PonoMusic player by Neil Young

[ErikTheNorwegian]

Pono

[psycho0815]

i dunno. i get the point of having a dedicated store for high quality lossless audio files, but i don't get why i need that on a portable player. in most scenarios there will be other factors like ambient noise and crappy headphones, that will have more of an effect than mp3 compression.
Also there's that:

I'm an audiophile. Explain what is so special about the PonoPlayer technology?
The PonoPlayer was designed with a “no compromises” approach to sound quality. We partnered with the engineering team at Ayre (www.ayre.com) to include some of their world-class audio technology in our PonoPlayer. The Ayre team describes their contribution to the PonoPlayer design as follows:

•   The digital filter used in the PonoPlayer has minimal phase, and no unnatural (digital sounding) pre-ringing. All sounds made (including music) always have reflections and/or echoes after the initial sound. There is no sound in nature that has any echo or reflection before the sound, which is what conventional linear-phase digital filters do. This is one reason that digital sound has a reputation for sounding "unnatural" and harsh.

•   All circuitry is zero-feedback. Feedback can only correct an error after it has occurred, which means that it can never correct for all errors. By using proprietary ultra-linear circuitry with wide bandwidth and low output impedance, there is no need for unnatural sounding feedback.

•   The DAC (Digital-to-Analog Converter) chip being used is widely recognized in the audio and engineering community as one of the best sounding DAC chips available today.

•   The output buffer used to drive the headphones is fully discrete so that all individual parameters and circuit values and parts quality can be fully optimized for the absolute finest sound quality. The output impedance is very low so that the PonoPlayer delivers perfectly flat frequency response and wide volume range using virtually any set of headphones.

sounds like technobabble to me, but i'm no audio expert. Also if you so "no compromises" about it, wouldn't it make sense to sell some proper headphones with that thing?

[geppa.dee]

Comments about the Pono, published last November by someone that understands this way better than me :
http://www.dansdata.com/gz143.htm
(Spoiler... "audiophoolery" as a word is respectfully left out... but argued...)
The followup, http://www.dansdata.com/gz145.htm , is also quite a funny read, although less OT.

[codeboy2k]

This is clearly audiophoolery.. I doubt that Neil Young's ears are good enough anymore to distinguish between FLAC and MP3 at 192Kbps.

You don't need FLAC on a portable device, and as pointed out, external (i.e. environmental) noise will overpower any useful extra bits of audio information available in the format.

By using proprietary ultra-linear circuitry ...

What the fuck is ultra-linear? is that better than linear?

[Macbeth]

Isn't PornoMusic usually recorded on VHS or some ancient 70's cine film, and it all goes "boom chica wow wow"

Why would anyone want to listen to that on a portable player, and lossless as that? 

[Rigby]

Analogy time.

When my wife watches TV and sees commercials for exercise machines, I am quick to tell her that they don't work if she expresses interest.  She asks how I could possibly know that, and I respond with something like "if any of them worked, they wouldn't keep coming out with new ones."  If ANY of these machines worked as they expect you to believe they do, you would not see new machines invented, but rather refinements on the working design.

It's the same with audio stuff.  Every so often, a new thing comes out that claims to be sooo much better, blah blah, so on and so forth.  That's what the product previous to this one claimed, and the one before that, and the one before that.  In mathematics, what we should be seeing is called a limit.  1 + 1/2 + 1/4 + 1/8 + 1/16 on to infinity = 2.  Incremental gains toward the goal.  We should be heading to a specific point and each generation will make slighter and slighter improvements over the previous generation until we get to a point where our fallible human ears cannot tell reproduction from reality. 

We know we're after perfect sound reproduction, so there should be, by now, zero discernable difference between high quality audio gear, IF they are all true to their claim of perfect audio reproduction.

In truth, none of them are, they all add the distortion that sounds good at the time and claim perfect reproduction.  Warmth and midrange and all that ultra-whatever nonsense is just distortion added to make it more pleasing.  This is fine, I'm OK with this, but just SAY THAT THIS IS WHAT YOU'RE DOING.  You're not perfectly recreating anything except all the marketing tactics invented by others before you.

[Rigby]

Ok, what's the sample frequency and bandwidth?

Wasn't CD marketed as the format which could not be exceeded, as heard by the human ear?

Until I have 128 ears and can hear 128 tracks independently I don't see what a "master recording" could get me over CD.  I have human ears, which are very limited in terms of frequency response and fidelity.

[Marco]

What the fuck is ultra-linear? is that better than linear?

A good opamp headphone amplifier will probably trounce it in THD.

[Rigby]

What the fuck is ultra-linear? is that better than linear?

It's like linear, but ultra.  Geez, I can't believe I have to explain that.   

[tszaboo]

It also has ess 9018 dac, which is nice. Probably one of the best thing currently on the mobile market.

[MatCat]

Having been in recording studios I can tell you right now the quality of sound is far beyond anything a CD is going to give you, and my hearing sucks.  There is certainly a market for a player that is capable of high fidelity sound, and I doubt much that anyone that does buy one of these is going to use crappy headphones with it.

Would I buy it?  Probably not I am not that big of an audio snob, but I can tell you there is certainly room for quality over MP3s and CD, storage capability has finally caught up to make a player using lossless quality a possibility.

[Rigby]

Ok, but what are the specs on that audio?

Seems to me that you were using better headphones/monitors than you would elsewhere and you attributed it to the quality of the recording rather than the quality of the drivers.

[rollatorwieltje]

It's easy to improve the sound quality these days. Most music is completely crippled by the record label to sound as loud as possible. Virtually no dynamic range at all. So if they offer albums that are not compressed to death they may actually have a valid product. But they just made themselves look silly with all those wank stuff about the player.

[MatCat]

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

CD lossless quality recordings: 1411 kbps (44.1 kHz/16 bit) FLAC files
High-resolution recordings: 2304 kbps (48 kHz/24 bit) FLAC files
Higher-resolution recordings: 4608 kbps (96 kHz/24 bit) FLAC files
Ultra-high resolution recordings: 9216 kbps (192 kHz/24 bit) FLAC files   

[mamalala]

By using proprietary ultra-linear circuitry ...

What the fuck is ultra-linear? is that better than linear?

If it has tubes inside:

http://en.wikipedia.org/wiki/Ultra-linear



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.

On that note:

http://people.xiph.org/~xiphmont/demo/neil-young.html

Also, while MP3 is a lossy compression scheme, they don't lose stuff willy-nilly. Given a decent bitrate, one will have a very hard time to discern between an original recording and a 192 or 256 kbit MP3 version thereof, at least 99% of the time.

Greetings,

Chris

[Prime73]

here are a few interesting points on the subject: http://lefsetz.com/wordpress/

[tjaeger]

http://people.xiph.org/~xiphmont/demo/neil-young.html

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.

Also, while MP3 is a lossy compression scheme, they don't lose stuff willy-nilly. Given a decent bitrate, one will have a very hard time to discern between an original recording and a 192 or 256 kbit MP3 version thereof, at least 99% of the time.

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).

[geppa.dee]

This is anew way of both listning, buying and you get accsess TO THE MASTERTAPES IN FULL RESOLUTION.
Not CD quality.. full resolution master tapes..

http://www.dansdata.com/gz143.htm
High bit rate and sample depth are used in recording studios for the same reason photo editors use 48-bit colour and very high resolution instead of 24-bit colour and image dimensions the same as the final target. The larger format doesn't actually sound or look any better than the final - actually, works in the editing progress generally look and sound a lot worse than the final product, which is kind of the idea. But extra resolution and depth gives a bigger range for processing and correction without running out of resolution and creating odd artifacts.

Look for a 24bit color depth TIFF picture and compare to a high quality JPEG render of it (high quality MP3 analog, lossy) or even BMP/PNG (FLAC, lossless). I think you'll like the later two far more...

[Prime73]

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.

Look for a 24bit color depth TIFF picture and compare to a high quality JPEG render of it (high quality MP3 analog, lossy) or even BMP/PNG (FLAC, lossless). I think you'll like the later two far more...

[HackedFridgeMagnet]

 nbvck7-\*+

-;p;/0p;/[p;ll;llko-0-0              o


Edit: woops was cleaning keyboard. Now decided I had nothing worthwhile to add.

[mos6502]

PonoMusic was founded by Neil Young in 2011 to create a movement to revive the soul of music, and to recreate the vinyl experience in the digital realm wheedle money out of gullible fools.

Fixed that for ya 

[Nerull]

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.

[linux-works]

It's easy to improve the sound quality these days. Most music is completely crippled by the record label to sound as loud as possible. Virtually no dynamic range at all.

and movies, otoh, are 100% the opposite.  which I really hate!

I'm way past the phase of 'wow, loud BOOM.  COOL!!!'.  in fact, loud noises annoy me.  I want dyn range compression, not expansion.  there is a half-assed attempt at trying to have a 'nite watching mode' on dvd players (etc) but I don't think it does a good job.  it would not be hard to have an alternate track (or a diff track) that lets us non-loud-noise people be happy with movie sound tracks.

I'm constantly riding the volume control when I watch movies, daytime or nite-time.

[linux-works]

nbvck7-\*+

-;p;/0p;/[p;ll;llko-0-0              o


Edit: woops was cleaning keyboard. Now decided I had nothing worthwhile to add.

actually, you probably kept the NSA busy for a good half hour trying to decrypt that message.

LOL

[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!

[TMM]

This is anew way of both listning, buying and you get accsess TO THE MASTERTAPES IN FULL RESOLUTION.
Not CD quality.. full resolution master tapes..

http://www.dansdata.com/gz143.htm
High bit rate and sample depth are used in recording studios for the same reason photo editors use 48-bit colour and very high resolution instead of 24-bit colour and image dimensions the same as the final target. The larger format doesn't actually sound or look any better than the final - actually, works in the editing progress generally look and sound a lot worse than the final product, which is kind of the idea. But extra resolution and depth gives a bigger range for processing and correction without running out of resolution and creating odd artifacts.

Look for a 24bit color depth TIFF picture and compare to a high quality JPEG render of it (high quality MP3 analog, lossy) or even BMP/PNG (FLAC, lossless). I think you'll like the later two far more...

That's not a fair comparison because 24bit images are comprised of 8bit colour channels and usually have a good dynamic range. If you mess with the LSBs you're going to notice something.
Most music recordings on the other hand are no where near the 16bit noise floor to start with so if you discard a few LSBs here and there it's unlikely that anyone will notice. You have to have very high end gear to even listen with 16bits of effective accuracy.

I'm yet to meet a person who can reliably discern a difference between a 320Kbps MP3 (LAME encoder) and a 1411Kbps CD/FLAC/WAV file. That's a 4.4:1 compression ratio. If you compress an image 4.4:1 it will be noticeable even on a very average consumer spec monitor.

[tvtech]

Hi Guys

As of today 13 March 2014 @17H00 South African time Pono is sitting at 7,482 backers and $2,437,137 raised out of $800,000 goal with still 33 days to go 

Regards
tvtech

[tjaeger]

I'm yet to meet a person who can reliably discern a difference between a 320Kbps MP3 (LAME encoder) and a 1411Kbps CD/FLAC/WAV file. That's a 4.4:1 compression ratio. If you compress an image 4.4:1 it will be noticeable even on a very average consumer spec monitor.

Easy.  At least it used to be, I don't know if I could still do it and have no desire to find out.  This was rock music, not some contrived sample to bring out compression artifacts.  On a Sennheiser HD600.

Code: [Select]

foo_abx 1.3.1 report
foobar2000 v0.9.4.2
2007/05/22 18:24:46

File A: C:\...\abx\orig.wav
File B: C:\...\abx\insane.mp3

18:24:46 : Test started.
18:25:35 : 01/01  50.0%
18:25:45 : 02/02  25.0%
18:25:58 : 03/03  12.5%
18:26:08 : 04/04  6.3%
18:26:16 : 05/05  3.1%
18:26:25 : 06/06  1.6%
18:26:33 : 07/07  0.8%
18:26:48 : 08/08  0.4%
18:26:57 : 09/09  0.2%
18:27:04 : 10/10  0.1%
18:27:07 : Test finished.

 ----------
Total: 10/10 (0.1%)


[tvtech]

Hi Guys

Pono is set to break KS records...8,020 and  $2,608,020 so far.....early days yet....another 33 days to go.

All the people donating to Pono surely cannot be tone deaf or hard of hearing........I can just imagine the calls to the Factory in China where all all Manufacturing will take place......gear up NOW...we are already overwhelmed by the response"...

This is the very best thing I have seen in about 40 Years of dedication to music. If I have my little way..then Fleetwood Mac would join and be part of this.

The album " Rumours" comes to mind.

This can only get better and better.

Regards,
tvtech

[HackedFridgeMagnet]

Just picked out a couple of points from the blurb.

All circuitry is zero-feedback. Feedback can only correct an error after it has occurred, which means that it can never correct for all errors.

Zero Feedback Wonder how they do that? Might make it hard to control the volume.

By using proprietary ultra-linear circuitry with wide bandwidth and low output impedance, there is no need for unnatural sounding feedback.

Proprietry ultra linear circuitry, sounds like something they should be hiding not advertising. If I wanted proprietry I would get an IPod, Apple are the experts there.

[linux-works]

By using proprietary ultra-linear circuitry with wide bandwidth and low output impedance, there is no need for unnatural sounding feedback.

Proprietry ultra linear circuitry, sounds like something they should be hiding not advertising. If I wanted proprietry I would get an IPod, Apple are the experts there.

playing devil's advocate, its conceivable that you can have a circuit that gives a linear response and is a new approach (maybe using dsp or other hybrid methods).  who knows.  but to put blame on a design because its 'proprietary' does not mean anything bad, per se.  lots of amp and dac designs are 'proprietary', which means that someone came up with a novel approach.  that's ALL it means.  why read more into it than that?

apple (etc) are 100% not relevant here.  why muddy the waters, so to speak?

[HackedFridgeMagnet]

I guess I'm thinking open source hardware and software would be good.
That's all, not an uncommon desire.

[linux-works]

if its open source hardware, its probably cpu-based playback.  that could be good since those are the only real solutions to true gapless playback on ALL mp3's (not just the 'treated' mp3's that have magic in the headers to describe the real endpoint in the last bufferful of data).

for portable use, I really can't understand why you'd need 'the best' audio playback.  if you are at a desk, you can use slightly larger players and even portable hard drives.  if you are truly on the go, your ambient noise level is probably going to mask any high end sound benefit.

once smart phones came on the scene, dedicated music players kind of died, sales-wise.  almost the same for cameras; smart phones are killing portable LSD camera sales (lsd = 'little silver digital', LOL).

[BravoV]

All circuitry is zero-feedback. Feedback can only correct an error after it has occurred, which means that it can never correct for all errors.

Zero Feedback Wonder how they do that? Might make it hard to control the volume.

Not an electronics expert here, I mean in term of electronic circuit design's analogy, isn't that like driving a car but totally blind folded ?    cmiiw

[linux-works]

its very very common to use negative feedback (like on the tee shirt, lol) to control amps.  you can extend the feedback loop to include buffers, too (so that you have voltage gain, first, and then current gain after).

but, its also possible to have topologies that do not have any feedback at all.  they are harder to keep the distortion low but they have no 'speed' issues since they don't depend on a control mechanism to keep them in check.

I have no problems with feedback based amp topologies.  it all depends on how well the concept is executed and both non-fb as well as fb designs can work well.

[elCap]

This is great! Thanks for sharing.
I will definitely back this project. Not so much for the player, but for the musicians that are doing this to improve the quality of music reproduction. I hate the loudness of many albums these days, the compressed dynamics, clipping and other carp. Or the new super duper remastered release of an old classic that just sounds terrible. I really hope real music is back!

One problem might be that I have to buy all my albums once again.. But if Pono delivers the music quality I'm after, I'm OK with that.

Then I must say that it's too bad that every time a music product topic starts here at EEVblog there is always talks about Audiophoolery or whatever. I wonder if any of these people ever have tried listen to real audio gear? Tried to do some blind A/B tests? It is not that hard, and can be quite fascinating. Yes, I consider myself an audiophile. I can hear the difference between different power cords any day of the week, blind test, random or whatever. Piece of cake. But to be able to do that, you have to train, know what you want to get out of the music.

[Rigby]

I can hear the difference between different power cords any day of the week, blind test, random or whatever. Piece of cake.

I'd like to see some proof of that.  Enough that it's clear they aren't lucky guesses.

[HackedFridgeMagnet]

I can hear the difference between different power cords any day of the week, blind test, random or whatever.

Nonsense.

If you said 'power chords' that would be different.

[Macbeth]

Then I must say that it's too bad that every time a music product topic starts here at EEVblog there is always talks about Audiophoolery or whatever. I wonder if any of these people ever have tried listen to real audio gear? Tried to do some blind A/B tests? It is not that hard, and can be quite fascinating. Yes, I consider myself an audiophile. I can hear the difference between different power cords any day of the week, blind test, random or whatever. Piece of cake. But to be able to do that, you have to train, know what you want to get out of the music.

Scientific testing, especially with extraordinary claims like perfectly up to spec cheap IEC mains power cables are going to produce a more accurate sound reproduction (especially within the realms of human hearing). A single blind study with your audiophool friends just doesn't cut it. You need a double blind test. The JREF will actually award you $1,000,000 USD if your claims and YOUR agreed double blind protocol are statistically valid. You do not have to even claim to know how your claim works, just prove it does, and $1,000,000 is yours! Slight caveat - there are so many cranks and crackpots going for the challenge that don't even know how to word an application form, that potential claimants should at least have some media exposure of their claims (or you push your vendor to prove in your case?)

[linux-works]

you know, if an IEC power cord change makes a difference, I would actually be blaming the power supply subsystem.  its actually hard to imagine any power supply being so bad that an input cord would actually matter, assuming no gross errors and all cords are at least to local code standard.

[SeanB]

Only thing I will add to this is that I can hear the difference between MP3 at 128k and 192k, so all my MP3 recordings are at 320k. Most of the music I have is in Ogg though at the highest bit rate it supports. That way I can always go down, and get a usable copy. Thogh the originals in most cases are old LP records and a good number of 78RPM records that are in no way high fidelity. There I have a faithful copy of the original rumble, noise and scratches. Because I had no player capable of playing the 16 2/3 RPM records they had a little resampling done to them to get the time right.

As to playback, you have a choice of filters, 20kHz brick wall with really poor impulse response ( destroys transients like drums, piano and any other percussive instrument) or a very limited high frequency response that starts at 10kHzand rolls off at audible frequencies. A good reason for oversampling is to make the filter only handle very high frequencies so it can have a gentle roll off and low phase errors in the band of interest. As well most CD's still have a rumble filter applied to roll off at under 50Hz to reduce LP record noise ( kind of not needed with a digital signal) and have some horrible DC offsets as well.

[tvtech]

Hi Guys

Metallica has joined (mystery to me... alll they make is noise).

Sitting at 9,499 backers and $3,166,929 and 32 days to go.....Just wait if they get real artists like Bob Dylan, Tracey Chapman,
or a lot of others to contribute/join......never mind the likes of Pink Floyd etc......

This thing should explode and go totally viral if handled the correct way.......Lets see how the Pono team handles this....I see history @ kickstarter being defined by this project.

The race is on between Pebble and this.....

tvtech
.

[tvtech]

Just hit $5,006,834

16 Days still to go.

Awesome 

[Fsck]

you know, if an IEC power cord change makes a difference, I would actually be blaming the power supply subsystem.  its actually hard to imagine any power supply being so bad that an input cord would actually matter, assuming no gross errors and all cords are at least to local code standard.

that's what I'd think too. I'm sort of surprised that no audio people actually care about the quality of their mains and just... use mains with their fancy power cords. if they really did care, they'd be running everything from a double-converting UPS at minimum, or for the more extreme, generate their AC from a large motor.

you don't even need a scope to show how poor quality mains is, a logging dmm is all you really need. did it out of curiosity a while back with a BM869 when it arrived to test out the software, which was set to poll every 0.5s for 4 days. amusingly, the graph of the voltage log it produced of the 4 days looked like 8 extremely noisy sinusoidal cycles.

[richms]

I had some dropouts on my system solved by swapping a dirt cheap IEC from the computer parts box out for a hospital grade shielded one, It was stuffed in with all the interconnects, but after changing it, no more HDMI dropouts when the fridge or similar clicked on or off. Not sure I would go beyond that, and perhaps a better HDMI cable would also have solved the dropout problems.

[CanadianAvenger]

the difference might be due to the presence of an emi choke on your higher grade cord.

[Corporate666]

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.

As for this Pass guy, I will say this... there is no shortage of people who are willing to sell someone what they want to buy-regardless-of-function.  If that person requires an emotional plea from the seller that "this product really will be everything you have convinced yourself it will be", then there are plenty of sellers who will offer them just that.

The psychology of why people buy things - especially luxury/unnecessary items, is quite fascinating.  But a great summation is here:

http://www.gmarketing.com/articles/13-why-people-buy

Items like 3, 4, 5, 9, 10 and 25 are powerful reasons why people fall for audiophoolery.

[linux-works]

except, if you actually don't know who nelson pass is, you should check.  he's not a 'phool'.  I won't speak for him, but he's approachable on diyaudio and so anyone who thinks he's not what he claims he is, you have the complete freedom to call him on his designs and go one on one with him.  I have no doubts that he'd hold his own with anyone here.

there are a lot of fake designers.  he is NOT one of those.  a little research on the man would easily and quickly reveal this.

[Corporate666]

except, if you actually don't know who nelson pass is, you should check.  he's not a 'phool'.  I won't speak for him, but he's approachable on diyaudio and so anyone who thinks he's not what he claims he is, you have the complete freedom to call him on his designs and go one on one with him.  I have no doubts that he'd hold his own with anyone here.

there are a lot of fake designers.  he is NOT one of those.  a little research on the man would easily and quickly reveal this.

Audiophools are a self-serving group like any cult.  My point is simply that there mere existence of these products in no way substantiates that they are superior to other products anymore than the existence of e-meters proves the existence of thetans in Scientology.  There are people who want to believe in something and choose Scientology; and e-meters help satisfy that belief.  And there are people who want to believe in audiophoolery, and there are numerous products and people who are happy to satisfy their belief (and take their money). 

The higher you go up the pyramid of audiophoolery, the more abstract the terms become, the more faith is required over empiricals, and the more money is required.  It goes back to the psychology of purchasing that I posted and why people buy the things they do.

Nelson Pass... famous for his belief that performance tests on an amp do not tell the whole story besides listening to it.  Gee, what a coincidence 

[linux-works]

the idea is that listening+testing=more complete analysis.  I see nothing wrong with that.  I would not rely soley on either one, to be honest.  I think that's all he (and others) are saying.

the ones that ignore test gear entirely, those are the ones I am suspicious of.  but those are not the people I was referring to in any of my posts.

[Fsck]

the idea is that listening+testing=more complete analysis.  I see nothing wrong with that.  I would not rely soley on either one, to be honest.  I think that's all he (and others) are saying.

the ones that ignore test gear entirely, those are the ones I am suspicious of.  but those are not the people I was referring to in any of my posts.

I would basically agree, except that very few (I've personally never heard of anyone) people collect enough samples of subjective listening tests to even make any reasonable suppositions. Even the psychology-types learn basic sampling skills so if any of those guys have bachelor degrees or above, they should try to get a refund.

[GK]

The post-DAC analogue circuitry of this thing is apparently designed by Charles Hanson. Charles is the guy behind Ayre Acoustics, the magic myrtle blocks and the brown noise "irrational but effective" compact disc.

The claims made wrt negative feedback are pseudo technical drivel to the n-th degree. Charles Hanson is an anti-negative feedback crusader. It's his schtick to differentiate his audio products from the competition. According to him negative feedback is "bad for the sound".


 

[jancumps]

.... Charles Hanson is an anti-negative feedback crusader. It's his schtick to differentiate his audio products from the competition. According to him negative feedback is "bad for the sound".


 

Oh no! here on the eevblog we only give negative feedback. Now we're stuck.

[CanadianAvenger]

On the whole listen vs test thing.

About 20 years ago, my company got its first internet connection using an ISDN telephone line. After a few months we started having some problems where the connection kept dropping out. The phone company techs kept coming out and throwing their expensive testers on the line, sometimes leaving them to log over several days, and each time the tester said the line was perfect. After about 5 such service calls, I managed to convince the guy that the tester was not telling the truth. He was an older guy and said he would try one more thing. He reached into his bag, and grabbed his old POTS line tester [basically a clip on analog phone handset] he clipped it on, and almost immediately we could both hear the problem. Every once and a while there was a loud thwacking crackle on the line, like a lightning strike. Sure enough after the tech moved the line to a clean pair, we never had another drop-out after that.

Tests are only as good as the test case they are designed to measure. Listening allows for a much broader evaluation, though to lesser precision.