[stv014]

I think it is possible for some height information to be reproduced in headphones without binaural or surround sound. if you ever played a fps with headphones & heard an helicopter flying overheard, for example.

 
The game may have used HRTF if it knew you are using headphones, though. Especially with a sound card that supports some kind of virtual surround (e.g. Dolby Headphone). Other than that, one expects a helicopter to be above, so in that case imagination can easily fill in the missing spatial information. It is possible to tell front and rear sounds apart by turning slightly (just like in real life if the direction of a sound source is ambiguous), e.g. if it appears to come from the center, turning to the left by a small amount will move the perceived position to the right/left if it is at the front/rear, respectively. How much it moves also gives a hint about its elevation.

 

[bigshot]

I think it is possible for some height information to be reproduced in headphones without binaural or surround sound. if you ever played a fps with headphones & heard an helicopter flying overheard, for example.

 
That is just a regular sound effect that your mind is placing above you because you know helicopters fly overhead. If you recorded a chain saw or power lawn mower the exact same way, they wouldn't seem like they are overhead.
 
I was listening to the binaural hair clippers and my mind kept snapping the clippers in front of my head, then behind. Unless I focused on holding it in one place, it went all over the place. Your interpretation of the cues are more important than the cues themselves sometimes.

 

[The Walrus]

Hey guys,
This question just popped into my mind. (maybe it has been answered, sorry if that's the case) Since we don't have ears at the top of our heads, how does our ears tell whether a sound is coming from above? Or is it just our brain interpreting and placing the source of the sound in space?
By the way, what do you say about Ultrasone's claim here: http://www.ultrasone-headphones.com/en/technology/slogic 
 
cheers 

 

[Don Hills]

...  Since we don't have ears at the top of our heads, how does our ears tell whether a sound is coming from above? Or is it just our brain interpreting and placing the source of the sound in space? ...

 
The shape of your pinnae (outer ears) alters the sounds you hear. A sound from overhead will sound different than the same sound from in front or behind you. It works best with sounds that you are familiar with. You've probably observed this yourself - you can often tell where a familiar sound is coming from (such as the snap of a twig), but when you hear an unfamiliar sound you look all around trying to locate the source, especially when it occurs on your midline - from directly in front to directly behind you.

 

[sonitus mirus]

  Hey guys,
This question just popped into my mind. (maybe it has been answered, sorry if that's the case) Since we don't have ears at the top of our heads, how does our ears tell whether a sound is coming from above? Or is it just our brain interpreting and placing the source of the sound in space?

 
 
http://artsites.ucsc.edu/ems/music/tech_background/te-03/teces_03.html
 
"Height information is provided by the shape of our ears. If a sound of fairly high frequency arrives from the front, a small amount of energy is reflected from the back edge of the ear lobe. This reflection is out of phase for one specific frequency, so a notch is produced in the spectrum. The elongated shape of the lobe causes the notch frequency to vary with the vertical angle of incidence, and we can interpret that effect as height. Height detection is not good for sounds originating to the side or back, or lacking high frequency content."

 

[analogsurviver]

  The increased bandwidth in higher resolution digital audio only increases the maximum frequency captured. It does not increase the bandwidth available to sound in the audible range. 44.1 kHz captures everything in the audible range, higher sampling rates capture exactly the same thing with the addition of frequencies we can't hear. Likewise there is nothing analog captures within the audible range that Redbook can't. We've been over this.
 
Can you please also explain why depth and height cues cannot be captured by Redbook but width cues can? What is the difference between them that makes this true?

Width - or span from extreme left to the exteme right and in between - can be most easily reproduced because they mainly rely on loudness of the signals, that is to say amplitude. Front to rear or depth perception has to involve some time cues - as the sound does not stop at some precisely "boundary" in any given real listening enviroment and echoes off walls etc mainly define the acoustics of that room.  Rise time of redbook  is approx 14 microseconds - or bandwitdh to 20 kHz. IF even I accept that this is "enough" - that means that this has to be maintained from input to output -from the sound waves impigning on the diaphragm of the microphone to the sound impigning on the eardrum - with every component in between fast enough to maintain that 14 uS risetime even under worst of conditions. In series as they are, that means each and every component does add so and so much delay/filtering - that means even the best equipment available does nor find maintaining that 14 uS easy - OVERALL. 
 
There is no (pre)amplifier with infinitely short risetime or infinity bandwidth, no cable, and most certainly no microphone. Each and every component is slowing the original sound down somewhat - and this can CLEARLY be heard in redbook vs "higer speed" ( be it analog or digital ) : redbook is mostly flat surface as far as depth is concerned, under best of circumstances it has some depth that is definitely limited and does not match the same as analog or hirez digital and most certainly not live feed from the microphone.
This has no "boundaries" except for the real boundaries in that room ( walls etc ) - redbook never reaches not nearly close in regard of depth, because it can not convey those tiny time differences that do give us perception of depth - it is not amplitude mostly/alone. Sound after climax of an orchestra does "travel" - to the other side from the performers, to the rear wall, it reflects off the rear and side walls, etc - and these time cues are not nearly good enough with redbook. Or, they are - until
higher resolution version OF THE SAME RECORDING/MASTER is heard. "Perfect sound forever" brigade is VERY likely to say "it must be another mix/master" - because it is tiny tiny details at high frequencies that distinguish between "recording" or "real" - at least one octave above what redbook can provide. It is those barely audible entrances of strings etc that are on DSD audible and redbook only cathes up few micro(mili?)..seconds later - once the amplitude is high enough and/or the natural delay/ringing of digital filter has settled down - and clearly, their conclusion is - it MUST be different mastering ?!?!? I can not vouch for others as to how have they recorded and/or whether various resolutions of digital are really the same mix/master - when I listen to my original DSD128 master and its redbook counterpart derived from it (NO other stunts other than conversion of formats ) it does sound to me as described. YMMV.
 
The most significantly audible jump in quality in digital occurs from redbook to 88.2 or 96 kHz sampling frequency - this IS "night and day". Further increase in resolution is not that audible at first instant - but does matter in the long run. SACD or DSD64 is not decisively significantly better than redbook - but DSD128 IS. 
Going up from here, DXD ( 352-376 /24 and up to 752/24 ) and DSD256 and DSD512 should finally close the gap between microphone feed and recording - because they introduce errors/delays that really should not matter anymore. There are bottlenecks long before the limits of these formats are reached, most dominant being the microphones. Slowly but securely more 100 kHz and beyond mics are appearing - even if one thinks it is a waste of everything and total nonsense, the use of such mikes means they cover audible band more easily than designs that struggle even within audible band.
 
ALL THE ABOVE MAKES SENSE ONLY IF AND WHEN SIMPLE RECORDING TECHNIQUES ARE USED - THAT MEANS TWO MICROPHONES. Any multimiking will introduce such gross time errors that advantage analog/hirez has over redbook may well NEVER be heard. Which limits the higher frequency rate digital to acoustic non amplified music - and certainly not pop etc, where natural sound without electricity practically does not exist. Here, the "tools" used are just too crude to warrant going to the above lengths regarding resolution of recording - but I would love to be proven wrong on this one.
 
All of the above also means it is not possible to make a remaster of say Kind of Blue that could challenge the modern recording on technical terms. It is what it is - great piece of music - but not anything to write home about regarding sound quality. It is about the best available at the time of its creation.
 
And precisely because it is great piece of music, I am going to listen to it now. Haven't in a long while - and will do it off vinyl which says mastered from original analog master tape. CBS roughly 90's. There were and will be digital "whatevers" - the real master for KInd of Blue is analog master tape and the best approximation is second generation of analog master tape - or so it used to be. New FAST digital may in the end displace it and analog record from the throne - but never redbook.

 

[RRod]

I think the "perfect sound forever" crowd would also say "double blind or meh."

 

[sonitus mirus]

  Width - or span from extreme left to the exteme right and in between - can be most easily reproduced because they mainly rely on loudness of the signals, that is to say amplitude. 

 
I just finished reading the information in this link from NYU:
 
http://www.cns.nyu.edu/~david/courses/perception/lecturenotes/localization/localization.html
 
In this link they report about 2 cues available for sound localization, inter-aural intensity differences and timing differences.
 
 
The paragraph about Measurement of timing differences was most interesting.
 
"Very small differences in time between the two ears requires quite large differences in intensity to compensate for the perceived displacement of the sound."

 

[analogsurviver]

  I think the "perfect sound forever" crowd would also say "double blind or meh."

I agree.
 
TROUBLE ?
 
Computer audio is not far enough to allow decent ABX - in native DSD. The moment we are forced to go into PCM for the ABX software(s) to be enabled, much of the difference has already been thrown away. I did try MUCH time to see how well various softwares "translate" into real life sound scenario using mainly iFi nano iDSD DAC that allows VERY decent quality native DSD playback - for $/Euro 189, which is an affordable price. Trouble is, nano is not that great with PCM - therefore I will try the iFi audio micro iDSD DAC - which is reportedly one hell of a lot better, across the board, but particularly with PCM and does , as the first, support the ludicrous fast PCM and DSD formats from my previous post - at $/Euro 500.
 
When and if software for ABX in native DSD ( or native DSD vs PCM WITHOUT DSD being converted into PCM ) will be released ( IF it can be done at all ... - I am not a mathematician, but it might well be impossible, similar situation as mastering in purely 1 bit DSD is not possible ). Then it would have to be perfected to a point it will no longer "click" whenever going from PCM to DSD or vice versa - thus making sure listener can not "count clicks" and thus essentially cheat.
 
Computer software and hardware is FAR from "sounding equal" - at least at the present state.  And can cost as little as say 500 $ ( nano or similar DAC + "some" computer ) to sky is the limit - and although I would not expect the sky is the limit machine to play MP3s better than 500 $ rig can play DSD, it can well turn out it can coax out of redbook better result than lower price rig out of DSD. HOW can then such comparisons be fair - because even the software would tend to be MUCH different - just look at the requirements for the HQ Player. It requires a dedicated heroic specified PC for music exclusively.
 
I find this situation similar to analog turntables; not so big difference in magnitude, but perceivable and audible nonetheless. So presently sending the sound around the globe and expecting it will sound exactly as intended on the other side is wishful thinking at best.  
 
I am the last person ever to go into cables debate (unless applications that REALLY call for such and such cable - or it basically no longer works as intended ) - but these differences might be small enough to be masked by cable defficiences. And no, I do not spend any ludicrous amounts for cable, one that I like to use (with clear knowledge it can be improved upon - at normally hefty price increase  ) costs the whole of Eur 2.00 (actually, it IS below 2 per metre ). It is "good enough" to pass the information required.
 
So, it is a bit more complicated than it seems at first glance. Hope a decent solution to the problem will be found in reasonable future.
 
I would like to add that I am not apriori against redbook/CD - and admit that while posting around SQ, I did get to learn that redbook does not carp out at say approx -90 dB, but can, properly dithered and played with good software, achieve above ( or better said, below ) - 100 dB. Trouble - some softs are good in this regard, others suck. And not much is being written or said about it.
 
Compared to above, analog is "easy" - at least it is more predictable. Trouble is that it tends to be pricey at quality level mentioned.

 

[RRod]

  I agree.
 
TROUBLE ?
 
Computer audio is not far enough to allow decent ABX - in native DSD. The moment we are forced to go into PCM for the ABX software(s) to be enabled, much of the difference has already been thrown away. I did try MUCH time to see how well various softwares "translate" into real life sound scenario using mainly iFi nano iDSD DAC that allows VERY decent quality native DSD playback - for $/Euro 189, which is an affordable price. Trouble is, nano is not that great with PCM - therefore I will try the iFi audio micro iDSD DAC - which is reportedly one hell of a lot better, across the board, but particularly with PCM and does , as the first, support the ludicrous fast PCM and DSD formats from my previous post - at $/Euro 500.
 
When and if software for ABX in native DSD ( or native DSD vs PCM WITHOUT DSD being converted into PCM ) will be released ( IF it can be done at all ... - I am not a mathematician, but it might well be impossible, similar situation as mastering in purely 1 bit DSD is not possible ). Then it would have to be perfected to a point it will no longer "click" whenever going from PCM to DSD or vice versa - thus making sure listener can not "count clicks" and thus essentially cheat.
 
Computer software and hardware is FAR from "sounding equal" - at least at the present state.  And can cost as little as say 500 $ ( nano or similar DAC + "some" computer ) to sky is the limit - and although I would not expect the sky is the limit machine to play MP3s better than 500 $ rig can play DSD, it can well turn out it can coax out of redbook better result than lower price rig out of DSD. HOW can then such comparisons be fair - because even the software would tend to be MUCH different - just look at the requirements for the HQ Player. It requires a dedicated heroic specified PC for music exclusively.
 
I find this situation similar to analog turntables; not so big difference in magnitude, but perceivable and audible nonetheless. So presently sending the sound around the globe and expecting it will sound exactly as intended on the other side is wishful thinking at best.  
 
I am the last person ever to go into cables debate (unless applications that REALLY call for such and such cable - or it basically no longer works as intended ) - but these differences might be small enough to be masked by cable defficiences. And no, I do not spend any ludicrous amounts for cable, one that I like to use (with clear knowledge it can be improved upon - at normally hefty price increase  ) costs the whole of Eur 2.00 (actually, it IS below 2 per metre ). It is "good enough" to pass the information required.
 
So, it is a bit more complicated than it seems at first glance. Hope a decent solution to the problem will be found in reasonable future.
 
I would like to add that I am not apriori against redbook/CD - and admit that while posting around SQ, I did get to learn that redbook does not carp out at say approx -90 dB, but can, properly dithered and played with good software, achieve above ( or better said, below ) - 100 dB. Trouble - some softs are good in this regard, others suck. And not much is being written or said about it.
 
Compared to above, analog is "easy" - at least it is more predictable. Trouble is that it tends to be pricey at quality level mentioned.

 
Good PC audio isn't hard. A Magni/Modi stack gets you there for a vast majority of cans. Just combine it with a good player (again easy to find), find masters that aren't total sh@#, and you're done. Everything beyond that has left ABX territory and is into desk aesthetics.
 
It's certainly possible to do DSD -> Redbook -> DSD, I just don't know the technical issues enough to know how it would affect ABX testing. I do know my own ears can't discern even 16/88.2 from Redbook, so I don't see how DSD does anything other than really fancy noise shaping. And my personal opinion is that if "cable deficiencies" matter, then we're into snake oil territory.
 
I didn't think dithering was in any way hard these days, and I've even seen arguments that it's not even necessary, as the distortions it's hiding rarely stand out within the context of actual music. The sox "dither -s" command does to my ears a perfect job, and that's free…

 

[analogsurviver]

   
I just finished reading the information in this link from NYU:
 
http://www.cns.nyu.edu/~david/courses/perception/lecturenotes/localization/localization.html
 
In this link they report about 2 cues available for sound localization, inter-aural intensity differences and timing differences.
 
 
The paragraph about Measurement of timing differences was most interesting.
 
"Very small differences in time between the two ears requires quite large differences in intensity to compensate for the perceived displacement of the sound."

True. You can create an experiment at home; first, listen to your stereo speakers in normal central position. Then go say 10 degrees off axis ( that is usually less than "one chair from dead center left or right ) - and measure how much you have to compensate with amplitude intensity to perceive the sound in dead center again.
 
That is why time differences are alfa and omega in audio - troughout the evolution, it was being perfected to allow our survival in nature. And that is why limiting the range of time differences by using too slow devices, such as redbook, is wrong. If fast enough devices are used, 2 chanell stereo can have quite a decent depth - which all but pancakes with redbook. One does not hear a different tune with "something fast" compared to redbook; but under favourable conditions, the recreation of acoustic space can be so good that it entices listening on this ground alone.
 
Where did it all began? Turntables, to be exact phono cartridges. A phono cartridge can have all but one parameter from excellent to divine - if there is phase difference between the two stereo channel output, it will not sound good no matter what. It is something phono cartridge manufacturers tend to keep as low profile as possible - because it is highly sample to sample dependant. Thus it is entirely possible to get better sound with a lucky (next to ) ideal sample of an entry level cartridge than with TOTL that is slightly "off". Specs for phono cartridges DO NOT include phase difference between the channels - if they  did, that would have driven the cost of cartridges up at least 30 % to cover for the "rejects" that currently are being sold "bussiness as usual".
 
And if anyone thinks all CD players have good phase relationship among the two channels, he/she is in for a rude awakening. It is true that nowadays lower priced "redbook", computers included,  is guilty of this. By running the faster sampling frequency ( ALL the way your hardware and software can support ), this phase difference goes proportionally lower and enables MUCH better
left to right and even front to back localization - despite the fact that DAC of said device does not support above usually 48 kHz sampling rate.
 
But even redbook machines with perfect phase between the two channels can do absolutely nothing if signal time differences are shorter than their own rise time - that is why faster devices are required, not because of < 1 % people that can hear above 20 kHz. Differences in time can be perceived by all humans, even with 20 kHz capability HALVED - and that is why an elderly gentleman with lots of experience with listening can subjectively evaluate say a tweeter better than a teenager with perfect hearing but without experience required.
 
I never claimed I have superhuman hearing capabilities - just getting in writing ( remember, I am not a native English speaker ) what i know to be true proved much more difficult that I have ever emancipated.

 

[diamondears]

I don't know the objective measurements, but subjectively, the FIRST time I listened to a DSD128 recording, it's a truly wow moment. I couldn't believe my ears. The realistic-ness of the sound is unprecedented. I'm a believer. And I believe that DSD will be the mainstream format in 2 years time. If just 1 major popular label will invest on this, everything else will follow as the incremental improvement is very substantial that people cannot be "deceived" anymore by the un-investing labels.

 

[diamondears]

For non-believers (yet...), try buying the DSD128 version of the Jazz on the Pawnshop album (condensed album/selected songs in 1 album), and compare it to the regular red book or even high-res PCM version. The difference is SUBSTANTIAL. One sound/instrument becomes farther, another the same, and another less farther/loud. Amazing.

I was a non-believer of high-res PCM as I don't hear substantial improvements over red book or 16/44. But DSD, totally different scenario.

 

[headwhacker]

 I don't know the objective measurements, but subjectively, the FIRST time I listened to a DSD128 recording, it's a truly wow moment. I couldn't believe my ears. The realistic-ness of the sound is unprecedented. I'm a believer. And I believe that DSD will be the mainstream format in 2 years time. If just 1 major popular label will invest on this, everything else will follow as the incremental improvement is very substantial that people cannot be "deceived" anymore by the un-investing labels.
Edited by diamondears - Today at 12:26 pm

 
Quote:

For non-believers (yet...), try buying the DSD128 version of the Jazz on the Pawnshop album (condensed album/selected songs in 1 album), and compare it to the regular red book or even high-res PCM version. The difference is SUBSTANTIAL. One sound/instrument becomes farther, another the same, and another less farther/loud. Amazing.

I was a non-believer of high-res PCM as I don't hear substantial improvements over red book or 16/44. But DSD, totally different scenario.

 
Are you sure it's not just 2 different masters/source you are comparing? Is the PCM version deived from the same DSD source?

 

[Don Hills]

Originally Posted by analogsurviver /img/forum/go_quote.gif
 ... limiting the range of time differences by using too slow devices, such as redbook ...
 
And if anyone thinks all CD players have good phase relationship among the two channels, he/she is in for a rude awakening. It is true that nowadays lower priced "redbook", computers included,  is guilty of this. ...
 
But even redbook machines with perfect phase between the two channels can do absolutely nothing if signal time differences are shorter than their own rise time - that is why faster devices are required, not because of < 1 % people that can hear above 20 kHz. Differences in time can be perceived by all humans, even with 20 kHz capability HALVED ...

 
It is a common belief that Redbook (or any digital chain) is limited in time resolution to the time represented by the time between samples. This is incorrect. 16/44.1 digital can resolve time differences into the nanosecond range, orders of magnitude less than the time between two samples. If you doubt, I can refer you to videos and papers proving it.
 
Some very early CD players had a phase shift between channels due to sharing a single DAC between channels due to cost. In one I have seen the test results for, the difference amounted to 22 degrees phase shift at 20 KHz. All players and DACs that you can buy since the 90s have one DAC per channel. Phase shift is essentially zero.
 
Finally, rise time and frequency response are one and the same thing, viewed in different domains (time versus frequency).