds ailiased (is that a word? My spell check doesn't think so) well regardless I'm not arguing with you because your right except for a few random lucky people and dogs most of us can't hear higher than around 19-20k but again it's not that your hearing higher frequencies but that due to the ultra high sample rate those 15k-20k we can hear are WAY better represented with the overtone structure that is there, albeit that you can't hear even at a live performance, but that you can feel.

Eh? Want to try that again, in English, with your brain turned on?