24-bit/192kHz is pointless?

[PeterSt]

May I ?

 

Square waves as synths can produce them.

Oops.

 

We may actually wonder what they did (or still do) to prevent them coming into the grooves.

And don't say it will go through analogue equipment during mastering. That would be the most unfair (to reality).

 

Peter (nos convict)

 

[shenzi]

The one time I heard a SACD disc via electrostatics, at a hifi show, it took me a little while to realise that it did sound more relaxing to listen to. I think I was working less to hear the HF. It was a test disk so nothing very exciting musically - jangling keys featured, I remember. However, although I thought I heard a difference, I am quite willing to believe it was to do with filters and nothing to do with increased bandwidth.

 

I was so glad to abandon the LP. In the days of analogue, give me open reel any day.

 

[iansen]

Ashley,

 

I have re-read my original post and kan see no reference, either implict or explicit, to the superiority of vinyl over digital. I have been following the thread and was konfused as to how vinyl could be described in “bits”.

 

I now know that you are using it to refer to dynamic range. My kwestion was genuine, not rhetorical.

 

How does that make me a flat earth, vinyl supremacist?

 

Now, if I can just get back to my “military spek” power supplies (komplete with kables best suited to kold war surveillance applikations in 1950s East Berlin)… oh yeah, and spell everything with a “k”!

 

 

 

[Ashley James]

iansen I apologise profusely if I managed to suggest you were anything more than asking a question that I thought demanded a helping of Amateur Dramatics to answer!

 

My answer was general and an anticipatory attack on anyone who might soil this temple to the cause of common sense and science based audiophilia buy mentioning something that sane people threw in canals or lobbed over hedges on the sides of A roads in the early eighties when they acquired a CD player. Now these should be thrown away too in our quest for music and a speedy route to it.

 

I love getting the best I can from sixty year old cars, I love riding my motorbike and recently I've started to enjoy the company of cyberfriends on forums like this. However I've always loathed and resented the difficulties involved in firing up music playing equipment, selecting a record and putting it into something before I could hear it.

 

I was a deprived child from a family of well meaning lunatics, which meant among other things, not getting a TV till years after all my school friends. Compensation, in my parents eyes, was a choice of children's books published fifty years earlier at the turn of the twentieth century. In one of these we (my two brothers and I) learned that children regularly put on plays for their parents. In our case these had to be accompanied by music from Aunty Hilda's Gramophone that required winding up and feeding with 78's. Sadly there wasn't room backstage and it ended up the other side of a stairwell where my parents were required to sit and watch. With all three of us trying (and failing disastrously) to act, it was a major inconvenience to have to descend from our makeshift stage and climb over to the machine to change a record and wind it. The parents often did a bunk before this service was needed, but as budding actors in full flow, we felt compelled to continue. And here things became difficult. We were all Prima Donnas, no one wanted to stop acting and start winding, with the result that things often got "ugly'. Fortunately the parents usually returned in time and injuries were minor, but I now realise lasting mental damage was done!

 

In more recent years I've discovered that one should look to one's formative years for an explanation as to why we may behave as we do now. This has convinced that my irrational loathing for vinyl can be traced back more than fifty years to Aunty Hilda's high maintenance Gramophone and the difficulties in deciding who should perform the tasks involved.

 

As is customary in modern reviews I feel honour bound also to tell you the music that we played, it was: Finlandia or a bit of it, the Teddy Bears Picnic, the Post Horn Gallup and Dame Clara Butt singing Jerusalem. It was she who Sir Malcolm Sergeant described as being audible in France while performing in London. She was a patron of the arts, so allowed to sing, even though she was pretty grim.

 

I'm sorry to have slightly digressed, but I thought with a full explanation, you might forgive my tirade.

 

Yours regretfully

 

Ashley

 

[glancy]

Do not forget, it is about the Music! not the technology. the Duke was right "If it sounds good, it is" I have been a recording and mastering engineer almost all my life. I recorded my first album at 14. I had my Mon drive me to Columbia records to get it manufactured.

I will never forget the first CD I listened to in the studio. I just said, you have got to be kidding, this is not a sonic improvement. A lot has changed since then.

I just received RR's SACD Tutti. I can now compare the same selection from a very fine analogue recording of "Dance of the Tumbler", the HDCD at 44.1, the SACD DSD, and the HRX at 176.4.

The biggest difference is between HDCD and SACD, the difference between SACD and HRX files is another veil being lifted between you and the preformers!

The Duke is right!

 

[mpmct]

The biggest difference is between HDCD and SACD, the difference between SACD and HRX files is another veil being lifted ...

 

 

Golden Ears,

 

Do you mind sharing a description of your system components,

all the way through to, and including, your transducers?

 

 

 

 

 

[glancy]

It is about the music--- Dan Lavry commits are theoretically correct, but my ears tell me differently.

I have found that the best way of keeping a natural sound in a digital systems is a good master clock. I use the Apogee Big Ben. I clock together at 176.4 the, Lynx, DAC and CD transport.

My systems consist of;

Lynx16e w/ Canare AES cables

Esoteric P-05 CD transport

Esoteric D-05 32 Bit DAC

Audio Research tube Reference 3 preamp

Audio Research Hybrid HV220 power amp

Vandersteen 5A speakers

Audioquest Sky, Volcano cables

 

 

[Ashley James]

There have been various discussions and links to articles on this forum and most conclude that there will not be an audible difference between straight 16 bit material and stuff with higher Bit and Sample rates. I also mentioned experiments that we'd done that supported this opinion. Most also agree that there are benefits in using much higher bitrates during the production process though.

 

The HDCD digital filter was better than a standard one and would have reduced out of band nasties, so might have made a difference with some amplifiers.

 

IMO many of these discussions are deeply flawed because they don't take account of all the factors that affect the sound quality of the recordings we all buy. I recently purchased a DVD with 24 Bit WAVs on it, it didn't sound at all good and it was easy to see why. It had been made on analogue tape recorders using vintage studio kit and it sounded like it! Similarly I had a visit a year or so ago from quite a famous Classical Music Producer who'd recently recorded and produced an Opera on an old Neve Mixing console. In its time it was the best you could buy, but compared with the modern equivalent, it was hissy, harsh and aggressive and I though the recording unacceptable. I would have been angry if I'd bought it.

 

The fact is that 15 years ago it cost about £1/2 to £1 Million to equip a studio that sounded very much worse than a Macbook Pro, some outboard gear and a few Mikes do now for less that £10K.

 

The differences you are all hearing and attributing to clocks, DACs etc is almost certainly not due to them, they will be as a result of differences in recording caused by a myriad of factors that you are unlikely to be aware of.

 

Ashley

 

 

[Purite Audio]

'there will not be an audible difference between straight 16 bit material and stuff with higher Bit and Sample rates.'

Get some decent speakers Ash then you might be able to hear the difference, honestly you do come out with complete bs .

 

[shenzi]

So nice to see a well-reasoned argument.

 

Unless you know and have eliminated all the other variables, you cannot be sure that what you are hearing is a difference in bandwidth alone. It seems to me that Ashley and his company are doing more testing and being more open about the results than a lot of people in this industry.

 

[darrenwm]

I'm undecided.

 

I appreciate that 16/44 is about the limit of human hearing. 16 bits allows a dynamic range of 96db, and a sampling rate of 44.1kHz will capture the statistical human hearing range of 20Hz-20kHz.

 

However, what are the other variables involved? Will a higher sample rate that takes twice as many "snapshots" capture musical information that has an audible benefit? Do frequencies above 20kHz, which are statistically "inaudible", have an affect on frequencies within the audible range?

 

My high-res recordings sound fantastic, but they still do if I send them via an Airport Express, which reduces things to 16/44...

 

The recordings are very good in the first place, and this is the probably the biggest variable.

 

Darren

 

[PeterSt]

It seems to me that Ashley and his company are doing more testing and being more open about the results than a lot of people in this industry.

 

With the conclusion that there are no diferences, unless because of the recording ?

Yeah, so we have one recording, two (DAC etc.) setups, and if we hear a difference it is the recording.

 

The fact that I hear differences all over makes me a fraud, right ?

Ashly, you could read back in this very thread about some various explanations, proofs, outlays, that the differences just are there, and if one can't hear them one may wonder what is going on with equipment, ears and everything involved.

 

Besides all, allow me to say that I don't like charismatic expressions without proof and reasonable argumentation.

 

Me, myself and I don't hear any differences ever. You better believe me. If you hear them anyway, it must be you.

 

This is an insult to those who hear the differences. They proceed though. Or have a chance anyway.

Peter

 

[glancy]

Me thinks we have a non-beliver in our midst----and you know what happens to non-belivers in England!

Open your mind and trust your ears!

We as an industry knew 16/44 was a problem as soon as it was released. This is not news. Therefore because of our standard 24 bit 88.2 or 92 is our next step.

Lets work together to make the playback of music in our homes represent the porformance the best we know how------

 

[shenzi]

The research referred to in the Audio Critic article (linked above) cites double-blind testing conducted over more than a year and concludes that "there’s no audible difference between the original CD standard (“Red Book”) and 24-bit/192-kHz PCM".

 

However, it is also quoted as stating: "It should also be pointed out that more bits and a higher sampling rate in recording are still a good thing because they permit a little bit of unavoidable sloppiness, so that you can still comfortably end up with 16-bit dynamics and 20 kHz bandwidth. Meyer and Moran do not say that 14 or 15 bits in a truncated CD are just as good as 20. What they say is that spot-on 16-bit/44.1-kHz processing is as good as it gets, audibly."

 

Which I think is where Ashley is coming from and may account for the differences people are hearing.

 

Extraordinary claims require extraordinary proof and all that. Eliminating all the variables is very hard work and few of us have the facilities, time or money to do more than satisfy ourselves that we think we can hear a difference.

 

[Ashley James]

What Shenzi has quoted covers it really. Coops you made it personal when what I said wasn't.

 

The fact is that if you were to own an original CD player now ( I heard an original 14 Bit Philips the other day) you'd be amazed to discover that it actually sounded excellent. Dull and not as clear, but not unpleasant.

 

The problems some people heard with their CD players were as a result of poorly designed or old amplifiers responding badly to out of band artefacts the digital process produced. Above 20 kHz, Philips original 16 Bit digital filter only reduced these to 30dB below the level of the music and this was what caused the problem. In the early nineties we used to do Musical weekends in the Lake District and we'd use our original Philips based DAC and plug it into our Amplifier to show that it was better than the CD player it was using as a digital source. Next we'd plug in into another make of amplifier from a Company that was squealing from the roof tops that CD has been introduced too early, it sounded dreadful and we should buy a certain turntable instead. It did sound dreadful for the reasons described above and because the amplifier in question was a twenty year old design!!!

 

This example was the worst I can think of, but later on when Philips introduced a Bitstream convertor, the magazines decided it was far better. But it wasn't. Its Clock frequency was very close to the IF frequency of FM tuners and it radiated strongly enough to stop any working within 100 feet! Meanwhile Sony introduced the CDP555ES, which had the Philips 16 Bit Crown S1 DAC and Sony's far better Digital filter, so was the best sounding of its day. Sadly prejudice against Japanese products and bias towards Britain's own "Digital Expert" rather clouded reason.

 

Mart made the point at the time that if the "Measuring" reviewer had a cheap scope switched on when he did his magic measuring, he'd have seen the screen dancing in all directions!

 

What this means is that CD was stunningly good from day one, that early digital recordings made with the original Sony PCM701F still sound excellent, even without Apogee's add on anti aliasing filters having been used. Try Nimbus CDs, especially the Hayden Symphonies made in the Esterhazy Palace.

 

 

Some decades ago two Scientists called Fletcher and Munchen, did tests to discover how loud a source had to be to appear to have a flat amplitude response to us. They found that our ears are not sensitive to low frequencies, but they are as it rises, by 3 kHz, they've reached a significantly higher level and from then on it reduces again and quite steeply.

Therefore it is not surprising that musical instruments don't generally go much higher. There is a whistle in some Church organs that reproduces about 8kHz, but the highest note on a violin is 3.2 kHz. This is why for the first 20 years or so, Radio only had a 3 kHz bandwidth. In the early fifties FM extended this to 15 kHz because they'd realised harmonics and overtones extended further and because telephone relay amplifiers had a 20 kHz bandwidth to avoid phase problems when lots were used in series.

 

This means that none of us hear much above 5 or 6 kHz when we're listening to music because there isn't there and our ears aren't sensitive enough to hear it if there was. Therefore the 20 Hz to 20 kHz is actually beyond human hearing. Although you feel very low frequencies, you don't hear them either.

 

There are sound scientific reasons to use a greater bitrate in the production of music, but no one should expect to hear this. Just because a 24 bit recording sounds better than a 16 Bit one it doesn't mean it's because there are 8 more bits. There are many other factors that could make the difference and there aren't enough 24 Bit recordings around yet for us to be able to judge.

 

I do hope this makes good sense.

 

Ashley

 

[PeterSt]

Therefore it is not surprising that musical instruments don't generally go much higher. There is a whistle in some Church organs that reproduces about 8kHz, but the highest note on a violin is 3.2 kHz.

 

Hi Ashly,

 

You are talking about the base tones here, and seem to forget about the harmonics.

 

In the early fifties FM extended this to 15 kHz because they'd realised harmonics and overtones extended further

 

No you didn't forget. :-)

 

This means that none of us hear much above 5 or 6 kHz when we're listening to music because there isn't there and our ears aren't sensitive enough to hear it if there was.

 

Or do you ? now I'm confused.

 

Anyway, hop over, and I'll unplug everything but my tweeters which (proved by measuring) won't produce a thing under 7500 Hz. If you don't hear that, ok, but stop (your) theories there.

Might you hear as much as I do (and in fact I am sure you will) ... then I don't know what this is all about.

 

Besides all, I think I said earlier in this thread that some people seem to confuse a higher sample rate with a higher frequency. Although this is true (88K2 would allow for 44,100 Hz) *this* is non-sense hence totally unrelated (apart from shifting nyquist frequencies and things). I said it earlier : this is about resolution. A phenomenon equipment / speakers easily can't cope with.

Besides that, you have to learn to hear it.

 

It has all been said already ...

Peter

 

 

 

[glancy]

I could set-up a blind test that would confirm 16 bits is enough.

I could set up a blind test that would establish 24 bits is a lot better.

I have set up a test that esablishes the 24/176 is modestly better sounding the 24/88

But I don't build equipment. I just enjoy recording and listening to the best representation of the proformance.

The industry has been recording for a decade at 24 bits.

Ashley I notice you build an SACD player and you do not hear a difference?

A lot of people are happy with Itunes, but I do not have to be.

long word lenghts and high sample rates are here to stay. rejoice!

 

[Ashley James]

Clancy we've never built an SACD player but we have listened to SACDs played on a standard machine and on a dedicated one and heard a difference. However the CD in question's sleeve notes clearly stated that the producer in question didn't like digital and so this was a completely analogue production! Therefore any difference between the two versions was a direct manipulation by the Record Company who would like to have sold us another licence as the original Red Book one had run out.

 

SACD failed because to use it meant buying a license and paying royalties for every machine sold and we like most manufacturers saw no benefit in it.

 

If you'd read some of my previous posting you'll see that I'm very much in favour of advancing technology and improving sound quality, just as keen as I am to respect what is scientifically possible and to support what I and others think they can hear with cold, hard, scientific fact.

 

During recent experiments we set up some tests on a prototype DAC, we rigged it so that we could switch sample rates from 32 kHz to 192 kHz whilst playing some 24 Bit material. Switching was done using an IR handset and outsiders were asked to compare the different settings and pick out the best sound. None could hear any difference despite the fact that the 32 kHz rate reduced the bandwidth to 16 kHz! We did some other sneakier tests I shan't tell you about too!

 

As I keep saying there are benefits in 24 Bit recording and production, but if these recording are reduced to 16 Bit for sale I don't think you'd hear a difference. I may be wrong, but it's much too early to judge because there isn't enough material out there yet for a representative sample.

 

We haven't made a CD player for nearly two years and the last one used twin Wolfson WM8740s with AD797s in the analogue stage. It sounded good, we know that because that's what customers and agents have told us.

 

Ashley

 

[PeterSt]

To add some confusion, let me repeat what I said earlier in this thread :

 

Personally I think 16/44K1 sounds better. So far ...

Why ?

Because it is always apples and oranges because of matters nobody (here) seems to be interested in.

 

One (repeated) example :

The best sound is perceived with the smalles buffer size possible. Besides this is general, this is true too for XXHighEnd. And, obviously I know EXACTLY what is going on in there, opposed to all the guessing people have to do with Foobar et al. So let me be unique to that aspect.

Now, when I want to play at a higher sample rate, the player (none of them) can't keep up the small buffer size (of mind you, 48 samples). Increasing the buffer size alone makes the sound "bad".

 

Read it again, because it says that comparing apples and apples would need to compare with the larger buffer size for both situations. Is that useful ? NO, of course not. It will make red book worse because of it.

 

Guys, this is a fact. And because of this fact nobody will be able to compare in the proper way.

Also, anyone comparing anyway (recognizing that he doesn't change a thing at switching) listenes to the wrong red book version.

 

Conclusion (if you're still with me) ? anyone switching (and not adjusting) between e.g. 44K1 and 88K2 should be able to hear a better result from 88K2 (but think of the recording and it's base like Ashley pointed out). Although this finding would be correct, it doesn't say a thing about the absolute differences, because you'd have to switch (buffer size) and THEN things are apples and oranges again.

 

44K1 works out better, because the net result of the buffer size being smaller works out better.

 

For those who don't hear differences between DACs anyway, skip this.

 

Peter

 

[haiku1]

A few years ago I really wanted to embrace SACD and DVD-A, so I went and purchased a Linn Unidisc Player. But everything I bought left me disappointed. DVD-A (proclaimed to be THE HiRez format) has been the worst. I`ve bought ELP`s "Brain Salad Surgery", Deep Purples "Machine Head", both sounded a lot worse than my Original Vinyls from the `70s. So for me HiRez is a complete rip off.

 

[mpmct]

My systems consist of;

Lynx16e w/ Canare AES cables

Esoteric P-05 CD transport

Esoteric D-05 32 Bit DAC

Audio Research tube Reference 3 preamp

Audio Research Hybrid HV220 power amp

Vandersteen 5A speakers

Audioquest Sky, Volcano cables

__________________

Golden Ears

 

OK, so roughly... $60,000 for that gear, MSRPs,

from what I can find of prices with a quick search. ( gulp )

Yes? If yes, I would suggest that context is, um ... kind of an

important bit of info to include in your commentary. :-)

 

 

[Cavaille]

Let´s face it people, if someone wants to achieve a certain outcome in some test (even double-blind-testing) it can always be achieved. For example, if you want to show, that 16/44.1 sounds exactly like analog you could easily prove that. Every statistic can be tweaked or interpreted this way. It is what politicians are doing all the time. Everyone will interpret statistics in a way that serves him and his intentions best. So I always have doubts about certain statistics...

 

Peter, you wrote, that a high buffer size decreases sound quality? Did I get this right? If so, I´m really interested how, because I´m getting better sound with high buffering size. Indeed I achieve the best sound when the whole file is buffered in my RAM. I achieve the same effect with foobar2000, WinAmp or other players that offer a configurable buffer size.

 

Ashley, there are a lot of studies (here we are again with studies *lol*) that show, that people actually don´t hear frequencies over 15 kHz or less. But in fact they do notice them. Our Ear, or better, our body does notice a lot of things. Now it´s getting psychoacoustical: our ear has the ability to hear much much more - but our brain neglects this information as useless, it uses a lot of processing power to eliminate unnecessary information. But what if these informations are already missing before our brain kicks in? Does this produce a stress situation because the brain is not used to NOT erase this? Or is it actually easier for us? Nearly every instrument produces high amounts of harmonics, an organ can reach up to 40 kHz, brass or violins are going even higher. The major amount of percussion frequency lies beyond 20 kHz.

 

Personally I like knowing, that these frequencies are still there. My equipment can reproduce them and with an age of 33 I´m still able to hear frequencies at 19 kHz. But I´m even more interested in a perfect impulse response - and the impulse response of 44.1 kHz is far from perfect. I agree, that you can squeeze the sound of 24 Bit into 16 Bits. And for sure there are many recordings from TELARC from the late 70´s that even today are sounding exceptionally good (the SOUNDSTREAM recorder was a wonderful thing). A skilled engineer can do a lot of good things with both, but the headroom is much better with 24/96.

 

[PeterSt]

Peter, you wrote, that a high buffer size decreases sound quality? Did I get this right? If so, I´m really interested how, because I´m getting better sound with high buffering size. Indeed I achieve the best sound when the whole file is buffered in my RAM. I achieve the same effect with foobar2000, WinAmp or other players that offer a configurable buffer size.

 

The buffer I am talking about is not the buffer from Foobar et al. It is the soundcard buffer (the one passing through SPDIF) which sounds better at 48 samples vs. the 96 needed for hirez (or 128 for DXD).

Besides that XXHighEnd (being a "memory player") works without a buffer similar to the other players. That is a.o. a reason why even Direct Sound can be used with 48 samples soundcard buffer size ...

 

Btw, I fully agree with your TELARC remarks.

Peter

 

[shenzi]

"if someone wants to achieve a certain outcome in some test (even double-blind-testing) it can always be achieved"

 

No, that's the whole point of double-blind testing. It doesn't matter what the experimenter is aiming for, he or she has no idea of the item under test either. And the nature of a rigorously set up scientific experiment is that all the methodology is exposed so anyone wishing to argue with the techniques or replicate the experiment to check the results, can do so.

 

But I don't think we're going to get a consensus on this one.

 

[Cavaille]

No, that's the whole point of double-blind testing. It doesn't matter what the experimenter is aiming for, he or she has no idea of the item under test either. And the nature of a rigorously set up scientific experiment is that all the methodology is exposed so anyone wishing to argue with the techniques or replicate the experiment to check the results, can do so.

Even a double-blind can be influenced by selecting the participants beforehand. If you´ll choose inexperienced people they for sure will tell "It all sounds the same to me" or they will try to hear something that maybe just isn´t there. You see, where I´m heading? Not only statistical data can be intepretated differently, no, even the setup of such an experiment is important. But the most important point is "Who is paying the bills?" In order to receive more financial help, funds etc. you certainly want to make this person happy.

 

The buffer I am talking about is not the buffer from Foobar et al. It is the soundcard buffer (the one passing through SPDIF) which sounds better at 48 samples vs. the 96 needed for hirez (or 128 for DXD).

Peter, how can I change this value? I´m not using my SPDIF interface, can I change it nevertheless?