PonoMusic player by Neil Young

[linux-works]

The Kickstarter clearly shows the specs as being 192KHz / 24 bits, they even give a lovely comparison of recording qualities:

Which is a bit silly for listening purposes. For one, the human ear doesn't go that far up. Then, you will surely get problems with winding a suitable speaker to reproduce it (even an earphone-speaker). The amplifier might want to have a word here as well. Oh, and please show me the equipment used in recording that can actually make use of that frequency range as well.

you miss the point.  its not about reproducing audio up that far; the main reason why there are high sample rates are:

1) you want to shift the filtering up high enough so that its outside the audio band.  nothing to do with hearing high freqs.

2) on initial capture, you want the highest res samples you can get.  when making finals for USERS, though, its almost always downres'd to 96k or even 88k.  this is generally recognized as the right balance of samplerate for playback vs filesize.

similar to photography: I will capture images at raw but I'll never 'release' raws to end users; they get 8bit jpgs.

[mamalala]

I wish people would quit quoting this guy.  If he actually cared about vorbis quality at the high end, he would have incorporated the aotuv patch a long time ago.

I was not about any vorbis stuff, but more about the general points he makes. Which are quite valid as far as 192kHz/24bit lossless is concerned...

Actually, if you train yourself to identify MP3 compression artifacts (which is a terrible thing to do and will haunt you for years afterwards), this is surprisingly easy.  MP3 is simply not transparent at any bitrate.  AAC is a different story and did become transparent for me somewhere between 192-256 (IIRC, this was quite a while ago).

Yes, MP3 has flaws, no doubt. In fact, any lossy compression scheme is bound to have flaws. But let's not forget that AAC was developed after MP3. So it's quite natural that advances in technology over these years will produce a better algorithm. However, the main point still stands, that a 192 kHz signal is rather detrimental to good quality for the _listener_.

And i think that in this regard he has quite a valid point. And as said, let me see the equipment that is even capable to produce and record signals in that range. An no, i do not mean "hey, here is my 192k soundcard!". Consider the source as well. And the human ear.

Greetings,

Chris

[mamalala]

The Kickstarter clearly shows the specs as being 192KHz / 24 bits, they even give a lovely comparison of recording qualities:

Which is a bit silly for listening purposes. For one, the human ear doesn't go that far up. Then, you will surely get problems with winding a suitable speaker to reproduce it (even an earphone-speaker). The amplifier might want to have a word here as well. Oh, and please show me the equipment used in recording that can actually make use of that frequency range as well.

you miss the point.  its not about reproducing audio up that far; the main reason why there are high sample rates are:

1) you want to shift the filtering up high enough so that its outside the audio band.  nothing to do with hearing high freqs.

2) on initial capture, you want the highest res samples you can get.  when making finals for USERS, though, its almost always downres'd to 96k or even 88k.  this is generally recognized as the right balance of samplerate for playback vs filesize.

similar to photography: I will capture images at raw but I'll never 'release' raws to end users; they get 8bit jpgs.

Uh, well, somehow you actually make my point. Having audio files at 192k makes no sense for the listener. Even 88k would make no sense: you simply are not able to hear a 44k tone anyways. If it has any effect, it would be intermodulation in the amp or (if the amp can handle it) the transducer. And at the end of that chain is the human ear.

I find the argumets for such samplerates especially funny in the cases where people bring up master tapes. Show me an analogue tape recorder that even goes to 30kHz. Which would mean an equivalent of a 60kHz samplerate.

At some point you just record artifacts and noise. You simply can not capture what is not there to capture in the first place. And it makes no sense to capture stuff beyond what is "consumable" by a human either. Going back to your photo example. Hey, why not capture images in the range between _deep_ infrared and _high_ ultraviolet? None  of which a human can see ...

Oh, and what is considered the RAW format on digital cameras is not so much about colour resultion per se, but the way the actual image sensor is constructed and processed. And 8 bit JPG's?

Greetings,

Chris

[linux-works]

88k and 96k do make sense for the listener.

it makes life easier on the dac and whole digital audio system.

its not about the listener; its about implementation.  you should try to get that point.

[mamalala]

88k and 96k do make sense for the listener.

it makes life easier on the dac and whole digital audio system.

its not about the listener; its about implementation.  you should try to get that point.

If you want me to "try to get that point", you better explain what the point actually is. The DAC certainly does not care at which clock frequency it is fed, unless you want to feed it a too high one. And the "digital audio system" ends right at the amp, or the speakers. So, what's your point here? Explain it, substantiate it, and then let's go from there...

Greetings,

Chris

[linux-works]

seek you a few hours at diyaudio.com - there is endless discussion about why higher sample rates give better audio results, pretty much across all types and designs of dac chips.

again, I'm not arguing about human hearing.  mine ends -way- below 20k and that's for sure.  that's not my point.

you can start with why asrc (asynch sample rate converters) often can reduce jitter.  this can have an audible effect; certainly a measurable one.

and as far as raw photo capture is concerned, it is because I get higher bit depth at raw (not to mention there is no processing done) and it makes sense to do your editing at high bit depth and then dither down to lower depth to 'save as' for the end user.  even when I shoot only jpg, I'll import, upres to 16bit/pixel color, do my edits at full math, then convert back down to 8bit/pixel before I save as jpg.  if I need to edit again, I reload the saved 16bit/pixel file, edit, then convert down to 8 again, and so on.  this is not mysterious stuff, its just about doing all your edits using the highest precision you can get (internally), not having rounding error compound so much and then reducing back at the final stage for the end user.  for audio, that would be 16 or 24bit depth and 44k or 88/96k of samplerate.

btw, at 20khz, a square has a lot more components than 20k   if you want to record a lot more -about- the wave, you need more than nyquist 2x.  having more samples can reproduce the -shape-, which is more than just the fundamental.

having more (overkill) means that you are certainly more accurate down where the human cutoff is, at that 20k mark that is so frequently referred to.  2x of 44k is 88k and that's the reason for 88 (88.2, really).  and 2x of 48k is 96k and that's why that is used.  beyond 96k, it does not make sense for the end user and it just wastes file size and adds network and codec load, which I find wasteful

[tjaeger]

It all depends how you look at it. a glance on a tv that shows your pics as a slideshow (as an example) vs 100% zoom on a calibrated monitor where you look closely at every single detail and colour. Same goes for music. most listeners listen audio on relatively low quality speakers/headphones where 128 mp3 sounds pretty good. Hoverer if you let someone to listen the same mp3 on a high end setup - they will notice that it doesn't sound as good as flac.

But only if they expect to. If they aren't aware that there should be a difference, they won't hear one. And if you tell them the mp3 should sound better than the flac, they'll hear that too.

Placebos are great that way.

You are just as wrong as the people who think that expensive power cords improve sound quality.  128 kbit/s MP3s are trivial to ABX on halfway decent equipment.

[mamalala]

having more (overkill) means that you are certainly more accurate down where the human cutoff is, at that 20k mark that is so frequently referred to.  2x of 44k is 88k and that's the reason for 88 (88.2, really).  and 2x of 48k is 96k and that's why that is used.  beyond 96k, it does not make sense for the end user and it just wastes file size and adds network and codec load, which I find wasteful

That's where you are wrong somehow. Look up intermodulation on amplifiers. Pretty much every decent amp low-pass filters the signal. There is absolutely no sense in presenting a much higher bandwidth signal to it. The amp won't handle it, your ears wonÄt handle it, and if introduced, it most likely will make things worse due to IM. Really, it's just a silly audiophool snakeoil thing...

Greetings,

Chris

ETA: Oh, "and 2x of 48k is 96k and that's why that is used"... are you serious with that? Because if so, then you pretty much disqualified yourself from such discussion...

[Rigby]

why don't you explain it instead of just attacking and leaving.  don't do a drive-by.

[linux-works]

That's where you are wrong somehow. Look up intermodulation on amplifiers. Pretty much every decent amp low-pass filters the signal.

define low-pass.  some of the better designs have bw up to 100k.  it has a lot to do with keeping phase; again, not max human hearing range, but phase.

a headphone amp that I run goes well to 100k, but no one is suggesting we hear anywhere near that high.

its fine if you disagree, but don't be so sure of yourself, that's all I'm saying.  a lot of serious designers create high bw amps and preamps. 

[linux-works]

You are just as wrong as the people who think that expensive power cords improve sound quality.  128 kbit/s MP3s are trivial to ABX on halfway decent equipment.

true.  one thing that most people can notice, if you put your attention to it, is fade-ins and fade-outs (such that you'd hear as the song ends).   I've heard a kind of quantization noise as the levels are lower and lower (the recording fades out and pushes the mp3 encoder's limits on how 'blocky' the sound starts to get).  its like watching video that has been compressed too much; as the background turns darker, the blocks get bigger and are not smooth.  you can hear a choppy sound as the music fades to zero with mp3.  with clean flac, you should not hear that.

admitedly, its not a show stopper.  I hear it on quiet piano and classical and rarely on rock or pop.  but its an artifact of mp3 and you can hear it if you pay attention AND if your material has soft dynamics.

[mamalala]

That's where you are wrong somehow. Look up intermodulation on amplifiers. Pretty much every decent amp low-pass filters the signal.

some of the better designs have bw up to 100k.  it has a lot to do with keeping phase; again, not max human hearing range, but phase.

define low-pass.

Are you for real? Or just a lousy attempt at being a troll? "define low-pass" ... Really?

... headphone amp that I run goes well to 100k ...

And the actual headphones connected to that amp go up to what exactly?

its fine if you disagree, but don't be so sure of yourself, that's all I'm saying.  a lot of serious designers create high bw amps and preamps.

And i'm saying that your arguments account to nothing more than pure bullshit. Who are these "serious designers" when it comes to audio amplifiers?

It all boils down to the same thing in the end: if your ear is not capable of registering it, then it's not worth recording it. If that excess information causes potential trouble, you _really_ should not put it out there.

Again, inform yourself about IM, the bandwidth of human hearing, and in consequence what it means to have ultrasonic stuff played back for listening while normal audio frequencies are played back at the same time and are supposed to be listened to.

Greetings,

Chris

[linux-works]

ok, chris.  you know everything.

I'm done with you.  you don't have an open mind and I won't be able to reach you.

[linux-works]

for everyone but chris, I submit these 2 excellent amp designs as examples of high bandwidth (and high upper freq) amps:

the nelson pass f5 (specs: http://www.dms-audio.com/nelson-pass-f5 )

the AMB beta22 ( http://www.amb.org/audio/beta22/specs.html )

both exceed 100k by a large margin.  yes, that is 'higher than human hearing'.  that's not the point, as I've been saying.

those are just 2 examples that popped into my head.  very well respected designs, too.  go ahead, shoot holes in those designs if you think you can.

[mamalala]

ok, chris.  you know everything.

I'm done with you.  you don't have an open mind and I won't be able to reach you.

No, i don't know everything.

And poisoning the well doesnt really work either, especially on a forum where you can expect to have lots of people who understand technology.

Again: Show me what equipmet produces sound in that range at all, and then what equipment is used to record that, and show me the reason for doing all that in the first place, when it all is mixed down and filtered to a 44.1 kHz CD in the end anyways. And if you are unhappy with the CD samplerate, then show me what the benefit of a higher sample rate is.

And no, handwaving away isn't going to help. It's beyond stupid to cough up some illbegotten audio-amp designs that allegedly do 100+ kHz to justify an audio format with, for example, 192 kHz sample rate. If we would follow that logic, hey, lets go 1 GHz sample rate then. After all, we have amplifier stages for that already, used in high end scopes, you know ...

So, put up or shut up. Simple as that. This means: What is the benefit of using a sample rate that results in an audio spectrum far beyond the human hearing range, except for the "can do it" factor? And please, don't even start to think about analogue master tapes or vinyl records...

Greetings,

Chris

[mamalala]

both exceed 100k by a large margin.  yes, that is 'higher than human hearing'.  that's not the point, as I've been saying.

Then what is the point? Just that it can be done? Yea, lots of stuff can do that frequency range. Still doesn't make sense for audio to be listened to by humans...

Again, what is the point of a 100kHz+ amp?

Greetings,

Chris

Edit: This reminds me of the gaming folks who spend boatloads of money on a graphics card, only to proclaim "hey, in game X i get 200 fps now!". Which is pretty useless when the screen they look at has a refresh rate of 60 Hz...

[linux-works]

I won't speak for the designers.  go ask them on their forums.  mr. pass is available on diyaudio and if you post your question about why he has such a wide bandwidth, I'm sure he can give a better answer than I can.  same with the AMB amps.

my specialty is not analog amp design, but I'm simply pointing you at valid, respected designs that go way beyond 20khz.  one reason I was given, when I asked, was that the phase doesn't shift at even 20k, when you have a much higher bw allowed by the amp.

there are also instruments that actually do have useful info into the 30k range.  and pro audio systems can record that high (special ones can).  digital audio at redbook (44.1) cuts off hard at about 20k but with new high res 'file based' audio, you can get downloadable 88k and 96k files and they really could contain some of those upper harmonics.

seriously, go ask papa (nelson, lol) why he designs stuff with such high bandwidth.  if you are serious about your views and you want to know first-hand, GO TO DIYAUDIO and ask!  stop badgering me about this.

[HackedFridgeMagnet]

Edit: This reminds me of the gaming folks who spend boatloads of money on a graphics card, only to proclaim "hey, in game X i get 200 fps now!". Which is pretty useless when the screen they look at has a refresh rate of 60 Hz...

n00b

60 Hz ~ 16milliseconds. Many monitors give below 5ms update time.

The high frame rate gives smaller latency, that is were you get some benefit. Especially if you are having network or other issues.

I wish I had 200fps.

I would rek.

[mamalala]

... that the phase doesn't shift at even 20k, when you have a much higher bw allowed by the amp.

Ahh, there it is, the dreaded phase shift. The audiophools pet-peeve...

there are also instruments that actually do have useful info into the 30k range.

Useful to whom? Bats? Cats? Dogs? Surely not humans, since they simply can't hear that...

and pro audio systems can record that high (special ones can).  digital audio at redbook (44.1) cuts off hard at about 20k but with new high res 'file based' audio, you can get downloadable 88k and 96k files and they really could contain some of those upper harmonics.

Woah! Please, drop some more technical words like "redbook". Makes it sound so much more valid!

Yes, digital pro-audio recording gear can record way beyond 20k for quite some time now. That's nothing new. And what do you know, they have a higher resultion as well, go figure! But then, this is _recording_ stuff. Things that get mixed down later on. And even there, only the higher resolution really makes good sense, and not the higher sample rate as such.

It simply does not matter what some audio file "could" contain. What you can't hear, you can't hear. At best it goes unnoticed. At worst it will introduce IM, and even may stress your amp and speakers. In short: No gain at all, but instead something to lose.

seriously, go ask papa (nelson, lol) why he designs stuff with such high bandwidth.  if you are serious about your views and you want to know first-hand, GO TO DIYAUDIO and ask!  stop badgering me about this.

Why should i? You brought it up, you are to defend it. I bring up my reasoning why it is quite silly. You, so far, only have brought up "but they do this" and now "ask papa". Really? Is that the best you can do?

Again, to make it simple and clear: What improvement do you expect from playing back stuff at a frequency range that far exceeds what a human can hear? Are you aware of the potential drawbacks that result in having such a high sample rate for playback? What kind of speakers/headphones do exist to even make use of such a frequency range?

If you have no valid arguments then just say so. "Go ask papa" simply doesn't cut it. Physics doesn't care about some Nelson Pass guy, nor does it care about me or you. So, care to present some real argument now?

Greetings,

Chris

[mamalala]

Edit: This reminds me of the gaming folks who spend boatloads of money on a graphics card, only to proclaim "hey, in game X i get 200 fps now!". Which is pretty useless when the screen they look at has a refresh rate of 60 Hz...

n00b

60 Hz ~ 16milliseconds. Many monitors give below 5ms update time.

The high frame rate gives smaller latency, that is were you get some benefit. Especially if you are having network or other issues.

I wish I had 200fps.

I would rek.

Yeah, right. Sorry, i completely forgot that a fast panel can reallly suck the frames out of the graphics card. Here i was thinking that stuff like frame rate defined by the interface was the end of it. And yes, at 200 fps you have no more network lag. In fact, you see what people are going to do before they even know that they will do it! Like a magic time machine!



Greetings,

Chris

[HackedFridgeMagnet]

In fact, you see what people are going to do before they even know that they will do it! Like a magic time machine!

Yeah want one of those too!

[linux-works]

chris, please stop before you look even more foolish.

if you have never heard of nelson pass, PLEASE google it a bit first.

you are really acting foolish, here.

I'm out of this thread.  if that displeases you, too bad.

[mamalala]

chris, please stop before you look even more foolish.

if you have never heard of nelson pass, PLEASE google it a bit first.

you are really acting foolish, here. that high

I'm out of this thread.  if that displeases you, too bad.

that high

I know who Nelson Pass is, don't worry. He's just another audiophool. They come for a dime a dozen, you know.

However, he is not the one who wrote what you wrote. You were. And so far it seems that you are unable to defend your own arguments. "Ask papa", "look at DIYAUDIO", my ass. Yea, he claims to be able to build amps that go 100k+. Hey, tell you what: There are HAM's out there who can build you an amp that even does MHz! Some even in the GHz range! Go figure!

But none of that has any bearing on the usefulness of having audio material for playback that can reproduce a frequency range that far exceed the human hearing capabilities.

So, again, what _in_your_own_words_ is the supposed benefit of having audio files at, lets say, 96k or 192k sample rate? Don't bother with audiophoolery verbiage. Bring on real arguments.

Greetings,

Chris

[mamalala]

In fact, you see what people are going to do before they even know that they will do it! Like a magic time machine!

Yeah want one of those too!

Easy! Just get the latest and greatest graphics card. Betterm get 2 or 3 of them and link them together. Then only play DOOM, so you get 1k+ fps! Next, upgrade your router and PC to 10 gigabit fibre. Don't worry about your 10mbits DSL connection, all will be fine!

If in doubt, put some funny looking pebbles on top of your monitor, they will align the light rays it emits. And wrap your fibre cable in some funky stuff, otherwise those photons may shake around and arive all dodgy and stuff at the router. You don't want dizzy photons there, that's for sure!

Greetings,

Chris

[dexters_lab]

what a nonsense of a product

even when my hearing was better when i was younger i couldn't tell the difference between 320MP3 and CD. I have listened to two identical sources at 24/192 and 16/44.1 and could not tell the difference.

not only that but why a portable player? your even less likely to experience a difference when your walking down the highstreet with earbuds in!

Maybe its a ploy by the Flash makers to sell loads more MicroSD cards!