HydrogenAudio

Hello all!

I was wondering if anybody who owns an SACD/DVD-A audio player could comment on the quality of their machine and if they can hear a difference between a standard 44.1Khz/16 bit CD and the newer formats. I am asking this because I have been tempted to purchase such a player, but cannot decide between the two formats. In addition, since I am on a low budget, I wonder if SACD/DVD-A are even worth it? Isn't the 192Khz sampling rate a tad excessive (or the 2.8(?)Mhz DSD stream on SACD)? 44.1Khz seems fine to my ears. . .24-bit word-length would be nice, but how much difference would it make, esp. considering that 99.9% of the 900 recordings I own were done in analog?
I probably won't purchase the SACD/DVD-A format unless there is a proven difference (statistically significant) in subjective quality in a laboratory, and a meaningful one at that.  Any thoughts on this, or papers somebody could point me towards?

Originally posted by Cygnus X1
I probably won't purchase the SACD/DVD-A format unless there is a proven difference (statistically significant) in subjective quality in a laboratory, and a meaningful one at that. 

Wow, I don't think you'll be buying one then! 

Remember these formats are mainly about marketing and 'secure digital'.

Don't forget HDCD - yet another format CD - now (I think) owned by Microsoft.

What a shame this is happening - I hate having more non compliant CD formats (assuming its some sub-channel kludge). It's an interesting idea but I think if no one can tell the difference there really is no point to it all.

Ruairi

Originally posted by rc55
I hate having more non compliant CD formats (assuming its some sub-channel kludge).

well, at least HDCD is 100% redbook compliant (I think) - all information needed for a hdcd decoder to work is stored in the LSB of the audio data

That's quite interesting.

If only there were some open tools to make HDCD... - then again I'm contradicting myself - I probably couldnt tell the difference.

Ruairi

Originally posted by rc55
That's quite interesting.

If only there were some open tools to make HDCD...

the basic idea is that upon "encoding" a 20-24bit audio signal to hdcd filters are applied if necessary (dynamics compression on low levels, limiter to prevent clipping), and when you play the cd through a hdcd decoder, the control stream in the lsb triggers complementary filters to undo these operations, effectively pushing the resolution of the sound to 18-20bit.

I don't know how much of this can be really heard though, since I never had the opportunity to listen to a HDCD

Hi,

I own a Tosh (model 500) DVD/ DVD-A/HDCD player and I have a few HDCD encoded CDs. I don't have any DVD-A yet !

As for the HDCD encoded CDs (24bits), they are perfectly compatible with any CD player (16 bits). Frankly, the difference between the same CD played on my HDCD compatible player and my old Technics CD player using the same amp is not that obvious. You really nead a very quiet listening setup and set your amp with a comfortable volume to notice a slight improvement (mainly in the stereo image and in the treble) but i'm not an expert listener.

IMHO the main point with DVD-A or SACD is the ability to reproduce multi-channel recorded stuff. And I think this is quite an improvement over 2 channels CD or HDCD. You don't need to be an expert to feel the difference between a stereo recorded concert and a multichannel recorded one (5.1 -> 7.1). But of course you need more hardware !

Christian

Well, there's a huge difference between 6 channel DVD-Audio and normal stereo CD-Audio when it comes to actually "feel" the music... (Assuming the [re]mastering has been done well...) But quality-wise, there's not much difference for the majority of people... Even if you have the equipment to deliver 96kHz, 24-bit audio per channel you probably won't have the hearing to actually be able to hear any difference if you only use two channels...

I've never heard SACD so I can't say I know how it sounds... But it seems cool to be able to play them in normal CD players without a SACD player... Stereo audio only though, but still...

The only thing I "really really" don't like about these two new formats is the encryption and therefor I won't buy any new discs of the newer formats for a long time... I already own one DVD-Audio disc, but that's enough for me... Normal CD-Audio works just fine at the moment with my equipment...

The opinions everybody has stated here largely reflect my own regarding the two newer formats. Another thing I did not consider was the fact that the only "new" recordings I buy are classical and jazz, since IMHO the best rock and roll recordings have long ago been cut. Classical may sound good in 7.1 if the engineer knew what he was doing; the added ambience would probably be much more pleasing (and natural sounding) than using DSP effects to create reverb, a capability found on many modern recievers. Save for that fact, I am one of those people who would rather hear two well-recorded channels of audio rather than 5 or 6 channels of echoey, poorly mixed sludge. Opening another can of worms, I also agree with several posters that the DRM on the newer formats is appauling. How can the record companies put out consistently worse material (i.e. bands completely lacking in musical talent and ability) and complain when people don't buy new albums and/or put compressed audio on the internet of better, older music? In any case, it looks like the ubiquitous audio CD will continue to be my format of choice (though I have gone to a 512MB solid-state player and --alt-preset standard for portable use).

Here was a big topic about PCM (that is CD or DVD-A) versus DSD (that is SACD) in George Massenbourg professional audio recording forum :

http://www.musicplayer.com/cgi-bin/ultimat...ic;f=3;t=002225 (http://www.musicplayer.com/cgi-bin/ultimatebb.cgi?ubb=get_topic;f=3;t=002225)

I'll try to summarize what I remember of it, from the audiophile point of view :

First point, impossible to find scientific well documented papers about how the DSD technology of SACD is supposed to improve the sound. Only advertisement hype.

Then, as it was discussed in an older huge thread ( http://www.musicplayer.com/ubb/ultimatebb....t=000822#000000 (http://www.musicplayer.com/ubb/ultimatebb.php?ubb=get_topic;f=3;t=000822#000000) , more than 1000 posts, better read from the end  ) there is no proof that frequencies above the limit of the audition affects the sound (but, according to the non-linearity of human hearing, no proof of the opposite either). In fact, there is one article, but we can't read it : http://www.aes.org/events/110/workshops/HW-7.html (http://www.aes.org/events/110/workshops/HW-7.html) . It might only deal with distortion in electronic gear, I don't know. A similar claim was made by Mike Richter (see the end of the huge post linked above), but the experimental setup is not rigorous, at least as it is reported.

Therefore the improvement promised by 192 kHz sample rate is not granted.

High resolution PCM (96, 192 kHz, 24, 32 bits) is required in the mixing process, that's a fact, because the signal get amplified or changed, and inaudible parts can become audible after processing (think about increasing treble, or lowering the speed, for example), but can it improve sound for the final user ?

One argument is that DACs must use lowpass filters at 22,050 Hz, that affect the sound, because the lowpassing process must be smooth in order not to create artifacts, therefore the lowpass must run from 19 or 20 kHz, for example, to 22,050 Hz. A higher sample rate gives more margin.
On the other hand, it is quite impossible to mix audio in pure DSD. A conversion to PCM is needed to apply even a volume/balance adjustment. Some prototypes of pure DSD processing machines exists, but they are not yet distributed. Thus an SACD is in fact just like a copy of DVD-A !

Another argument, common to PCM and DSD, is that DVDA and SACD behaves like partly converted audio data. A DVDA is like a CD that would have already been oversampled before analog conversion, and an SACD goes even further in the process. In short, it's a bit like if half of the D to A conversion had already been done before burning, so that you only need half a DAC in your player to play the SACD. therefore you benefit from a state of the art professional quality for the first half of the DAC (before the mastering of the CD) and a classic second half (in your player).

Now the negative part : frequencies above 20 kHz, that are the main advertised advantage of these new technologies should be rejected anyway, in order not introduce distortion in the rest of the chain !!!

Let's finish with this exellent joke : http://www.prosoundweb.com/recpit/viewtopic.php?t=1556 (http://www.prosoundweb.com/recpit/viewtopic.php?t=1556)

From Nika :

I have talked to a lot of manufacturers about this, and many of them say that there will be no benefit to higher sampling frequencies with their boxes. This includes Lexicon amongst many others. 

Many manufacturers say that the only benefit is the potential audible ability of us to hear above 20kHz. While they say that THEY can't hear a difference, they make the boxes for those who claim they can.

Sorry if I was too technical for this board, it has nothing to do with my new status .:red:

Thanks Pio2001 for your input and info!

For all the arguing people engage in over high-sampling rate/word length audio, it seems from my perspective that there is a lack of empirical support for using ungodly sampling rates and/or 32 bit wordlengths.                                                                    I also find it amusing when people complain about aliasing. . .am I correct in that this hasn't been a problem with CD players for at least a decade? I can recall when CD players came out that they used a very steep lowpass filter around 19-20Khz, which lead to some harshness (and if I recall correctly) even some phase shifting near the filter frequencies. Of course, oversampling has since been implemented to use much less steep cutoffs. In either case, the filtering around ~20Khz leaves 2.05Khz between the lowpass and the "last" possible sample values, and given that there isn't much in the way of dynamics around 24Khz and up, I am not sure why aliasing would even be discussed anymore.                                                                                                                        In addition, if the companies like Sony are providing recordings and machines with ultra-high frequency content for those "who can hear it", why then is most of the SACD catalouge comprised of old 1950's jazz? Im sorry, but I am doubtful that 1950's two or three track tape technology could capture those frequencies , even at 15ips, especially considering the primitive FeO2 tape coatings used at the time. I have a great recording from 1959, Dave Brubeck's Time Out, both in analouge (vinyl) and on CD (with 20-bit remastering). They both sound the same to me, though my vinyl introduces a host of surface noise defects that I find annoying to say the least.
I think I'll pass on the SACD player and instead use the money to build a box to play MPC's or mp3's in my car:D

I say we don't need DVD-A/SACD, as current solutions offer good enough audio. If CDDA is not enough, just use auio-only DVD with DD or DTS tracks. That gives you mc capabilities and (with high enough bitrates) better sound in plain stereo, as well. I guess even here it will be hard to tell the difference between a 44,1kHz 16bit int track and a DD/DTS track, which is 48kHz and (theor.) 32bit float. In case of DD/DTS it depends on the amp with how many bits it decodes, and of course on the mastering with how many bits the source was digitized before encoding. On should note that DD/DTS can also be recored on a plain CD - then with 44,1kHz.

So DVD-A/SACD is only a marketing hype to
a) make money (of course) without much (or any) improvement to audio quality to 99,99% (guessed number) of the people (including me I guess)
b) make more money by adding copy protections into the standard

Yes, the music industry is trying to copy protect CDDA, but that is by breaking the standards (Red Book) and in the end it is not very effective.

But of course there are always people with too much money (the Laserdisc/upcoming D-VHS guys out there) who probably will have fun in buying DVD-A/SACD...

Originally posted by DarkAvenger
I say we don't need DVD-A/SACD, as current solutions offer good enough audio. 

Exept if they manage in the future to built DVDA or SACD players that sounds as well as 3000 € CD players, but for 200 € only. That would be interesting 
In my opinion, it's the only relevant question about audiophile (=not talking about multichannel, I mean) possibility of these new media.

I have heard SACD, HDCD and DVD-A.  I own two SACD players (Sony NS500V...  $200!).  SACD is the superior format IMHO.  The difference between CD and DVD-A is always described as "subtle".  The difference between CD and SACD is definitely noticable.  My parents (truly non-audiophiles) heard the Sony demonstration at Hi-Fi '99 in Chicago with me and both commented on how much better SACD sounded.

Would I recommend SACD to everyone?  No.  I have *very* high resolution listening rigs at work and home and, on them, these differences are very noticable.  On a boombox or computer speakers, I doubt the difference would be noticable.  Beyond that, there are some serious flaws in Sony's marketing of SACDs.  The catalog is weak at best.  ALL of Sony's releases are non-hybrid (they won't play in a normal CD player).  You can't rip MPCs from these discs.

So, for me, SACD is a curiosity.  I'll get some cornerstone discs in the SACD format (Kind of Blue, etc.) because the SACD sounds markedly better than the CD.  I'd like the format to take off, but don't think it will.

Originally posted by Pio2001

Exept if they manage in the future to built DVDA or SACD players that sounds as well as 3000 € CD players, but for 200 € only. That would be interesting  
In my opinion, it's the only relevant question about audiophile (=not talking about multichannel, I mean) possibility of these new media.

Pio - FWIW, my $200 Sony NS500V playing the SACD sounds better than my $900 Rega Planet 2000 playing the CD.  The NS500V sounds awful playing CDs, though.

The real problem with saying one sounds better than the other is that you can't make any true comparisons at the moment.

SACD material will have been recorded either with analog gear and then digitally recorded from the master into DSD (which supposedly sounds quite good) which is tough to compare to a pre-existing CD version since the converters used are probably different and could have been lower quality for the original CD.  Tech advances over the past few years have enabled some pretty amazing sounding converters, so more recent CD's might sound better in comparison than the same master recorded 10 years ago onto CD.  So that's tough.

Some material originally done at 16 or 20-24 bit at 44.1 at 48Khz sampling rates and then converted to DSD probably won't sound much better than the original unless some additional tweaking is done, which would probably make a version released on CD sound better as well.

Anything done these days in digital format is probably recorded at 24/48Khz, some are moving to 24/96, but that takes alot of processing power to mix and record.  anything at 48Khz is upsampled to 96Khz for DVD-Audio, possibly when moving to DSD, but again, you're limited by the performance of the converters at 48Khz.  As was previously mentioned low-pass filters are used in 44.1 and 48Khz conversions and the quality of the filters also affects the recorded sound.  A good chunk of the material currently on DVD-Audio is simply upsampled 24/48 material, so comparing it to the CD version wouldn't reveal any huge differences.

Due to the different companies involved, it's impossible to compare a DVD-Audio release to a SACD release because nothing's available in both formats to make a direct comparison.  Even if you could, the test would have to be done by a third party since in the DD vs DTS comparisons, the mixing engineers can often tweak things differently for different encoding processes, making one sound "better" than the other.  Mixing, mastering and encoding all play in heavily on the final sound.  Better clarity in the high end or deeper bass could simply be due to different instruments, asthetic tastes and many other things that have nothing to do with the final release.

In truth, there is no real answer and the most likely answer is that DVD-Audio, using PCM technology which is already in use across the industry will win because it involves the least financial investment.  Using my $800 8-channel recording unit sitting next to me I could mix and created DVD-Audio discs if I already had the appropriate software.  SACD would require a significant investment and the same situation exists in most digital studios around the world.  Spend $3000 on DVD-Audio software or spend $30K+ on DSD conversion hardware, which do you think most studios would go for?

G

Originally posted by cmokruhl
I have heard SACD, HDCD and DVD-A.  I own two SACD players (Sony NS500V...  0!).  SACD is the superior format IMHO.  The difference between CD and DVD-A is always described as "subtle".  The difference between CD and SACD is definitely noticable.  My parents (truly non-audiophiles) heard the Sony demonstration at Hi-Fi '99 in Chicago with me and both commented on how much better SACD sounded.

It is not hard to produce desired test results. Just do a sloppy job when mastering the CD and an excellent job when mastering the SACD. How many CD<>SACD comparisons have you done and can you list the titles?

Originally posted by Annuka
It is not hard to produce desired test results. Just do a sloppy job when mastering the CD and an excellent job when mastering the SACD. How many CD<>SACD comparisons have you done and can you list the titles?

Joe Satriani "Engines of Creation"  ->  recent recording so the "old ADC" argument doesn't work.
Miles Davis "Kind of Blue"  ->  compared against recent SBM remaster
Santana "Abraxas"  ->  compared against original
Stevie Ray Vaughan "Texas Flood"  ->  compared against recent SBM remaster

In all cases, the highs sounded much purer and more natural.  Being a guitarist, I know the sound of a guitar and there's a much more natural attack and purity of tone in the SACDs.  Even good sounding CDs sound "digital" when compared to the SACD.

Originally posted by cmokruhl


Joe Satriani "Engines of Creation"  ->  recent recording so the "old ADC" argument doesn't work.
Miles Davis "Kind of Blue"  ->  compared against recent SBM remaster
Santana "Abraxas"  ->  compared against original
Stevie Ray Vaughan "Texas Flood"  ->  compared against recent SBM remaster

In all cases, the highs sounded much purer and more natural.  Being a guitarist, I know the sound of a guitar and there's a much more natural attack and purity of tone in the SACDs.  Even good sounding CDs sound "digital" when compared to the SACD.

Black magic and witcvhcraft.
Where's your proof - double blind listening tests or other validated test methodology.
The sort of 'analysis' you make above is similar to the high end hi-fi media.
It may be true, but you demonstrate no proof. The placebo effect is powerful. If I buy a new SACD machine, drop my new $big SCAD in the tray, smoothly glide it into the soft lit front panel and press play - my god its going to sound good!

Originally posted by Ruse
Black magic and witcvhcraft.
Where's your proof - double blind listening tests or other validated test methodology.

Just because I didn't do double-blind listening tests does not necessarily mean there is no difference.  I mean, if I listen to a rock CD followed by a classical CD and comment that they sound different, would you say that my opinion is invalid because I didn't do a double-blind listening test?

Its this comment particularly that needs proving:

Originally posted by cmokruhl
Even good sounding CDs sound "digital" when compared to the SACD.

With reference to many comments above about the recording, mixing and mastering of each medium, I don't think this assertion will be settled as fact or fiction for a long time.

The new generation of CD players with upsampling techniques blurs the issue as well. You see, its not just an issue of the media, but the recording, mixing, mastering and the player technology. So when you make a bald statement like the one I have quoted, I just think that you have not really considered all the variables much.

Some random thoughts to add to this thread:

A few years ago Audio magazine (RIP) published a very technical article that tested the affects of HDCD and the upshot was that at best it did nothing and at worst it raised the noise floor of the recording.  Of course, it was a toss up as to whether it was worse to encode / decode with HDCD at all or to have an HDCD encoded CD that was not decoded.  Anyway, most of the affect was marginal at all.

I have heard both DVD audio and SACD, and they sound quite good.  As to whey they are primarily old jazz releases the reason is that SACD is targeted initially at audiophiles who as a group prefer old jazz and orchestral (classical) recordings.  Britany fans aren't running out to buy SACD gear; middle age weathly audiophiles will buy it, so give them the recordings they like.

DVD-Audio is a sad Frankenstein of a format that had potential when it first started.  Originally intended to be the mother of all audio formats, technically its okay.  It is very flexible (44.1kHz, 48kHz, 88.2kHz, 96kHz, and 192kHz sampling rates; word lengths of 16, 20 and 24 bits; and 2 to 6 channles of audio; lossless MLP [Meridian Lossless Packing] compression is available; ability to store data, pictures, lyrics, etc.; of course encryption; and "watermarking" to hunt down the evil doers who might bother to copy one of these things and prevent unauthorized recordings/broadcasts).  The encryption was originally the same as DVD-Video (of course), but DVD-Audio was about to launch at the time that the DVD-Video encryption was compromised.  So, DVD-Audio was delayed a year to redo the encryption.  Some nice things about the format are that in dual layer, 16bit, 44.1kHz stereo with MLP (true CD audio equivalent) I think I calculated play times of about 20 hours.  Now, if you could use ogg....The encryption and watermarking were not enough to protect this fortmat, so we have the further restriction that you can only get analog outputs from DVD-Audio (and SACD) players.  There are some high-end propriatary digital links, but the manufacturers have you run six analog cables to your receiver, and you have to mess with delays and crossovers in the DVD-Audio or SACD player separatetly from your home theater Dolby Digital settings lest you defile your prescious 24bit/192kHz DVD-Audio analog signal by crunching it through an AD/DA cycle in the receiver....

So, there you have it, SACD and DVD-Audio crippleware.  On top of it all, SACDs and DVD-Audio discs cost more than CDs ($25 US +).  Tough to see a mass market for these as much as they are technically impressive.  Remember that 24bits gives you a dynamic range of 144dB - not many audio systems have that kind of accuracy, and it really become pointless in a car/airplane or even a home with an air conditioner.

Rant over.

Originally posted by DigitalMan
So, there you have it, SACD and DVD-Audio crippleware.  On top of it all, SACDs and DVD-Audio discs cost more than CDs ( US +).  Tough to see a mass market for these as much as they are technically impressive.  Remember that 24bits gives you a dynamic range of 144dB - not many audio systems have that kind of accuracy, and it really become pointless in a car/airplane or even a home with an air conditioner.

I would go further and say that there is no chance of either of these formats having any kind of mass market. However, their development was not driven by customer's desires for multi-channel remixes of their favorite CDs (or their disgust with that wretched 16/44 sound). Their development was driven by the major label's demands for a secure format and their inability to make unrippable CDs. And I suspect that as soon as most new CD players will handle SACD and/or most DVD players will do DVD-Audio, then the phase-out of the CD will begin.

...assuming that the market puts up with that... 

Originally posted by bryant

I would go further and say that there is no chance of either of these formats having any kind of mass market. However, their development was not driven by customer's desires for multi-channel remixes of their favorite CDs (or their disgust with that wretched 16/44 sound). Their development was driven by the major label's demands for a secure format and their inability to make unrippable CDs. And I suspect that as soon as most new CD players will handle SACD and/or most DVD players will do DVD-Audio, then the phase-out of the CD will begin.

...assuming that the market puts up with that... 

The sad thing is that we'll whine and complain, but if that's the only way in which we can procure new music, we'll buy it.  It'll piss me off to no end though if they keep the prices at $25 and CD's at $15-$18.  At that point I might just stop buying music.  Old tech is supposed to drop in price, not increase.

G

To clarify:

HDCD is not 24 bits. It is 16 bit, otherwise it wouldn't be playable in a normal CD-player. Aside from the filtering stuff, there's the dynamics compression/expansion which results in some extra dynamic range, and there's subtractive dither going on, but that's it.

I owned a HDCD-player not long ago and liked it alot. I think the problem with HDCD is that the mastering engineer has a number of options to choose from, and choosing full-fledged HDCD means compromising undecoded playback, and vice versa. Hence, many HDCD's don't sound as good as they could. Good examples of great HDCD's are Neil Young's "Silver & Gold", Rodney Crowell's "Houston Kid" and Bruce Springsteen's box-set "Tracks" (or "18 tracks") - all guitar-heavy stuff that sounds very good on a HDCD-player. Beck's "Mutation" and "Midnite Vultures" are great too, but mastered to loud, in my opinion.

/ Uosdwis

More clarifications on HDCD,

Sorry I was wrong saying in a previous post that HDCD is 24 bits. HDCD is actually encoded on 20 bits but
the encoding scheme makes it compatible with 16 bits CD players.
So it will play at 16 bits on a normal CD player and at 20 bits on an HDCD compatible player.

check http://www.hdcd.com/about/whatisHDCD.html (http://www.hdcd.com/about/whatisHDCD.html)

Christian

Originally posted by Ruse


Black magic and witcvhcraft.
Where's your proof - double blind listening tests or other validated test methodology.
The sort of 'analysis' you make above is similar to the high end hi-fi media.
It may be true, but you demonstrate no proof. The placebo effect is powerful. If I buy a new SACD machine, drop my new $big SCAD in the tray, smoothly glide it into the soft lit front panel and press play - my god its going to sound good!

So in this case we have a negative placebo effect?

He already owns a Rega Planet 2000 CD player priced at $900. This is truly a high-end audiophile unit. With such a unit in his system, the other units are probably sweet too.

The SACD player is a cheap low-end unit at $200 that quote: "sounds awful playing normal cds." At this price I would be surprised if it would even play music.

So we have three "results":

- A high-end CD player with an excellent 16bit DAC that sounds nice.
- A low-end SACD player with cheapest 24bit DAC that sounds awful when playing normal cds.
- A low-end SACD player with cheapest 24bit DAC that sounds nicer than the high end unit (if not nicer, then just as nice or maybe even close to jut as nice).


Double blind tests are nice. But in this case not relevant, as we are changing two factors: Player and media. Before performing the tests, we need some music in CD and SACD format mastered by 3rd party. I would like four different types:

A: Extraordinary mastering of both CD and SACD.
B: Extraordinary mastering of SACD, quick mastering of CD.
C: Extraordinary mastering of CD, quick mastering of SACD.
D: Quick masterign of both SACD and CD.

I think the results would be like this:

A: Very subtle differences
B: Major improvements -- like comparing DVD-Video with VHS
C: The most interesting result. If the SACD comes even close to the CD, then there is hope for new music yet.
D: Time is money, so this is probably the most relevant test

Originally posted by bryant

I would go further and say that there is no chance of either of these formats having any kind of mass market. However, their development was not driven by customer's desires for multi-channel remixes of their favorite CDs (or their disgust with that wretched 16/44 sound). Their development was driven by the major label's demands for a secure format and their inability to make unrippable CDs. And I suspect that as soon as most new CD players will handle SACD and/or most DVD players will do DVD-Audio, then the phase-out of the CD will begin.

...assuming that the market puts up with that... 

No new format has any chance for mass market before all hardware manufacturers and the label companies agree on a standard and work together.

History has shown that quality matters very little. VHS won over BetaMax, Dolby Digital is standard sound format in DVS-Video -- DTS is better... I own many DVDs with poor image quality... some of them worse than the VHS equilvalent.

Right now Sony is selling expensive SACD players to the few people who just "need" the newest technology at any price. When this phase is completed, we'll see what happens.

It isn't hard to convince the public to change -- especially not if both industries work togetger. Just keep releasing crap dymanic compressed CDs and good SACDs/DVD-As.

In my opinion, the sonic advantage of DVD-A and SACD is pure marketing hype, in an attempt to renovate part of the established base of CD players, also avoiding the possibility of making copies in these new formats.

It is also my opinion that standard redbook cd properly implemented (easy today) is transparent , I mean, beyond the capabilities of human hearing. All perceived differences in the new formats are due to a carefully done mastering.

I've read that JVC's XRCD standard redbook cd's, which are also carefully mastered, sound as good as SACD discs.

It would be interesting to know, as Pio pointed, at the recording/playing chain of these formats, how many of the microphones and processing devices (at the side of the recording) and speakers (at the side of the reproduction) are capable of recording, transmitting, or reproducing signals above 20-21 KHz and taking advantage of the 24 bits of dynamic range.

Besides, I've not read of any properly implemented blind test that showed perceived differences between those formats.

So, I find it very unlikely that the mass market is going to adopt a new system that offers no advantages (apart from multichannel sound), and is not copiable. In case of DVD and CD, they offer obvious superior quality and ease of use over VHS and vinyl.


If you have a 24/96 sound card, you can do you own 24/96 versus 16/44.1 tests with the material provided at:

http://www.pcabx-pro.com/technical/sample_rates/index.htm (http://www.pcabx-pro.com/technical/sample_rates/index.htm)

It is physically *impossible* to record at 24bits w/o noise. There is always (thermal) noise. (To prevent this try to go next to 0°K...but then don't forget to take care of other maybe unwanted effects. ) But by this you'll have "natural" dither.

@Annuka

DTS only sounds better because of the bitrate. Dolby/DVD consortium was stupid enough to limit DD to 448kbps... I wonder how DTS would sound at that bitrate.

As Pio2001 explained, for SACD and DVD-A it is easier to sound better because of less quality DAC needed to produce good sound. But I have some doubt that the DA->AD->DA (player to amp to speaker) will do good for SACD/DVD-A...Yuck! (Thinking further, how many bits do you probably loose at the AD step again due to thermal noise?)  So to assert SACD's pure sound you must connect it to an analogue amp, as all newer receivers work digital, so in the end you are listening to a SACD on-the-fly converted to DVD-A...
I am pretty sure with identical mastering an upsampled CD won't be distinguishable from DVD-A (if you limit DVD-A to 16bit; to get more bits for CD just encode to DTS or DD). SACD may sound different (but maybe not better) as it is purely based on dither.

Thinking further:

I dunno how it is in the ultra high-end section, but usually digital amps are used in the the price regions I know. SACD can *never* sound better then DVD-A on such an amp due to the medium/technique! As I stated above the amp processes the signal digitally (mid-range: 48kHz at 16 or 24bit to high end 192khz at 24bit) the SACD will be converted DVD-A like, so if we can assume that it could be possible to connect DVD-A digitally the DA/AD steps will go away and we have a pure pcm transfer, DVD-A will sound more original than SACD.

If SACD does sound better on a digital amp it is due to mastering. I don't think that that DA/AD will improve sound...

So for me SACD makes absolutely *no* sense, and even sense of DVD-A is doubtful.

Originally posted by DarkAvenger
Thinking further:

I dunno how it is in the ultra high-end section, but usually digital amps are used in the the price regions I know. SACD can *never* sound better then DVD-A on such an amp due to the medium/technique! As I stated above the amp processes the signal digitally (mid-range: 48kHz at 16 or 24bit to high end 192khz at 24bit) the SACD will be converted DVD-A like, so if we can assume that it could be possible to connect DVD-A digitally the DA/AD steps will go away and we have a pure pcm transfer, DVD-A will sound more original than SACD.

If SACD does sound better on a digital amp it is due to mastering. I don't think that that DA/AD will improve sound...

So for me SACD makes absolutely *no* sense, and even sense of DVD-A is doubtful.

I've heard that only the REALLY expensive models has digital out. And we are talking $6,000 +.

HDCD being capable of pseudo 20-bit playback only affects the dynamic range of the music, not the apparent quality of sound.  As Ethan Winer uncovered in a semi-official test, most people can't identify the difference between 16 and 24-bit audio, unless dithering has been applied to the 16 bit audio.  24-bit provides advantages during the recording stage because the volume or a recorded track can be modified over a greater range without dropping any bits.  Most music though is compressed down to having a 14-16dB or lower average dynamic range during the mastering stage.  This is considered "good" dynamic range but most modern music squashed that down to something like 6dB i the interests of making the CD "loud."  24-bit properly recorded and converted to 16-bit theoretically should sound identical to a full 24-bit recording played back at 24-bit.  A 24-bit recording would probably retain all the musical detail better than 16-bit would if your CD player has a digital volume control and you turned it down there rather than adjusting the analog volume of the amplifier.

KikeG mentioned that he didn't see the advantages of high sample rates if recording and playback equipment generally can't touch frequencies above 20Khz.  The issue is really technical rather than purely being about the reproduction of nyquist frequencies.  Part of it deals with the acuracy of reproducing given frequencies.  At 44.1 you get 2.6 samples per 17Khz wav, at 96Khz you get 5.6 samples, at 192 you get 11 samples.  Higher frequencies also inherently reduce the effects of jitter error, again, bringing high-end CD player clock stability to the average user at 96Khz.  Another issue is distortions at or near the nyquist frequency.  44.1 sampled material must filter those out because they are still in an audible range, 96Khz pushes those well out beyond the range of human hearing allowing the DAC to do away with the filtering process.  The filtering process, if done poorly can affect the quality of audio, contributing to the increase in clarity many describe when switching to higher sample rates.

In all cases you're bringing high-end audio quality to the mass market.  It's a similar situation to what happened with CD's.  Analog turntables and tape decks produced subjectively better sound, on a really good system with a really good turntable with a very well maintained album.  However, CD players have lower noise and better clarity than the turntables or tape decks that most people used, so for some it was keeping things on par (albeit with lower noise and great conveniences like not destroying your albums as you played them) and for everyone else it was a great improvement.

Now, the rest of it, and probably the biggest reason its being pushed is monetary reasons.  Selling people something new, and selling them something that can properly be copy-protected.  In theory I agree with copy protection, especially as the artist's right to protect their property, but as a user who keeps all his music on his HD, I disagree with anything that prevents me from easily doing that.  I especially disagree with the workaround of downloading MP3's made someone elses determined quality settings.  Not only would they be MP3's instead of my prefered MPC's but they'd likely be at 128CBR since that's what the industry likes to tout as "CD quality."

G

Originally posted by gdougherty

...The issue is really technical rather than purely being about the reproduction of nyquist frequencies.  Part of it deals with the acuracy of reproducing given frequencies.  At 44.1 you get 2.6 samples per 17Khz wav, at 96Khz you get 5.6 samples, at 192 you get 11 samples. Higher frequencies also inherently reduce the effects of jitter error...

I'm not sure if you badly worded this or if you are misunderstanding something.

You present the number of samples for a given freq, and go on to say that 'higher frequencies also...', which implies that the number of samples in itself is an argument, which it isn't (and that's exactly what nyquist is about).

As Ethan Winer uncovered in a semi-official test, most people can't identify the difference between 16 and 24-bit audio, unless dithering has been applied to the 16 bit audio.

This is wrong. Most people could not identify the difference between the 16 and 13 bit files. There were two who could distinguish between the 16-bit ones, and _those_ preferred the truncated file on _subjective_ grounds. (http://ff123.net/24bit/24bitanalysis.html (http://ff123.net/24bit/24bitanalysis.html))

--
GCP

Originally posted by DigitalMan

So, there you have it, SACD and DVD-Audio crippleware.  On top of it all, SACDs and DVD-Audio discs cost more than CDs ( US +).  Tough to see a mass market for these as much as they are technically impressive.  Remember that 24bits gives you a dynamic range of 144dB - not many audio systems have that kind of accuracy, and it really become pointless in a car/airplane or even a home with an air conditioner.

Rant over.

Dynamic range is calculated by

6.02 dB * bits + 1.76 dB + Gain

Gain is around 15 dB for a sampling frequency of 44 kHz, around 36 dB for 96 kHz,
48 dB for 192 kHz.
It uses the effect taht the ATQ is not linear but frequency dependend.

So 24 bit/96 kHz supports 182 dB of dynamic.

Originally posted by Frank Klemm


Dynamic range is calculated by 

6.02 dB * bits + 1.76 dB + Gain

Frank, I've been wanting to ask this before: where does the 1.76dB come from?

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GCP

Originally posted by Garf
Frank, I've been wanting to ask this before: where does the 1.76dB come from?

Yeah, do you still get almost 2 dB of dynamic range if bits = 0 (i.e. no data)? That would be a pretty good deal!

Originally posted by Garf


I'm not sure if you badly worded this or if you are misunderstanding something.

You present the number of samples for a given freq, and go on to say that 'higher frequencies also...', which implies that the number of samples in itself is an argument, which it isn't (and that's exactly what nyquist is about).

I did in fact mean to state that higher sampling rates reduce the effects of jitter since, like the filtering I mentioned, it pushes jitter out beyond the point where humans can reasonably detect it.  Thanks for catching my miswording.  More samples do mean greater accuracy and faithfulness to the original analog signal, I did mean that portion of my statement.  Nyquist as I understand it is not about accuracy, but determining the cutoff that a given sampling rate can reasonable reproduce.  IMO 88.1 or 96Khz are sufficient, 192Khz would theoretically be better for accuracy, but I believe it's a point of diminishing returns, and with 192 the returns would be extremely diminished.

Originally posted by Garf

This is wrong. Most people could not identify the difference between the 16 and 13 bit files. There were two who could distinguish between the 16-bit ones, and _those_ preferred the truncated file on _subjective_ grounds. (http://ff123.net/24bit/24bitanalysis.html (http://ff123.net/24bit/24bitanalysis.html))

-- 
GCP

My mistake, the results still fully support my argument.  Only via modifications to the source, or during the resampling process can anyone really discern a difference between varying bit-depths.  Bit-depth allows for greater dynamic range, but not any inherent gain in audio quality.  Theoretically, if done without the need for dithering, a downsampled 16/96Khz recording would sound indistinguishable from an original 24/96Khz recording.  Not only that, but it would reduce the storage space requirements.

G

Originally posted by gdougherty
More samples do mean greater accuracy and faithfulness to the original analog signal, I did mean that portion of my statement.  Nyquist as I understand it is not about accuracy, but determining the cutoff that a given sampling rate can reasonable reproduce.  IMO 88.1 or 96Khz are sufficient, 192Khz would theoretically be better for accuracy, but I believe it's a point of diminishing returns, and with 192 the returns would be extremely diminished.

You have a very fundamental misunderstanding. At 44khz sampling rate, you can reproduce a 20Khz signal exactly as good as if you were sampling at 192Khz. This is the very essence of the nyquist theory. There is no gain whatsoever in increasing the sampling rate above two times the highest frequency that you want to record/reproduce.

If this feels intuitively strange, read up on the theory of sinc functions.

My mistake, the results still fully support my argument. 

I agree with this. I made the correction because as written your original statement suggests it's better not to dither to get better accuracy in reproduction, which is false. (Not dithering may introduce additional distortion which can make the music sound better subjectively in some cases, which is what likely caused the curious result)

Dithering is _good_. You don't need to dither, but you _want_ to do it. The reason why the result of the test is seemingly at odds with this is that the listeners were not comparing to the 24-bit original, but only comparing among the two 16-bit files alone. They had no idea of determining which one was more true to the original, so they had to make a completely subjective pick among them. In that case, it's possible they pick the more distorted file because it has more ' pizzaz ' or whatever. 

--
GCP

Originally posted by Garf


You have a very fundamental misunderstanding. At 44khz sampling rate, you can reproduce a 20Khz signal exactly as good as if you were sampling at 192Khz. This is the very essence of the nyquist theory. There is no gain whatsoever in increasing the sampling rate above two times the highest frequency that you want to record/reproduce.

If this feels intuitively strange, read up on the theory of sinc functions.



I agree with this. I made the correction because as written your original statement suggests it's better not to dither to get better accuracy in reproduction, which is false. (Not dithering may introduce additional distortion which can make the music sound better subjectively in some cases, which is what likely caused the curious result)

Dithering is _good_. You don't need to dither, but you _want_ to do it. The reason why the result of the test is seemingly at odds with this is that the listeners were not comparing to the 24-bit original, but only comparing among the two 16-bit files alone. They had no idea of determining which one was more true to the original, so they had to make a completely subjective pick among them. In that case, it's possible they pick the more distorted file because it has more ' pizzaz ' or whatever. 

-- 
GCP

That does sound counterintuitive.  I had assumed there was a point at which it didn't matter anymore, and I guess due to the extremely short duration of the signal it makes some amount of sense, but it still seems odd.  I shall have to read up on this.  So then really the advantages of 96Khz would primarily come from reduced jitter effects (if you believe in jitter effecting audio quality) and dropping the need to filter out distortions inherent to the digital process.

Yes, dithering is not bad, where I made a point to say things should be done without dithering I was intending more to say that the same dynamic range would easily fit within 16-bits in most cases and between the two, not adding any dither into the situation would be best for comparison's sake.

Very interesting....

G

according to Nika, http://www.musicplayer.com/ubb/ultimatebb....3;t=000822;p=33 (http://www.musicplayer.com/ubb/ultimatebb.php?ubb=get_topic;f=3;t=000822;p=33)

As to your other questions, higher sampling frequencies will not benefit us when it comes to jitter because the amount of jitter between like amounts of time will be the same. The error from 1 44,100th of a second to the next will be still x picoseconds, regardless of what the frequency derived for clocking purposes is.

There's little left for 96 kHz sample rates, in theory.
Now all I want is listening, no player is perfect, and I still wonder if cheap 96 kHz converters sound better than 44.1 kHz ones.

At least recording a vinyl in 44.1 and 96 kHz 16 bits on the Marian Marc 2 soundcard, I sweared that the 96 kHz captured incredibly much more of the "analog" vinyl sound than the 44.1 kHz... until all audible differences suddenly vanished under the ABX hammer :cry2:

Originally posted by Pio2001
At least recording a vinyl in 44.1 and 96 kHz 16 bits on the Marian Marc 2 soundcard, I sweared that the 96 kHz captured incredibly much more of the "analog" vinyl sound than the 44.1 kHz... until all audible differences suddenly vanished under the ABX hammer :cry2:

You know, statements like this are so universal that it makes me question the absolute validity of ABX testing (which some people around here treat almost like a religion) for determining what is and isn’t important in audio reproduction. While it obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect. For example, I have seen a study that showed that PC monitor flicker had a significant effect on people’s tested reaction time even when it was well below the threshold of being perceived directly. It doesn’t seem to me that unlikely that the auditory system might not have some similar characteristics.

The clash between the objectivist and subjectivist viewpoints are pretty obvious in this thread, and it also seems like there is very little constructive communication between the two camps. It could be that the truth lies somewhere in that sparsely populated middle ground.

Subjective analysis s very important for unearthing audio problems. Noone can argue that if you can't measure something or can't ABX it, then, ergo, it doesn't exist. The Romans could not measure that the Earth was a sphere, but we now know it is (except for some oddballs).

However, what has to be done, is to put the acid on those that claim to hear something for a certain reason. That is, they must do more than make the assertion - they must provide some reasoned proof, with whatever methodlogy they can. The testing must start with a null hypothesis, there must be a control, each variable should be accounted for, the method must be described, the results must be repeatable, etc, etc.

Byant, I'm sure you are not arguing that the scientific method is flawed, are you? (Don't read this as me implying that the double blind test = the scientic method.) There is no middle ground when it comes to the results. You can use all the intuition you like to design the experiment, but the final proof must be based on cold, hard logic.

Originally posted by bryant
obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect. For example, I have seen a study that showed that PC monitor flicker had a significant effect on people’s tested reaction time even when it was well below the threshold of being perceived directly.  It doesn’t seem to me that unlikely that the auditory system might not have some similar characteristics.

Then, it could be ABX'ed too. Maybe not with short tests, but with longer tests. Same applies to auditory system. If you can hear, feel, find any difference in any way, then it could be perceived also in a blind test. The only purpose of blind testing is to remove listener's expectation effect or bias, and make possible find differences only related to the sound.

You know, statements like this are so universal that it makes me question the absolute validity of ABX testing 

Or question the validity of those statements no matter how apparently universal they are. I mean, they are "quite" universal for most people, but not so universal for more technical audio educated people.

Originally posted by bryant

While it obviously is useful for determining what is directly perceivable, it is well known that stimuli below the threshold of perception can still have an effect.

As other people have already pointed out, if they have an effect, they can be ABX-ed. What ABX does, is measure if, in any way, a listener can notice a difference between two clips. This means that the listener doesn't have to perceive anything wrong with either individually, or perceive (in the common sense of the word), a problem. If in any way they can determine a difference, they will be successfull.

A simple example is ABX-ing a lowpass based on a felt lack of 'air' .

If in no way they can determine a difference, the clips _are_ 'identical' to them for any means and purpose.

As Ruse pointed out, ABX is not the only way to work. Any scientifically valid method is good. The reason why we always refer to ABX tests is that they are easy to perform and analysing the results is well-understood. For example, the large scale listening tests we perform don't rely on ABX to get results, but the analysis is significantly more complicated than that of an ABX test.

--
GCP

It is often more difficult to ABX a difference than to hear it.
One of the only people that managed to ABX a 16 bits dithered file vs a 16 bits truncated file (while some other heard a difference but couldn't ABX it) performed one try each day ! Making a difference 16 times in a row can be very difficult.

When an ABX is positive, it does prove that the difference is audible, but when it's negative, it doesn't prove that there is no difference, it proves that the difference is "quite" inaudible.

Originally posted by Pio2001
When an ABX is positive, it does prove that the difference is audible, but when it's negative, it doesn't prove that there is no difference, it proves that the difference is "quite" inaudible.

This is false.

Proof: ABX a file against itself.

Failed ABX does not prove there is a difference.

--
GCP

Originally posted by Pio2001
It is often more difficult to ABX a difference than to hear it.
One of the only people that managed to ABX a 16 bits dithered file vs a 16 bits truncated file (while some other heard a difference but couldn't ABX it) performed one try each day ! Making a difference 16 times in a row can be very difficult.

...but it's perfectly possible if you hear a difference. Your argument is an example.

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GCP

I think Pio is right. If you get a positive result, then you've proved you can hear a difference. But not being able to get a positive result doesn't mean you can't hear a difference, it only means that you've not proved you can hear a diference reliabily, so far.

Edit: after rereading, I think we all are saying same thing.

What an interesting thread!

Rather than posting in great detail about this subject, I really should write a web page about my thoughts and experiences Re: DVD vs SACD vs CD - it would save me repeating myself. Unfortunately I wrote in detail about this at r3mix.net and that has now been wiped, so I can't refer to it.


On the other subject discussed here, I believe wholeheartedly that for someone to demonstrate that they can hear a difference between two audio samples, they must be able to differeniate them even when they do not know which one they are listening to. That is the essence of a blind test. Sighted tests are unreliable - even when the audible differences are huge.

However, sometimes conventional ABX isn't the best way. When you listen to something for the first time, it sounds different from the second time, and VERY different the 20th time! In fact, by the time you've listened to something repeatedly 20 times, I'd suggest that your ears are very tired, and responding very differently to the stimulus from "normal" listening. Sometimes this repetition can sensitise us to artefacts, but other times it can dull our senses. Whether it helps of hinders will depend on the nature of the artefact or difference.

Pio made a good point along these lines:

One of the only people that managed to ABX a 16 bits dithered file vs a 16 bits truncated file (while some other heard a difference but couldn't ABX it) performed one try each day ! Making a difference 16 times in a row can be very difficult.

I would suggest that sometimes repetition helps, and sometimes it hinders. There are differences that it is very difficult to catch in an ABX test, but they still exist.


Let me add something else to the discussion. Two collegues in the lab were evaluating the audibility of phase distortion. Traditional psychoacoustics says that it is inaudible, but some audiophiles claim that it is audible.

The test was blind: You were played clicks with no phase distortion, and then clicks with hideous amounts of phase distortion (the same as you would get from typical loudspeakers). The clicks were in pairs: distorted then undistorted, repeated three times; Or undistorted then distorted, repeated three times. Having trained yourself, you simply had to identify which set you were listening to. You did the test five times.

The first shocking result was that two of us could reliably detect which set of clicks we were hearing each time. The second shocking result was that one of us couldn't detect the difference the next day! And then the next week, he could again! No one could figure out what had changed. The equipment certainly hadn't changed. The listener (OK, it was me!) couldn't say what had changed either. No cold or flu, no loud sound exposure, no tiredness, no fatigue.

So, that's just another thing to confound it all: In a stringent blind test, one time I could hear a difference, and another I couldn't!

(btw, with real music signals (as opposed to clicks) no subjects could detect the phase distortion, but that's not to say that no one would ever be able to detect it in real music.)


As for SACD vs DVD-A vs CD: SACD will die a very slow death, DVD-A will very slowly replace CD, but most people will buy it just for its multi-channel capability, which will be a big "wow" factor when pop music starts using it creatively.

IMO: CD does have a "sound" to it. It's very subtle, but it's a "glassy" effect. It's got nothing to do with high frequency harshess - that's poor quality convertors and lousy mastering - though both of these plague many commercial CD releases. The huge audible differences between CD and SACD are mainly due to mastering, and the large amounst of HF noise that SACD pumps through your audio system. All the people who assume that most modern DACs and ADCs are "close to perfection" haven't measured one. Finally, to my ears, I think the audible difference between a $500 CD player, and the $20K Linn CD12 is greater than the audible difference between 44.1kHz and 96kHz on very high quality convertors.

You shouldn't assume that everyone in the "industry" who is making the switch up from CD is doing it for "commercial" reasons. Most will just follow the crowd, and most big companies want the copy protection and increased revenue per disc. But some of these people live and breath music and audio, and simply can't stand what CD is doing to the music. For the rest, who can't hear a difference in a fair test, the added resolution gives much more leeway to f*ck up without damaging the sound, which will give better results to the consumer, even if those results could theoretically have been achieved with CD.


Now, is 5.1 the best choice for multi-channel music? No! But that's another story...


Cheers,
David.

http://www.David.Robinson.org/ (http://www.David.Robinson.org/)

Originally posted by 2Bdecided

IMO: CD does have a "sound" to it. It's very subtle, but it's a "glassy" effect. It's got nothing to do with high frequency harshess - that's poor quality convertors and lousy mastering - though both of these plague many commercial CD releases. The huge audible differences between CD and SACD are mainly due to mastering, and the large amounst of HF noise that SACD pumps through your audio system. All the people who assume that most modern DACs and ADCs are "close to perfection" haven't measured one. Finally, to my ears, I think the audible difference between a 0 CD player, and the K Linn CD12 is greater than the audible difference between 44.1kHz and 96kHz on very high quality convertors.

After reading this I remain with the question: what is the techical cause for this 'glassy effect' ?

--
GCP

That's the $64,000 question!

It's not even possible to answer this queston:

Is the audible "signature" of CD quality sound due to (a) a fundamental limitation of the format, or (b) current implementations of the format?


I sense that you may be thinking "Since there's no technical explanation for this, then it's all imaginary". However, the problem is that there are too many possible explanations for this, and it's difficult to prove or disprove any of them at the moment.

For example, in a real room with real audio equipment, you can measure the signal at the listener's ears (e.g. use a dummy head). And you can show that including information above, say, 20kHz in the recording that you replay over the audio system will cause different audible information to be received by the listener. If you examine closely the information BELOW 20kHz that reaches the dummy header, it will be different due to the inclusion of information ABOVE 20kHz in the recording. This is due to the non-linear effect of everything from the loudspeakers to air itself.

The problem is that it's difficult to quantify the subjective effect of this. With pure tones we can predict the distortion, and predict the audibility of it with a steady-state masking model. But what about complex musical stimuli and a good stereo recording? Here, microsecond timing differences between the ears give rise to perceived location. Also, the onset of sounds gives crucial information to the auditory system - and the transient information at the onset of a sound is very difficult to analise - certainly we don't understand quite how the ear and brain carries out this analysis.


Another example: In a band limitted system, though all signals below the nyquist limit can be accurately represented, the time resolution of these signals decreases as you approach nyquist. For example, the CD limit is 22050Hz, which means that you can store a 22049Hz signal (in a perfect system). But this signal has a time resolution of 1 whole second! That means, the signal can fade up over a second, and fade down over the next second - but any faster switching would produce harmonics over 22050Hz, which could not be stored in the system. As you move down from Nyquist, this problem falls away rapidly, but even in the high-teens of kHz, you need 1 or 2 cycles of the waveform before it responds perfectly - by which time the signal onset is long gone, and your brain has done its processing.

You can look at the implicit lowpass of the ear, and the bandpass induced ringing of the auditory filters and say "it can't matter - the ear smears the signal anyway". But then you can look at the response of the inner hair cells, and realise that the ear tries to regain the time smearing - and if you're looking at a wide-band rather than tone-like stimulus, it does this very very well.


So, there's two reasons why higher sampling rates might sound better (or at least different!): equpiment distortion, and insufficient time resolution. But the real reason they sound better may have nothing to do with either of these: it may be just that they move the essential real-world compromises further away from the audio band.

Cheers,
David.
http://www.David.Robinson.org/ (http://www.David.Robinson.org/)

Originally posted by 2Bdecided

Another example: In a band limitted system, though all signals below the nyquist limit can be accurately represented, the time resolution of these signals decreases as you approach nyquist. For example, the CD limit is 22050Hz, which means that you can store a 22049Hz signal (in a perfect system). But this signal has a time resolution of 1 whole second! That means, the signal can fade up over a second, and fade down over the next second - but any faster switching would produce harmonics over 22050Hz, which could not be stored in the system. As you move down from Nyquist, this problem falls away rapidly, but even in the high-teens of kHz, you need 1 or 2 cycles of the waveform before it responds perfectly - by which time the signal onset is long gone, and your brain has done its processing.

I don't understand this, can you elaborate?

For example, you say 'but any faster switching would produce harmonics over 22050Hz'. What is the problem of that? If you are sampling at 44.1khz, you are not expecting to store (or hear) anything above 22050Hz anyway, so I don't understand what the problem is

--
GCP

Originally posted by Garf


I don't understand this, can you elaborate? 

For example, you say 'but any faster switching would produce harmonics over 22050Hz'. What is the problem of that? If you are sampling at 44.1khz, you are not expecting to store (or hear) anything above 22050Hz anyway, so I don't understand what the problem is

-- 
GCP

It's actually something I found while reading a paper to better understand nyquist and the theorem's effect on audio sampling.  2B's example doesn't quite work since filtering generally occurs at about 98.something percent of nyquist to eleminate these high harmonic issues.  Over very short periods of time the calculations become less error prone as the calculations "lock" onto the signal.  Until then you have errors that produce small amounts of distortion in the sampling process.  These occur more noticeably as you approach the nyquist frequency.

Still the assertion that the higher harmonics affect the reproduction of the signal is interesting, though depending on the equipment the effect would be fairly miniscule given most equipment's inability to produce signals over 20Khz.  Some speakers can produce frequencies up to 30Khz, but I'd imagine in most recordings these higher harmonics aren't significantly there since most mics roll-off at 20Khz, maybe slightly higher.

Here's another related question regarding high-frequency content in music: I've oft heard area audiophiles talk about how the combination of harmonics in excess of 20Khz can create a discenable "beat" at lower frequencies. Since I am a symphony musician on the side, I can attest to hearing the effects of this so called "beat" at lower frequencies. For example, when a flute and clarinet play together in their high register, and are a third apart in pitch, a harmonic resonance occurs at a different pitch than the fundamental and third. However, the harmonic is within the range of hearing. . . . so does this hold up for ultrasonic harmonics, as I have been told? Not that I would be able to hear it anyway. . . my hearing is already seeing the effects of tinutitis (sp?)--when I was a kid I could hear the 19Khz switching signal from most TV's quite loudly. Now, my hearing sharply rolls off at @16.3Khz and I can't detect anything above 17.7Khz, period, and anything above 17Khz has to be very loud in order to detect it. Makes me wonder if having DVD-A solely for the ultra-hf content would be a waste since I don't know it's there to begin with??

Originally posted by Ruse
Bryant, I'm sure you are not arguing that the scientific method is flawed, are you? (Don't read this as me implying that the double blind test = the scientic method.) There is no middle ground when it comes to the results. You can use all the intuition you like to design the experiment, but the final proof must be based on cold, hard logic.

No, I am definitely not arguing that the scientific method is flawed. In fact, my brother and I have been discussing the design of some experiments that might show this effect if it exists (he has almost completed his doctoral studies in cognitive psychology at UCSC). If we could show that there was some level of some distortion that was impossible for a subject to identify in an ABX test, but that would show an effect in some other test that did not rely on reporting by the subject (for example, a reaction time test), then that would shed considerable doubt on the value of ABX testing as the final word on audio testing.

Originally posted by Garf
As other people have already pointed out, if they have an effect, they can be ABX-ed. What ABX does, is measure if, in any way, a listener can notice a difference between two clips. This means that the listener doesn't have to perceive anything wrong with either individually, or perceive (in the common sense of the word), a problem. If in any way they can determine a difference, they will be successfull.

If in no way they can determine a difference, the clips _are_ 'identical' to them for any means and purpose.
GCP

The hearing system is incredibly complex and involves layer upon layer of information processing between the raw data that comes from the nerves in the ear and the summary report that gets delivered to the conscious mind. Certainly this process involves throwing away a lot of information and comparing the information from the different ears and all sorts of other analysis that we don't understand. And there is even conscious control over the actual processing parameters when, for example, you strain to hear something specific.

So, it seems plausible to me that there might be cases where two different stimuli could generate the same conscious result, but involve different processing to get there. It could be that the auditory system detects the stimuli differently, but simply has no mechanism to report it (perhaps because it is not a sound characteristic that occurs in nature). However, that difference in processing could still have some other effect in the mind that, even while not directly reportable, could influence our response to the music.

For a simplified example, let's assume there's a type if distortion that even when below the threshold of audibility causes some extra processing to occur. Now, let's say that after 15 minutes of exposure to this a person becomes fatigued. While one could argue that this could be detected in an ABX test if it were run slowly enough and the person could takes notes and recover between runs, I would claim that the day to day distractions of life would make this impractical. And, besides, if it could be shown that it is having some effect on the subject then it might not really matter if it could ever be directly reported.

I don't claim that any of this is actually going on, and I certainly don't believe that everything that people think they hear (but can't ABX) is real. But I am saying that with so many people complaining about the way CDs sound for so long, there might just be something to it. And I am talking about recording engineers and musicians who know what a live feed sounds like, and know what their 30 ips master tapes sound like, and bring their 30 ips master tapes home because (they claim) the final CD doesn't come close. Well, if they can't ABX it then that means that either they're hopelessly deluded or the mind is just a little more complex than we previously thought.

This is getting thick. Here are three topics :

1-Time inaccuracy of high frequencies
2-Effects of ultrasounds
3-data transmitted by the ear but not percieved.


1-Time inaccuracy of high frequencies

The time inaccuracy of high frequencies is just a mathematic trick.
A pure frequency doesn't exist in reality. In maths, a given frequency is represented by an infinite sine wave. It never begins and it never ends. When you start and stops the sine wave at given times, limiting its lenght, the accuracy of it's frequency necessarily decreases.

A 22049 Hz frequency is defined with 1/22049 acuracy.
It's higher than 22048 and lower than 22050. Therefore to represent it with such accuracy, you must be able to draw three graphs for 22048, 22049 and 22050 Hz that are different from each other.
But a 22049 sine recorded at 44100 Hz produces a beat effect of 2 Hz (try in SoundForge or CoolEdit) :



I guess that the one second lenght (or 0.5 ?) is in fact the amount of data needed for the DAC to reconstruct a pure 22049 Hz tone with no beat. In real life, I don't think oversampling algorithms used in DACs use so much samples 


2-Effects of ultrasounds

There is no beat effect between audible and inaudible sounds. You can check it also in a wave editor. When you lowpass a beat effect between the two frequencies, the beat disappears. Only the lowest frequency remains with no beat at all.

The question is in fact not beat effect but intermodulation distortion.
When you play two frequencies A and B, the intermodulation are two frequencies at |A-B| and A+B. Therefore a 6 kHz square wave, which consists in 6 kHz sine + 18 kHz sine + 36 kHz sine etc, should give audible intermodulation at 12 kHz if it is distorded, therefore sounding different than a sinewave of the same frequency.
Nika and I tested this. All we could get was distortion in the playback system (in fact I actually got some, and I suppose Mike Richter god some too). We couldn't get such distortion in our ears, that would have proven the effect of unaudible frequencies on sound.
There were also tests from Mike Richter and Studioman adam too, but they didn't lead to conclusive results.
For more : http://www.musicplayer.com/ubb/ultimatebb....3;t=000822;p=33 (http://www.musicplayer.com/ubb/ultimatebb.php?ubb=get_topic;f=3;t=000822;p=33)
Scroll down until Nika's third post, beginning with "Pio, Thank you for responding."


3-data transmitted by the ear but not percieved.

That's true. Learning music, we get trained to hear for example each separate tone from which a chord is made. Without having learned music, the ear gives the same data to the brain, but we perceive just one rich sound.
Being more trained we can recognize fundamental and harmonics in one note played by one instrument, while it appears as just one note for untrained people.
Another example could be the recogniton of two very close notes. Learning the rudiments of music, we try to recognize two notes separated with 1/18th of a tone. People learing more end up trying to sort out notes separated with 1/100th of a tone ! In this case, however, I wonder if the ability grows with learning. I was always able to hear 1/18th of a tone from the beginning since I learned music when I was 15 years old.