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Ajax, Not sure if this is going to be useful ...

(I did not read the "report" of John Siau to remain objective)

 

I suppose I won't comply because of having the 16/44.1 act together (if all is well, that is). So say that all is explicitly optimized for that and I am not  bothered by HiRes at all. Say it is not necessary ...

With this in mind:

 

HiRes always sounds totally different, and usually not even for the better. The real explanation for that I don't have, but it could well be because I just don't focus on it. All I do know (and this could be important for context) is that ever back I started the DAC project (NOS1) to explicitly create a comparable apples to apples situation. This means, for example, that the DAC itself always runs at the same speed (say always at 32/705.6 or 32/768 depending on the base material). This is crucial for myself because a difference there would imply different electrical behavior and this would make things apples and oranges.

On a side note, and 10 years of more experience later, things still can't compare for 100% real, because there's a lot more file I/O involved for HiRes, which even is so when all runs from memory (we'd still call that file I/O because inherently (C-code) it is).

 

After this somewhat more technical babble, thus Yes, there's a difference discernible and it is totally easy to hear. It is just the character of the sound. Btw, this just as easily shows faked HiRes, because that does not show that character.

 

I suppose (but did not try to interpret for 100%) that this could be opposite to what you expect for responses. Thus in my case

a. I don't comply because of a too "good" system and should not respond in the first place;

b. but I find the 16/44.1 generally better sounding (with no good explanation).

 

Ad b.:

The majority of people will tell you the other way around (HiRes sounds better) but about as many people will work with fake(d) material without knowing it. But then of course the faked material *is* different and who knows may be better sounding than original 16/44.1. This is not strange, because I too (who not, these days) use upsampling (to 705.6 or 768) for the good reasons we can think of. Now careful because what I use for that is a. not very common (Arc Prediction in XXHighEnd) and b. all of my customers (with maybe one exception that I know of) do not like Hires for the really better as well. Or better put: they don't care a hoot like me. 16/44.1 is good enough, and the chance to run into real HiRes is too small (and the repertoire of Aix may not suite you).

 

Peter

 

 

 


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2 hours ago, PeterSt said:

It is just the character of the sound.

How would you describe the character of real HiRes sound compared to redbook?


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9 hours ago, PeterSt said:

 

 

..... 16/44.1 is good enough, and the chance to run into real HiRes is too small ....

 

 

 

 

Hi Peter,

 

Thanks for your thoughtful response.

 

I have always known that you have considered 16/44.1 adequate when upsampled in your XXHighEnd software prior to converting in your NOS1 DAC. MIska promotes a similar philosophy with his HQ Player as does Damien with Audirvana+ software.

 

John Siau states there are only 3 things that can benefit from hi-res, none of which he sees as being practical: 

 

1. An increased high frequency limit - majority of equipment can't produce frequencies over 22khz, and we can't hear over 22khz, so why bother.

 

2: An increased immunity to the clipping of inter sample peaks - this can be overcome by upsampling in the player software prior to converting in the DAC.

 

3: An increased SNR - you would have to play the music at an uncomfortably loud level to gain only a very minimal benefit.

 

The bottom line is that here is really little need for hi-res material (and in deed there is a very little "true" hi-res' available as most offered today is upsampled music recorded on tape from an era when hi-res was simply not available (pre 85).

 

My interpretation of Mark and John's blog, and everything I have personally heard is that when material is well recoded in hi-res (24/96) and then expertly mastered with dithering and noise shaping down to 16/44.1 for distribution it meet the requirements of even the most fastidious audiophile.

 


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10 hours ago, Ajax said:

I have often wondered how humans can "hear" the difference between a 16/44.1 audio file, that has been recorded at say 24/96, and expertly mastered using dithering and noise shaping down to 16/44.1, and the original 24/96 file.

 

 

The obvious reason, which accounts for most of the difference, is that the playback chain behaves differently when fed source in one format compared to another - the type of distortion heard varies, even though the musical content, from the original recording, is identical. Implementation strengths and weaknesses usually mean one format will do better than another - most of the good stuff to listen to is "no better" than 16/44.1, so the answer is to optimise for that format.

 

"Fake" hi-res is actually superior to the "genuine" article, because the signal to noise ratio of the latter is very poor in the ultrasonic frequencies - graphs showing lots of activity at those inaudible frequencies are highly misleading, because if you actually look at the waveform, closely ... it's just noise. Pure noise. No matter what's happening musically, say, almost pure silence, or a grand crescendo, there's a contant ultrasonic hiss which never changes. Which IMO is the main suspect for why "real" hi-res sounds different - the electronic circuitry is reacting to the presence of this constant, inaudible noise; the distortion spectrum is different.


Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

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Prokofiev’s “Lt. Kije”(Fritz Reiner, Chicago Symphony on RCA Victor)

 

Yes this is what is known as a good recording.  I only have it in the CD, but it is excellent.  

 

 


To paraphrase Rick James, "sighted listening is a helluva drug".

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12 hours ago, rickca said:

How would you describe the character of real HiRes sound compared to redbook?

 

I just upvoted Frank's and George's post because I feel the former leads to the latter.

 

53 minutes ago, gmgraves said:

The Redbook is exciting, rhythmic, and jaw dropping-ly gorgeous sound. The SACD is bland and ordinary.

 

So exactly that but with the reason of additional noise being the suspect.

 

HiRes always seems to be too smooth and in the end (say before halfway through the album) is too boring because of too much of a same sound. Beck's Sea Change could be a good example of that. Nice silk sound but in the end not enough sparkle. And the more gray-ish (but very soft-gray) is an explicit flavor on top of that. Not explicitly gray, but something in our brain which makes us perceive it like that (a result of too silky).

 

For me it is also easily audible that with Hires cymbals are rendered better and maybe even drums in general are (they are quite easy to focus on), but the dynamics lack compared to Redbook.

 

I also realize that it is super dangerous to claim something like the latter because with the (figurative) snip of a finger it could be distortion of Redbook (illegally rendered too high frequency) what one perceives regarding more snap or tinkle or even bell(s). But careful day-in day-out observing does not make me conclude that. It would imply an all-over-sauce again, and this is not so at all.

 

Quote

but the dynamics lack compared to Redbook.

 

The danger is that looking at the (sample) data, Redbook *is* more steep on the micro transients (sample-to-sample) but proper filtering/upsampling should solve that. Thus, because of the fewer samples with Redbook to go from A to B while the time to go from A to B does not change compared with HiRes, the bends in the road are more sharp and squealing tyres are the result. Upsampling makes shortcuts to the bends and all is fine again. With some poorer tyres the sound of the touching of the tyre to the road - a high frequency happening - is also recorded with Hires. During playback we can hear that too. With Redbook, a lower frequency recording means implying better sealed windows cutting off noise from the outside; this noise is now not recorded and thus it is also no continuous same background sound when the car radio is playing and we like to listen to the music of that alone.


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1 hour ago, gmgraves said:

Nothing is cut-and-dry. I have high-res recordings that sound great and I have redbook recordings that sound so good that they beat SACD remasters of the same material seven ways to sundown. Example: the JVC XRCD remaster of Prokofiev’s “Lt. Kiki”(Fritz Reiner, Chicago Symphony on RCA Victor) vs BMC’s RCA SACD remastering of the same, exact performance. There is no comparison. The Redbook is exciting, rhythmic, and jaw dropping-ly gorgeous sound. The SACD is bland and ordinary. I have a number of examples where the regular CD outperforms the High-res reIssue, irrespective of what format; high-res LPCM or DSD/SACD.

Are you using a CD/SACD Player? 

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I think the mistake in the analysis here is that no one is directly listening to a “file”, rather the output of the electronics that translate the file into audio. 

 

With the exception of the rare 16/44 NOS DACs that perform no upsampling, the vast vast vast majority of DACs that accept 16/44 input perform some type of upsampling and/or SDM ... the benefit of higher bit rate input is that the DAC need not do extraneous processing best done by a CPU. Why not just use the resolution that the music was recorded at? Avoid extra filtering steps. 


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12 hours ago, Rexp said:

Are you using a CD/SACD Player? 

I made that call using a Sony SCD-XA777ES. Unfortunately, it won’t play SACD discs any more (I suspect that the laser has stopped working). But definitely, the redbook JVC XRCD sounds much better than the BMC/RCA SACD of the same material!

These days, I’m using an Oppo UDP-205 for both SACD and redbook playback. It has a pair of ESS SaberDAC Pros In it, and they are arguably the best Delta/Sigma (single-bit) DACs right now that sport SACD/DSD. 


George

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10 hours ago, jabbr said:

I think the mistake in the analysis here is that no one is directly listening to a “file”, rather the output of the electronics that translate the file into audio. 

 

With the exception of the rare 16/44 NOS DACs that perform no upsampling, the vast vast vast majority of DACs that accept 16/44 input perform some type of upsampling and/or SDM ... the benefit of higher bit rate input is that the DAC need not do extraneous processing best done by a CPU. Why not just use the resolution that the music was recorded at? Avoid extra filtering steps. 

Hi Jabbr,

 

I agree that ideally we should replay at the same resolution as the recording (pre digital music was mastered specifically to be replayed on turntables by limiting the base so the needle did not jump out of the grooves. Today we limit the band width of the audio frequency so we don't end up with artefacts).

 

My reasoning of recording in hi-res and playing back at 16/44.1 was simply to achieve more head room during the recording process, which with due care is obviously not essential, especially today with so much compression being added.

 

The point of the article is that mathematically 16/44.1 is adequate for music playback, IF as George points out the recording and mastering has been done with sufficient care. My hearing is limited to 12khz (I'm 63) so for me personally there is no need to record at higher frequency rates (above 44.1), however, maybe there is benefit at recording at 24 bits, however slight.

 

I went to a hi-fi show in Melbourne, Australia about 4 years ago and heard the Devialet ensemble being demonstrated using only CDs and it was truly stunned by the sound, despite the poor acoustic environment of hotel show rooms. There were lots of competing gear with massive power amps and exotic cables but nothing to my ears came close. I bought the demo system, which included Atohm GT1 speakers and it now sits in my living room.

 

However, the best sound I have ever heard was my office system, which consisted of a Benchmark DAC1 (I just purchased a DAC2 second hand for $US900) driving a pair of Adam X7 active monitors listening to Gwyneth Herbert recorded and produced by Peter Gabriel's 'Society of Sound". This was recorded and distributed at 24/48. One night about 7 years ago my then 12 years old son (now an accomplished musician) came into say goodnight and said "that's spooky Dad, it's like she is in the room with us".

 

The point is John states (and my experience confirms) that extremely higher frequency rates aren't required, especially for old buggers like me, and if you don't play your music at pain levels you also don't need more than 16 bits.

 

As John says, it is all in the maths. It is important to remember the reason why we have digital audio in the first place is because a couple of very smart mathematicians, Shannon & Nyquist, developed a theorem that simply put states that if you record at twice the highest maximum audio frequency of the music then that is sufficient for perfect fidelity and no actual information is lost.

 

The other reason for reproducing the article is that the majority of music is available at 16/44.1, whether via CD, downloads or streaming, and manufactures should therefore be concentrating on improving the playback of that resolution, preferable using a combination of software and hardware, not offering us more and more exotic gizmos in an effort to differentiate themselves from other manufactures..

 

 

 


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37 minutes ago, Rexp said:

Most CD Players sound bad to me and I've never heard a good sounding Jazz CD, who is at fault? 

 

The people who design CDPs, and those provide the advice on what is required for achieving a good sounding system - jazz recordings are not a problem in themselves, but the typical instruments used by musicians playing this sort of music can very easily make for uncomfortable listening, if not reproduced well. I don't have a single jazz CD that can't be coaxed into providing a satisfying presentation, if I do enough to resolve SQ issues in the electronics chain - and this covers something like 100 years of recording this sort of music.

 

Get it right, and the impact of the music making takes your breath away - get it the tiniest bit wrong, and you'll hate the sound of it ...

 

Hardly anyone takes enough notice of the saying, "the devil's in the details" - but that's where the answers are ...


Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Ahhh, Mankind ... Porsche intellect, Trabant emotions ...

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4 hours ago, Ajax said:

One night about 7 years ago my then 12 years old son (now an accomplished musician) came into say goodnight and said "that's spooky Dad, it's like she is in the room with us".

 

One night, about 7 years ago my then 12 years old son (now an almost EE graduate) experienced a farewell party of his class at primary school and the next morning of that (they all camped in a couple of tents in the field over here) one of his class mates came in saying thank you. we did not know her really, and saw her suspiciously look behind the somewhat larger speakers. "What are you looking for ?". Her answer: the saxophone player.

So she appeared to study for playing the saxophone and she was sure there was a sax player in the room somewhere. But it was just music playing through loudspeakers ... (and 16/44.1 :-).

 

4 hours ago, Ajax said:

extremely higher frequency rates aren't required, especially for old buggers like me,

 

That is totally unrelated. The fact that you may not hear up to 20K any more, only implies that some inherently square sounds, change timbre because the square becomes sine. So what should happen is that e.g. a bunch of keys sound different to me my son than to you. But you wouldn't know and you also would not notice the difference occurring over time. One thing though:

 

In my theory it would be so that an instrument like the violin may become more easy to play through loudspeakers because the square sound (of resin'ed bow) will have become more smooth to you and the system does not need to render that squareness. Otherwise all we hear for real square sound is at such low frequency that even when you start to wear hearing aids you will still perceive that.

Notice that the square sound I talk about will be synth originated so it requires old fools like me to play that (kind of music) to begin with. Or Play AC/DC what you may be used to, like us with that guy living maybe 40 kilometers from here. But ZZ-Top is also fine for it. Or Rammstein. Or anything with a nice distortion guitar which, mind you, sounds beautiful when rendered as should (go figure about that because that won't go without amplification, so comparing with on-stage and the on-stage will lose ...). The distortion guitar is full with air and most will not have heard that (it requires "ultimate" speed of the whole playback system).

Now where were we ... Ah right, HiRes may not be needed.


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1 hour ago, PeterSt said:

Notice that the square sound I talk about will be synth originated so it requires old fools like me to play that (kind of music) to begin with. Or Play AC/DC what you may be used to, like us with that guy living maybe 40 kilometers from here. But ZZ-Top is also fine for it. Or Rammstein. Or anything with a nice distortion guitar which, mind you, sounds beautiful when rendered as should (go figure about that because that won't go without amplification, so comparing with on-stage and the on-stage will lose ...). The distortion guitar is full with air and most will not have heard that (it requires "ultimate" speed of the whole playback system).

Now where were we ... Ah right, HiRes may not be needed.

 

Can only agree with that. Hendrix is quite sublime when replayed well - I have a copy of the Blues album here, and this is a fabulous ride of sound ... every twitch of the fingers, effects unit hissing, Marshall amp gurgling is laid bare; with a driving rhythm second to none.


Frank

 

http://artofaudioconjuring.blogspot.com/

 

 

Ahhh, Mankind ... Porsche intellect, Trabant emotions ...

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14 hours ago, Ajax said:

The point of the article is that mathematically 16/44.1 is adequate for music playback

 

That point has been made for decades. Nonetheless modern DACs upscale/convert 16/44 before the actual digital analogue conversion. 16/44 is effectively lossy compression. Arguably the lossy compression is not audible but nonetheless if a recording is made at 24/192 or DSD256 why in heavens would I want to compress it?

 

Just give me the closest to native format thank you. That’s common sense to me.


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18 minutes ago, jabbr said:

16/44 is effectively lossy compression. Arguably the lossy compression is not audible but nonetheless if a recording is made at 24/192 or DSD256 why in heavens would I want to compress it?

 

Not arguing for unnecessary compression, but 24/192 is lossy, just like DSD256 is lossy, just like 16/44 is lossy. The question in all cases is precisely whether what's being lost is audible or not. Argument can be made for 24/96kHz being perfectly sufficient for human consumption. Dogs may need 24/192, bats may need 24/384 ;)

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1 hour ago, pkane2001 said:

 

Not arguing for unnecessary compression, but 24/192 is lossy, just like DSD256 is lossy, just like 16/44 is lossy. The question in all cases is precisely whether what's being lost is audible or not. Argument can be made for 24/96kHz being perfectly sufficient for human consumption. Dogs may need 24/192, bats may need 24/384 ;)

 

24/192 is not lossy when it’s the recorded or mastered resolution. The only reason to down convert to 16/44 is for compression. Similarly when DSD256 is the recorded format (ok DXD is easier for processing)

 

Similarly I store my photos RAW even though jpeg is fine for a drivers license.


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5 minutes ago, jabbr said:

 

24/192 is not lossy when it’s the recorded or mastered resolution. The only reason to down convert to 16/44 is for compression. Similarly when DSD256 is the recorded format (ok DXD is easier for processing)

 

Similarly I store my photos RAW even though jpeg is fine for a drivers license.

 

24/192 is lossy if you consider that you lose all the frequencies above 96kHz and probably a bit below due to filtering ;)

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46 minutes ago, jabbr said:

 

Generally when we use the term “lossy compression” regarding digital data, we mean that bits are altered when round tripping between the compressed and uncompressed format. If the digital data starts out at 24/192 then it’s not compressed by that definition. Similarly for a photo which is also always bandwidth limited by the digital sensor (and lens). Lossy compression has a specific meaning: the difference between FLAC and JPEG. 

 

Of course one could always make a higher resolution recording just as one could use a higher resolution imaging sensor — but the term “lossy compression” means that bits are thrown out with respect to the digital recording at whatever resolution it was made.

 

Thanks for the explanation. I created one of the first JPEG/JFIF decoder-viewer/converters that was released as shareware, as well as worked in research on lossy audio compression in the 80’s, so I do have a little background in this field :) My post was more tongue in cheek, but your comment is perfectly valid. 

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