View Full Version : Digital Facts and realities?
Bigman80
12-03-2019, 18:52
I was given this information via a read me file, just thought I'd share it:
Digital Facts and realities:
Most music is in the 50 Hz to 15 KHz range. The Human ear does not hear from the 20 Hz to the 20 KHz range. Almost all speakers, ear buds and headphones will not produce anything outside the 20 Hz to 20 KHz range. The best tape machines will not play anything above the 60 KHz range. So unless something is recorded digitally today 60 KHz is the maximum that can be played back from a Master tape.
The sampling rate must be double the highest frequency rate. Since all of the above holds true with respect to equipment the industry settled on the 44100 for Redbook Cd’s. They will reproduce a 1Hz to 22.5 KHz range. Beyond most equipment’s capability and the human ears capability.
The way a digital to analog converter (DAC) works it creates a smooth sine wave no matter what the sampling rate is. Therefor certainly anything above the 48 KHz range is a complete waste of space. The extra room above 40 KHz is used to dither noise. Hence the 44.1 and 48 KHz used on CD and Video. So much for the SACD in the sampling rate respect. A waste of space.
The Bit rate comparison of 16 bit to 24 bit might be the only saving grace for the SACD. However it has its questionable aspects on a technical level as well. The main function of the bit rate is signal to noise ratio. Keep in mind that we go deaf at the 140 decibel level. 16 bit is capable of 100 DB and 24 bit is capable of 150+ DB. Again most equipment is not capable of over 127 DB level. A well Mastered 16 bit is actually capable of the 120 DB level and is called 20 Bit. Some of the 24 bit is actually 22 bit and the last 2 bits are zeroed out. So the difference between a 16 bit and 24 bit is just about a moot point.
Mastering is the most important aspect when it comes to sound quality. A well mastered CD is absolutely better than a poorly mastered SACD. And vice a versa. First and foremost in deciding what is the best sounding choice should always be the mastering. Keeping in mind that larger is not better. With all of the above being true in most cases it is certainly a waste. Another consideration is age of the Master tapes with a lot of music. A good master done 20 years ago is potentially superior to a re-master today due to tape degradation.
Sony is a company with one goal in mind. To make money by keeping sales high. They invented the SACD. With that in mind and all of the above facts as well as numerous testing that has been done showing that 99% cannot hear a difference at all in the SACD compared to a CD the conclusion must be that it is about sales. As far as Sony is concerned. Also if you own an SACD and do not have an SACD player you are actually getting the CD layer when listening to the actual SACD.
I would think that the perfect media would be 48000-24. Not much larger than a CD and capturing any benefit a 24 bit might have. Which of course is questionable. Certainly not noticeable until you crank it up over the 100 DB range which almost none ever do. Again bigger is not better. The first link is from the makers of FLAC…
https://xiph.org/~xiphmont/demo/neil-young.html
http://www.head-fi.org/t/356139/sacd-vs-cd-a-true-difference-or-simply-placebo
http://users.ece.utexas.edu/~bevans/courses/realtime/lectures/10_Data_Conversion/AP_Understanding_PDM_Digital_Audio.pdf
Patrick Dixon
12-03-2019, 20:51
Which bits are the facts?
Minstrel SE
12-03-2019, 22:00
Well I cant hear much above 12khz and my ears are still ringing from my last trip to rockworld so I dont worry too much :)
Even those that do say there is a difference say it can be subtle and not worth the investment for most people. Im quite happy with the cd standard rate. Most people havent tried or cant hear the difference. There are still debates that the redbook cd version sounds better.
I havent moved towards higher resolution because I have not been bothered to do it. I was waiting for Neil Young to bring the price down to mass market levels :)
I can pick up cds cheaply...Im enjoying the music and it all seems better for me than shelling out for sacd or high res files.
I dont know how much more information I need and its not been marketed to me properly. I feel that this is mainly because they know im not getting much aural difference for my money
Hi Oliver
I would disagree with your comments.
I know if you disconnected the super tweeters in your pioneers YOU would notice
Which would suggest your observations are flawed.
Steve
Bigman80
12-03-2019, 22:43
Hi Oliver
I would disagree with your comments.
I know if you disconnected the super tweeters in your pioneers YOU would notice
Which would suggest your observations are flawed.
SteveAh, not my comments mate. This was in a "Read me" file on some music files I was given. Just shared it here to see what people thought.
And yes, I notice as soon as I turn the Super tweeters off
Bigman80
12-03-2019, 22:45
Which bits are the facts?No idea, that's why I shared it!
Like I said, it's from a "read me" file I was given.
Well the second sentence is obviously wrong! :lol:
I’m with Patrick, which parts are meant to be “facts”? I can’t seperate the subjective BS from thesupposedly technical parts...
Bigman80
12-03-2019, 23:54
I’m with Patrick, which parts are meant to be “facts”? I can’t seperate the subjective BS from thesupposedly technical parts...No idea!
I've largely ignored digital music but the Raspberry Pi/Allo Boss DAC has opened my eyes to the vast library that can be created, very quickly and enjoyed at the touch of a screen.*
I just felt this was an interesting read and wondered how much was actually fact Vs fiction.*
That's why it here, to get the opinion of those who might know!
Bigman80
12-03-2019, 23:55
Well the second sentence is obviously wrong! [emoji38]Probably but that was the title of the file.
Light Dependant Resistor
13-03-2019, 00:02
Just as Genesis "Get em out by Friday" predicted, we can fit twice as many in the same building site.. they say its all right.
The quoted text is a ploy to reduce further the capability of a already compromised medium. https://earthworksaudio.com/wp-content/uploads/2012/07/The-world-beyond-20kHz.pdf
with singular aim of money making.
Now how about instead extending the frequency response of CD to 48khz with sampling then at 100khz allowing a more gradual filter. I would be happy buying CD's with this added capability.
Probably but that was the title of the file.
Nothing probable about it, this is the second sentence:
"The Human ear does not hear from the 20 Hz to the 20 KHz range." (My emphasis)
Bigman80
13-03-2019, 00:08
Nothing probable about it, this is the second sentence:
"The Human ear does not hear from the 20 Hz to the 20 KHz range." (My emphasis)Lol, ah yes. You're right!
Surely there should be a "only" in there?
Lol, ah yes. You're right!
Surely there should be a "only" in there?
Yes that would make more sense; though beyond those frequencies, humans tend to perceive the 'sounds' rather than hear them. Infra bass is felt, whereas ultrasonic sounds can be perceived, by some people, directly through the scull.
Opti-cal
13-03-2019, 09:46
I have a love/hate relationship with scientific "facts".
We use facts indefinitely to build systems based on scientific theory. They help us understand how sound travels from recording format via source and through various amplification stages into your speakers/headphones.
However these "facts" don't really explain the effects these frequencies, that apparently cannot be heard, have on your brain. You certainly do not have to "hear" the frequencies for them to have an effect on your "hearing/portrayal" (and therefore your appreciation) of the music.
As others have eluded to sub bass frequencies are felt and ultrasonic ones perceived (they can also break glass etc). Also consider the effects these frequencies have on the rest of the recording.
We may not hear them but they may have some effect on other notes that were being recorded at the same time. I'm speculating in this previous sentence but having mixed digital music in the past I can tell you that sometimes removing what technically a human ear cannot detect in a recording, can effect that recording in both positive and negative ways.
I've also attempted to "clean up" ambient, live recordings and sometimes the silence (which often has detectable noise well above 15k) when removed, again, removes something more from the recording.
Plus think how devastated our K9 friends would be if we started cutting off the extra frequencies in our favorite albums, they expect to hear them!!
As with a lot of these topics its likely very subjective but I think this is an interesting discussion.
Cheers
Chris
Opti-cal
13-03-2019, 09:49
Infra bass is felt, whereas ultrasonic sounds can be perceived, by some people, directly through the scull.
Exactly Barry, spot on.
AJSki2fly
13-03-2019, 10:04
Ah, not my comments mate. This was in a "Read me" file on some music files I was given. Just shared it here to see what people thought.
And yes, I notice as soon as I turn the Super tweeters off
Yes you noticed that you switched them off, but did you determine what frequencies were coming from them that you then noticed were not there after turning them off? :eyebrows:
Morning Oli
What are your thoughts on this?
I have read many times that higher sampling rates only increase the highest frequency. But that is simply not true. It is true that the Nyquest rate, or whatever it is, says that the highest recordable frequency is half of the sampling frequency, this much is true. But that is not the Only thing it does. Lower frequencies are also sliced into finer samples, increasing accuracy. Just because there’s a beautiful smooth sine wave coming out, doesn’t mean it’s a perfect copy of the original. The same for bit rates, each sample has a numeric representation, a number between the lowest, and highest number possible, so the more numbers, the more accurate that sample will be.
And I hear these things from designers and engineers! If they don’t have a real grasp of what is going on, does that get in the way of the success of their designs? I guess anyone can get lucky now and then?
And I also agree that a good recording versus a bad recording makes one hell of a lot more difference than any of it. A poor recording sounds bad, no matter what you do, the better the equipment, the worse it sounds.
Russell
I have a love/hate relationship with scientific "facts".
We use facts indefinitely to build systems based on scientific theory. They help us understand how sound travels from recording format via source and through various amplification stages into your speakers/headphones.
However these "facts" don't really explain the effects these frequencies, that apparently cannot be heard, have on your brain. You certainly do not have to "hear" the frequencies for them to have an effect on your "hearing/portrayal" (and therefore your appreciation) of the music.
As others have eluded to sub bass frequencies are felt and ultrasonic ones perceived (they can also break glass etc). Also consider the effects these frequencies have on the rest of the recording.
We may not hear them but they may have some effect on other notes that were being recorded at the same time. I'm speculating in this previous sentence but having mixed digital music in the past I can tell you that sometimes removing what technically a human ear cannot detect in a recording, can effect that recording in both positive and negative ways.
I've also attempted to "clean up" ambient, live recordings and sometimes the silence (which often has detectable noise well above 15k) when removed, again, removes something more from the recording.
Plus think how devastated our K9 friends would be if we started cutting off the extra frequencies in our favorite albums, they expect to hear them!!
Excellent post, Chris. I concur completely, especially with the bit in bold! And that's why in audio, forming opinions based on said 'facts' will only ever take you so far, certainly in terms of understanding how we as humans 'hear' music.
Marco.
In a different thread I linked to a scientific paper which demonstrated the ability of humans to be affected by inaudible high frequency sounds.
Here is the paper again.
https://www.physiology.org/doi/pdf/10.1152/jn.2000.83.6.3548
I'm sure that, over time, 'science' will keep catching up with various phenomena. The gap between 'objectivity' and 'subjectivity' will increasingly narrow but will always be there. Thankfully as otherwise Forums will cease to have a purpose. :lol:
I'm sure that, over time, 'science' will keep catching up with various phenomena.
That is the key point, Edward! Scientific facts, when applied to audio, are rarely a 'done deal', but simply a reflection of our current learning. And that's the crucial point: we're continually on a learning curve, which is why it pisses me off when folk apply science to audio in such as 'black and white' way, simply to try and 'prove' a point:doh:
It's also why that, ultimately, I'll always trust my senses and gut instincts in audio, no matter what the supposed 'scientific facts' say to the contrary, and that approach will only change, as you say, when/if science catches up with what's revealed by the highly-attuned auditory senses we were born with!
Marco.
Bigman80
13-03-2019, 12:01
Morning Oli
What are your thoughts on this?Hi Mr C
I have none! I really don't know anything like enough about this subject.
What I know is mainly practical listening based so when we get into the scientific explanation of sound, I am lost lol
jandl100
13-03-2019, 20:29
Oh dear
I see that Nyquist Denial is here again.
I'll just leave it alone this time and listen to some music. :violin::harp:
Oh dear
I see that Nyquist Denial is here again.
I'll just leave it alone this time and listen to some music. :violin::harp:cardinal theorem of interpolation...
Is the correct term
Oh dear
I see that Nyquist Denial is here again.
Soz, what's that when it's at home?:scratch:
Marco.
jandl100
13-03-2019, 20:48
Nothing to do with your posts here, Marco. :)
Just someone saying that finer sampling under the Nyquist frequency gives better accuracy.
It doesn't, the musical waveform can already be reconstructed accurately.
This has been addressed several times on AOS, but it keeps coming back again.
Lol... Ok, I'll file it under 'techy shit':eyebrows:
Marco.
cardinal theorem of interpolation...
Is the correct term
Some of the first work on sampling theory was undertaken by Shannon whilst working at the Bell Telephone Laboratories. He showed that to accurately sample a sine wave having a particular frequency, the sampling rate has to be double that frequency. The sampling frequency is also known as the Nyquist rate, or frequency.
Nyquist was a mathematician (also working at Bell Labs), who amongst other things, undertook the analysis of electronic thermal noise, as described by Johnson (it is thus also referred to as Johnson noise), and provided the formula for the mean square thermal noise voltage of a resistor of value R, at a temperature T[/T] over a bandwidth [I]B: 4kTRB.
Oh dear
I see that Nyquist Denial is here again.
I'll just leave it alone this time and listen to some music. :violin::harp:
There is no straw that will not be clutched at. :)
AJSki2fly
14-03-2019, 08:04
Nothing to do with your posts here, Marco. :)
Just someone saying that finer sampling under the Nyquist frequency gives better accuracy.
It doesn't, the musical waveform can already be reconstructed accurately.
This has been addressed several times on AOS, but it keeps coming back again.
I think it’s time to file this stuff and listen to the music or we will be sliding into the pit of digital inaccuracy again. :eek::D
There is no straw that will not be clutched at. :)
Does that include measurists, desperate to use some supposed 'fact' to dismiss/pooh-pooh the existence of any phenomenon that superficially doesn't appear to fit with their scientific belief system?;)
Marco.
Light Dependant Resistor
14-03-2019, 10:10
Some of the first work on sampling theory was undertaken by Shannon whilst working at the Bell Telephone Laboratories. He showed that to accurately sample a sine wave having a particular frequency, the sampling rate has to be double that frequency. The sampling frequency is also known as the Nyquist rate, or frequency.
Nyquist was a mathematician (also working at Bell Labs), who amongst other things, undertook the analysis of electronic thermal noise, as described by Johnson (it is thus also referred to as Johnson noise), and provided the formula for the mean square thermal noise voltage of a resistor of value R, at a temperature T[/T] over a bandwidth [I]B: 4kTRB.
And under the Nyquist /Shannon sampling theorem there is nothing super sacred about 44.1khz at all, rather the theorem holds true for use of much higher sampling to be used ,representing twice the upper most audio frequency to be moved upward, and I would suggest twice from its present limiting figure. https://earthworksaudio.com/wp-content/uploads/2012/07/The-world-beyond-20kHz.pdf
The benefits would be less severe brick wall filtering between 20khz and 22.05, and catering to our ability to perceive frequencies higher than 20 khz. The CD standard has had a number of developments to lengthen available space. https://en.wikipedia.org/wiki/Rainbow_Books that may enable the same 79 mins but with extended frequency.
Going back to where this thread commenced the discussion quoted but not necessarily agreed to, was on a downward spiral path to reduce audio frequency to be lower than 20 khz, as I recall 15 khz being suggested. The opposite namely a doubling of upper frequency being what is actually needed by audiophiles using digital replay. The Denon 103 cartridge as example of earlier/ present technology having frequency response of 20 hz to 45 Khz
Light Dependant Resistor
14-03-2019, 10:57
Does that include measurists, desperate to use some supposed 'fact' to dismiss/pooh-pooh the existence of any phenomenon that superficially doesn't appear to fit with their scientific belief system?;)
Marco.
Here is a great example of just that Marco,
Dc Direct current, despite being hundreds of years since its discovery has received very little research , here is the Wikipedia article about DC https://en.wikipedia.org/wiki/Direct_current
"A direct current circuit is an electrical circuit that consists of any combination of constant voltage sources, constant current sources, and resistors. In this case, the circuit voltages and currents are independent of time. A particular circuit voltage or current does not depend on the past value of any circuit voltage or current. This implies that the system of equations that represent a DC circuit do not involve integrals or derivatives with respect to time."
but then contradicting this is
"If a capacitor or inductor is added to a DC circuit, the resulting circuit is not, strictly speaking, a DC circuit. However, most such circuits have a DC solution. This solution gives the circuit voltages and currents when the circuit is in DC steady state. Such a circuit is represented by a system of differential equations. The solution to these equations usually contain a time varying or transient part as well as constant or steady state part. It is this steady state part that is the DC solution. There are some circuits that do not have a DC solution. Two simple examples are a constant current source connected to a capacitor and a constant voltage source connected to an inductor. "
We can see the discussion of DC involved with differential equations usually collapses as a pure mathematics exercise https://www.youtube.com/watch?v=CRLXDnViSyI
but is then awkwardly limited in how it relates back to the actual application in a DC circuit, rather it gets caught up in itself - happy that it has solved the mathematics
Moreover what tends to happen is the former limiting explanation "A particular circuit voltage or current does not depend on the past value of any circuit voltage or current" takes precedence. , so really exciting applications involving DC ( like I do ) get totally ignored, and commented on as being impossible. We see the conventional explanation of DC above containing explanation "the circuit voltages and currents are independent of time" An example contrary to this being the thyristor used in a DC circuit where the end of the circuit instructs the beginning, or where a gate or base of a semiconductor placed at the end of a circuit instructs the beginning of a circuit to commence.
which is then unexplained or contrary to conventional thinking. But the annoying thing is that there are devices readily available that solve a lot of problems in electronics - that is when and if they are applied. Hence the study of DC needs expanding, to place time as a real factor in DC circuits.
Ah, not my comments mate. This was in a "Read me" file on some music files I was given. Just shared it here to see what people thought.
And yes, I notice as soon as I turn the Super tweeters off
Those super tweeters will be working below 20kHz. :)
Stratmangler
15-03-2019, 13:48
Those super tweeters will be working below 20kHz. :)
Chances are they'll be audible below 10kHz :eyebrows:
Here is a great example of just that Marco,
Dc Direct current, despite being hundreds of years since its discovery has received very little research , here is the Wikipedia article about DC https://en.wikipedia.org/wiki/Direct_current
"A direct current circuit is an electrical circuit that consists of any combination of constant voltage sources, constant current sources, and resistors. In this case, the circuit voltages and currents are independent of time. A particular circuit voltage or current does not depend on the past value of any circuit voltage or current. This implies that the system of equations that represent a DC circuit do not involve integrals or derivatives with respect to time."
but then contradicting this is
"If a capacitor or inductor is added to a DC circuit, the resulting circuit is not, strictly speaking, a DC circuit. However, most such circuits have a DC solution. This solution gives the circuit voltages and currents when the circuit is in DC steady state. Such a circuit is represented by a system of differential equations. The solution to these equations usually contain a time varying or transient part as well as constant or steady state part. It is this steady state part that is the DC solution. There are some circuits that do not have a DC solution. Two simple examples are a constant current source connected to a capacitor and a constant voltage source connected to an inductor. "
We can see the discussion of DC involved with differential equations usually collapses as a pure mathematics exercise https://www.youtube.com/watch?v=CRLXDnViSyI
but is then awkwardly limited in how it relates back to the actual application in a DC circuit, rather it gets caught up in itself - happy that it has solved the mathematics
Moreover what tends to happen is the former limiting explanation "A particular circuit voltage or current does not depend on the past value of any circuit voltage or current" takes precedence. , so really exciting applications involving DC ( like I do ) get totally ignored, and commented on as being impossible. We see the conventional explanation of DC above containing explanation "the circuit voltages and currents are independent of time" An example contrary to this being the thyristor used in a DC circuit where the end of the circuit instructs the beginning, or where a gate or base of a semiconductor placed at the end of a circuit instructs the beginning of a circuit to commence.
which is then unexplained or contrary to conventional thinking. But the annoying thing is that there are devices readily available that solve a lot of problems in electronics - that is when and if they are applied. Hence the study of DC needs expanding, to place time as a real factor in DC circuits.
Thanks for that, Chris. Most of it goes over my head, but I understand the basis of where you're coming from. Certainly, I've always admired how you're able to think outside the box and apply some lateral thinking to the subject of electronics:)
Marco.
Ali Tait
15-03-2019, 17:12
Chances are they'll be audible below 10kHz :eyebrows:
Yep. :-)
VanDerGraaf
15-03-2019, 21:13
Nothing to do with your posts here, Marco. :)
Just someone saying that finer sampling under the Nyquist frequency gives better accuracy.
It doesn't, the musical waveform can already be reconstructed accurately.
This has been addressed several times on AOS, but it keeps coming back again.
Jerry does this also mean that there is no improved resolution of timing accuracy/recreation of absolute rhythmic subtleties? If there are any links to anything on here I'd be keen to read them.
This article gives a pretty clear explanation: https://people.xiph.org/~xiphmont/demo/neil-young.html
Just listen for yourself..forget the theory from the Internet
Sound better to you?
If it does, it’s better.
(You won’t get in trouble with the hifi mafia, your good.) :)
.
(You won’t get in trouble with the hifi mafia, your good.)
Although one day - and that day may never come - they might ask you for a favour in return...
Although one day - and that day may never come - they might ask you for a favour in return...
True... :D
Mastering and the actual CD / file rules supreme.
On a high end system the differences between any compression, and lossless, bit perfect files, is clearly apparent.
Software volume is almost always the devil, I always notice when it’s on, like last night when my wife was playing some music. Software volume should always be at 100%
16/44.1 *is* already “high resolution”. Mastering makes it tough to compare files that are also available in 24/96, thus don’t worry about it and just listen to which one you prefer...
Jitter is more influential than people may realise. You just get more of everything when there is less of it. Async USB can only reduce incoming jitter levels, not eliminate it.
BNC is a better connection, RCA by design introduces signal reflections & ringing that can alter clock transition points. A real facepalm in design standards.
Sometimes there is some operating system work required to ensure that audio transmission is bit perfect, and in “exclusive” mode.
...that’s the stuff I have found makes a difference.
Mastering and the actual CD / file rules supreme.
On a high end system the differences between any compression, and lossless, bit perfect files, is clearly apparent.
Software volume is almost always the devil, I always notice when it’s on, like last night when my wife was playing some music. Software volume should always be at 100%
16/44.1 *is* already “high resolution”. Mastering makes it tough to compare files that are also available in 24/96, thus don’t worry about it and just listen to which one you prefer...
Jitter is more influential than people may realise. You just get more of everything when there is less of it. Async USB can only reduce incoming jitter levels, not eliminate it.
BNC is a better connection, RCA by design introduces signal reflections & ringing that can alter clock transition points. A real facepalm in design standards.
Sometimes there is some operating system work required to ensure that audio transmission is bit perfect, and in “exclusive” mode.
...that’s the stuff I have found makes a difference.
I would say the are all facts and realities.
Not so sure about jitter. Jitter sounds like tape wow and flutter. We've all owned or heard a cheap cassette deck (except maybe the youngsters) so we all know what that sounds like. The wobble on sustained notes. Who's ever heard a digital source sound like that? (unless it's actually on the recording like on the Boston album 'Third Stage' for example).
Even that cheapo DVD player you found in a skip has better jitter performance than any RTR deck or turntable by quite a large margin. There needs to be a lot of jitter before it is possible to hear it. http://www.ultrahighendreview.com/uploads/documents/Benjamin_and_Gannon_on_Jitter.pdf
if you look for issues with audio, you will find them.
walpurgis
16-03-2019, 18:42
if you look for issues with audio, you will find them.
Bit like women really. :D
Bit like women really. :D
Women hate wires.
Not so sure about jitter. Jitter sounds like tape wow and flutter. We've all owned or heard a cheap cassette deck (except maybe the youngsters) so we all know what that sounds like. The wobble on sustained notes. Who's ever heard a digital source sound like that? (unless it's actually on the recording like on the Boston album 'Third Stage' for example).
Even that cheapo DVD player you found in a skip has better jitter performance than any RTR deck or turntable by quite a large margin. There needs to be a lot of jitter before it is possible to hear it. http://www.ultrahighendreview.com/uploads/documents/Benjamin_and_Gannon_on_Jitter.pdf
? What jitter are you talking about?
AJSki2fly
17-03-2019, 07:03
Some might find this an interesting and informative read CD: Jitter, Errors and Magic (https://www.stereophile.com/reference/590jitter/index.html)
It gives and insight into some technical issues with CD and digital storage, most of us will have ripped CD’s to hard drives or some other form of storage and/or will be listening to CD’s so the issue discussed will be relavent and may even be with streaming.
? What jitter are you talking about?
Periodic jitter.
Light Dependant Resistor
17-03-2019, 09:31
Not so sure about jitter. Jitter sounds like tape wow and flutter. We've all owned or heard a cheap cassette deck (except maybe the youngsters) so we all know what that sounds like. The wobble on sustained notes. Who's ever heard a digital source sound like that? (unless it's actually on the recording like on the Boston album 'Third Stage' for example).
Even that cheapo DVD player you found in a skip has better jitter performance than any RTR deck or turntable by quite a large margin. There needs to be a lot of jitter before it is possible to hear it. http://www.ultrahighendreview.com/uploads/documents/Benjamin_and_Gannon_on_Jitter.pdf
I would accept the findings of this article with less respect than talking to your local politician.
"Jitter induced by the digital audio interface was not found to be an audible problem for any of the program material auditioned"
Where is explanation of any circuitry different to SPDIF ... not in the article.. so how can they possibly affirm anything different as being audible to SPDIF ?
I explored this very area in 1997 http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm and finding ridding your digital transfer of SPDIF
makes considerable difference - audible as better bass response and stereo separation.
With CD players as a segment of the digital replay devices, there is greater reward in finding the right op amp for the current to voltage analog output stage.
As example Sony CDP555ES benefits enormously from using 6x opa2107 . A marantz CD52SE mk2 the op amp to use is 2x LM4562
I would accept the findings of this article with less respect than talking to your local politician.
"Jitter induced by the digital audio interface was not found to be an audible problem for any of the program material auditioned"
Where is explanation of any circuitry different to SPDIF ... not in the article.. so how can they possibly affirm anything different as being audible to SPDIF ?
I explored this very area in 1997 http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm and finding ridding your digital transfer of SPDIF
makes considerable difference - audible as better bass response and stereo separation.
It's not an article its a study. With proper controls etc. There's no problem with Sp/dif that we can hear. That's really the funny thing, digital replay may not be perfect but it is way better than our hearing requires. and way more perfect than the alternatives of analogue tape or vinyl. Fact is we can tolerate large amounts of distortion or jitter before we start to notice them. If we couldn't no-one would ever play a vinyl LP again. And that's before you even get to the loudspeakers, A small loudspeaker can have 10% distortion in bass frequencies, no-one seems to mind. But 0.001% distortion from a CD and everyone is running around looking for a solution.
I think a lot of people underestimate just how much manufacturers will happily lie to us in order to get our coin. Even the big boys like Sony.
Good explanation of the different types of jitter here: https://ransomsnotes.com/index_htm_files/RansomStephensJitter360Pt3AllAboutTheAcronyms.pdf
Jitter was a solved problem before CD even came out. I've got vintage players here that will demonstrate that on a practical level. (Marantz CD54 from 1984, Technics SLP1200 from 1989).
The SQ issues people experience with CD have nothing to do with jitter.
Ever tried adding a better clock? ;)
The SQ improvements I’ve heard in my system, with a DAC I’ve used for 7yrs, just from a lower jitter signal, is blatantly clear.
It uses a simple SPDIF receiver with a PLL that isn’t doing anything much, so is very transport sensitive...
Jitter was a solved problem before CD even came out. I've got vintage players here that will demonstrate that on a practical level. (Marantz CD54 from 1984, Technics SLP1200 from 1989).
The SQ issues people experience with CD have nothing to do with jitter.
What are the SQ problems associated with CD if we exclude jitter?
What exactly is a 'better clock' though? And how are you certain that the improvement you hear after the modifications is due to reduced jitter and not some other factor? Now I'm not saying it couldn't be but for there to be an audible improvement the levels of jitter before the mod would need to be extraordinarily high. Possible but very unlikely. What is the DAC?
Periodic jitter.
Ok but how does periodic jitter manifests itself as sounding like tape wow & flutter
What are the SQ problems associated with CD if we exclude jitter?
Excessive use of compression in mastering and using amplifiers that are sub-optimal.
Ok but how does periodic jitter manifests itself as sounding like tape wow & flutter
There's a site where you can compare downloads with no jitter and with shedloads of jitter added. I've just been looking for it but I can't find it right now. It sounds exactly like tape wow. If I can find it I'll post it up.
Ever tried adding a better clock? ;)
The SQ improvements I’ve heard in my system, with a DAC I’ve used for 7yrs, just from a lower jitter signal, is blatantly clear.
Indeed. That was a key part of the improvements to my vintage Sony DAS-R1, which I had extensively modified by Audiocom. This thread might be of interest to anyone not au fait with the (undoubted) sonic benefits possible with clock upgrades to DACs: https://theartofsound.net/forum/private.php?do=showpm&pmid=427572
See post #5, from Mark, in particular. He truly is an expert on all things digital.
What I found was that it removed much of the 'glare', which in my experience affects ALL forms of digital replay, although to varying degrees depending on a number of factors, and which can make for a fatiguing sound, over long periods, and crucially that you often don't realise exists until it's gone!
Done properly, it's a fundamental way of improving any DAC, the effect of which is maximised when similar attention is paid to power supply arrangements.
Marco.
I am glad I don't take much of what is said on this subject too seriously here as there is a lot of bullshit,misunderstanding and plain wrong *facts* being suggested on this post as there are on many of the posts concerning digital audio.
I would recommend that if folk want to really know any of the facts surrounding digital audio they head over to the HB project where they can learn the realities of digital audio.
Which "post" are you referring to, Jim?
Marco.
Or come round to mine and have a listen to 1980s cd players with no glare, no harshness blah blah and re-think your whole perspective.
Using a Digione which has pretty good figures to go with its pretty good sound:eyebrows:
DigiOne has 2 outputs: BNC and plain RCA (coaxial). On both, we have achieved jitter of 0.6ps (1 picosecond – time taken by light in a vacuum to travel approximately 0.30 mm).
Noise. 50uV (batteries have about 12uV of noise).
What we achieved, in our opinion and based on public measurements, is the best transport at any price point (yes, even when compared with 5k USD/EUR transports).
So they say.
Have to say im pleased with it and for a ton, value for money is excellent....unbeatable i'd think.
I would recommend that if folk want to really know any of the facts surrounding digital audio they head over to the HB project where they can learn the realities of digital audio.
http://thehbproject.com/en
Respectfully disagree, I would not recommend that site on the whole. Some of what he says is correct but even a broken clock is right twice a day. Most of the issues he talks about as being a big deal make no difference to what we can hear.
As usual with audio mistakenly linking a cause to an effect can lead to some wrong (and possibly expensive) conclusions.
http://thehbproject.com/en
Respectfully disagree, I would not recommend that site on the whole. Some of what he says is correct but even a broken clock is right twice a day. Most of the issues he talks about as being a big deal make no difference to what we can hear.
As usual with audio mistakenly linking a cause to an effect can lead to some wrong (and possibly expensive) conclusions.
I would certainly take his advice than yours Martin. With respect I don't think you are technically knowledgeable enough concerning digital audio to be a useful guide as to what is right or wrong in digital audio.
You may have opinions but I feel that is all they are.
Or come round to mine and have a listen to 1980s cd players with no glare, no harshness blah blah and re-think your whole perspective.
Lol... There will be glare, as with ALL digital replay equipment; it's just that often you don't realise it's there (like me) until it's gone! However, I don't expect you to accept that until/if you hear it for yourself:)
Marco.
I was refering to Martins Post concerning Jitter.
"Jitter was a solved problem before CD even came out. I've got vintage players here that will demonstrate that on a practical level. (Marantz CD54 from 1984, Technics SLP1200 from 1989).
The SQ issues people experience with CD have nothing to do with jitter."
As you said in a previous post you don't often know you have "glare" in digital audio until it is gone and there are many ways of reducing jitter or poor clock timing. Every aspect of the digital chain from start to finished needs to be optimised in order to get a really good result including PSUs, cables,software, hardware, system set up etc etc....
Yes I fully agree - and experience bears it out:)
Btw, in future it's always best to quote *exactly* what you're referring to, as otherwise it can be unclear. In this instance, your post in question appeared directly under mine, so I thought you were refuting what I had written.
Marco.
Yes I fully agree - and experience bears it out:)
Btw, in future it's always best to quote *exactly* what you're referring to, as otherwise it can be unclear. In this instance, your post in question appeared directly under mine, so I thought you were refuting what I had written.
Marco.
Sorry, will do.:)
Someone who knows digital discussing Jitter.
https://youtu.be/7FyKVZ9D3E0
https://youtu.be/ZT_1UATci3c
https://youtu.be/yoKqnBJN8iQ
Seen those videos before, still love the ‘foot tapping.....touchy-feely crap’ comment. His description of how jitter may be perceived mirrors my own experience of improving the master clock in DACs and players. Most recently with the Kali reclocker on the RPi where it’s not just the improvement in detail retrevial but also in the foot tapping enjoyment stakes.
Yup.
all about....
timing.
:)
I would certainly take his advice than yours Martin. With respect I don't think you are technically knowledgeable enough concerning digital audio to be a useful guide as to what is right or wrong in digital audio.
You may have opinions but I feel that is all they are.
The technical part is done by other people. I simply relate their findings. And then there is also my own subjective experience, which tends to correlate with what the research says, at least broadly speaking. Subjectively I found that as I improved my amplification the problems I had with digital, and which many believe to be inherent to digital, went away. Even with budget players and players from the mid/late 1980s that were supposedly full of jitter, could strip paint, etc etc.
I do agree that it makes sense to optimise every facet of the digital playback system, it's unlikely to hurt and may have a cumulative effect. But IME it makes a lot more sense to optimise the amplification first.
Jitter DOES matter though Martin, technically which can be heard subjectively.
BUT...
Not everyone is bothered by it of course, or can hear it as the first video above says.
If it’s not hurtin’ in a particular system or set of ears, I’d forget it and just enjoy the music.
No point looking for problems, and as you say maybe in a particular system more benefit could be found upgrading elsewhere.
Jitter DOES matter though Martin, technically which can be heard subjectively.
.
.
It matters if there is enough of it to be audible I agree. The question should be does any DAC or CD player have enough of it to be audible? If you look on ASR site they have tested about 60 DACs and even the worst performing ones do not have jitter any where near audible levels.
Take a look at the research I linked to earlier where they tested the audibility of jitter. It's funny that even though I link to this stuff people think I am making it up myself. There's a whole world of research on this stuff out there, it's not some big mystery. These questions were asked and properly answered some time ago. Oh well.
Take a look at the research I linked to earlier where they tested the audibility of jitter. It's funny that even though I link to this stuff people think I am making it up myself. There's a whole world of research on this stuff out there, it's not some big mystery. These questions were asked and properly answered some time ago. Oh well.
Yes, but if you've carried out your own experiments and research into the matter, and what you've heard/discovered in that respect flies in the face of the technical research you've linked to, why should we change our (well deserved) contrary opinion?
Correct me if I'm wrong, but I don't believe you've carried out any clock upgrades to DACs you've owned, which addresses the issue of jitter, and thus have heard the 'before and after' effect? If you had done, your views on the subject might be different.
As I've said many times before, *ultimately* (that means once MY judgement criteria has been satisfied), I'll always trust my own ears, over ANY amount of technical research conducted by others, and this area of audio is but another valid example of such. I never form conclusive opinions on any subject, audio or otherwise, simply as a result of having read something somewhere.
Just because a so-called expert has supposedly created a 'fact', in reference to digital audio, doesn't automatically make it so... Therefore, always be prepared to apply some lateral thinking and have faith in your OWN judgement, regardless of what any 'experts' say.
Personal, practical experience is always what counts most!:cool:
Marco.
Jitter and Timing destroy many aspects of digital audio including bass, soundstage and can smear the audio image. When they are addressed as best as possible you can hear the improvements but you need to start with bit perfect ripped files first. CD is not going to be even in the starting blocks and will never give the best representation that can be delivered by digital audio.
Until you investigate this route Martin, you will just assume that what you have is perfectly OK which is fair enough. But considering you are a committed digital based audiophile you owe it to yourself to at least try it in your own system.
You heard Olivers Pi3 BOSS set up and I noticed you did not mention this in your post on the new speaker cables. I would be surprised if you were not impressed by what you heard from this £200 set up!
Go on give it a go and see what you are missing!
Jitter and Timing destroy many aspects of digital audio including bass, soundstage and can smear the audio image. When they are addressed as best as possible you can hear the improvements but you need to start with bit perfect ripped files first. CD is not going to be even in the starting blocks and will never give the best representation that can be delivered by digital audio.
Until you investigate this route Martin, you will just assume that what you have is perfectly OK which is fair enough. But considering you are a committed digital based audiophile you owe it to yourself to at least try it in your own system.
You heard Olivers Pi3 BOSS set up and I noticed you did not mention this in your post on the new speaker cables. I would be surprised if you were not impressed by what you heard from this £200 set up!
Go on give it a go and see what you are missing!
What you are describing about the audible effect of jitter is incorrect. And as the AES paper I linked to demonstrates, jitter is inaudible in all but worst case scenarios. All cd players deal satisfactorily with jitter. You are more actually likely to increase jitter by using a separate transport/streamer and DAC, not less likely. Although it still won't matter. And bit perfect is irrelevant, it makes no difference if you read the file from a cd or from a hard drive So we are at pretty much 180 degree opposing views :)
Not that it matters, I think it is good that all sides of the picture get discussed.
Ollies Pi set up sounded good. £200 is indeed a bargain. I'd have to listen to it in my system though before I could safely say it is an improvement on the Sony cd player I'm using right now which was an even better bargain at £50.
Ollie's pie set up sounds good because he has properly sorted amplification. That's why when we stuck a budget 1995 Sony CD player (£40 from ebay) in there last Autumn it also sounded good.
What you are describing about the audible effect of jitter is incorrect. And as the AES paper I linked to demonstrates, jitter is inaudible in all but worst case scenarios. All cd players deal satisfactorily with jitter. You are more actually likely to increase jitter by using a separate transport/streamer and DAC, not less likely. Although it still won't matter. And bit perfect is irrelevant, it makes no difference if you read the file from a cd or from a hard drive So we are at pretty much 180 degree opposing views :)
Not that it matters, I think it is good that all sides of the picture get discussed.
Ollies Pi set up sounded good. £200 is indeed a bargain. I'd have to listen to it in my system though before I could safely say it is an improvement on the Sony cd player I'm using right now which was an even better bargain at £50.
Ollie's pie set up sounds good because he has properly sorted amplification. That's why when we stuck a budget 1995 Sony CD player (£40 from ebay) in there last Autumn it also sounded good.
I think we will have to just disagree about quite a few of your comments on digital. I think they are all wrong but good for discussion as you say.
I would be interested in what you thought of the Pi in your system as it sorts many of the aspects that undermine digital and as a result it presents a superb digital sound. It is because of the lack of jitter, bit perfect files, well sorted PSU and excellent clock timing that it sounds so good, however this would be at odds with your theories.
In the end it is how it sounds that matters and for me it is approaching analogue very closely especially on some material.
That is a very high recommendation in my book as I am obviously a vinylista!
Light Dependant Resistor
17-03-2019, 21:00
Jitter was a solved problem before CD even came out. I've got vintage players here that will demonstrate that on a practical level. (Marantz CD54 from 1984, Technics SLP1200 from 1989).
The SQ issues people experience with CD have nothing to do with jitter.
Martin politely suggest you have a read of this study published in October of 1993 https://www.stereophile.com/reference/1093jitter/index.html
Although some DAC integrated circuits have improved such as those from Crystal used in a lot of CD players NAD Marantz to name two, the
fundamentals of addressing jitter outlined by Reme Fourre and also study done in the UK by Malcolm Hawksford https://www.youtube.com/watch?v=b5nzFejc2KA
remains elusive to most commercially available equipment.
In Malcolms talk the frequency response of digital is raised as i have pointed out already a number of times at 6 mins into the you tube video, is discussed as being inadequate or compromised
and Malcolm suggests 96 khz , inferring a audio frequency response to at least 45 khz allowing for some filtering before the Shannon/Nyquist point is reached.
I put it to you Sir that you are a Propellerhead. :)
https://youtu.be/7FyKVZ9D3E0
What you are describing about the audible effect of jitter is incorrect. And as the AES paper I linked to demonstrates, jitter is inaudible in all but worst case scenarios. All cd players deal satisfactorily with jitter. You are more actually likely to increase jitter by using a separate transport/streamer and DAC, not less likely. Although it still won't matter. And bit perfect is irrelevant, it makes no difference if you read the file from a cd or from a hard drive So we are at pretty much 180 degree opposing views :)
Not that it matters, I think it is good that all sides of the picture get discussed.
Ollies Pi set up sounded good. £200 is indeed a bargain. I'd have to listen to it in my system though before I could safely say it is an improvement on the Sony cd player I'm using right now which was an even better bargain at £50.
Ollie's pie set up sounds good because he has properly sorted amplification. That's why when we stuck a budget 1995 Sony CD player (£40 from ebay) in there last Autumn it also sounded good.
Martin politely suggest you have a read of this study published in October of 1993 https://www.stereophile.com/reference/1093jitter/index.html
Although some DAC integrated circuits have improved such as those from Crystal used in a lot of CD players NAD Marantz to name two, the
fundamentals of addressing jitter outlined by Reme Fourre
.
It's an explanation of jitter and how to test for it. I'm not saying jitter does not exist.
Note in the article there is no mention of what jitter sounds like and what levels it has to be at before you can hear it. It's just taken as read from the opening of the article - 'Until recently, all problems in digital audio systems...' - that there are problems with the sound of digital audio and we need to look for them and solve them. No mention of how they manifest themselves, something which you would have thought might be slightly pertinent. Why do you suppose that was omitted?
blackmetalboon
17-03-2019, 21:57
It's an explanation of jitter and how to test for it. I'm not saying jitter does not exist.
Note in the article there is no mention of what jitter sounds like and what levels it has to be at before you can hear it. It's just taken as read from the opening of the article - 'Until recently, all problems in digital audio systems...' - that there are problems with the sound of digital audio and we need to look for them and solve them. No mention of how they manifest themselves, something which you would have thought might be slightly pertinent. Why do you suppose that was omitted?
Very good points/questions, Martin. I’m with you on this subject, a non-problem looking for a solution. I know I may be unpopular for this view, but how many people who have this problems with the digital reproduction of music come from the analogue generation? When people have commented on my (digital) system as being very analogue sounding, I’ve not seen it as a compliment.
Light Dependant Resistor
18-03-2019, 05:04
It's an explanation of jitter and how to test for it. I'm not saying jitter does not exist.
Note in the article there is no mention of what jitter sounds like and what levels it has to be at before you can hear it. It's just taken as read from the opening of the article - 'Until recently, all problems in digital audio systems...' - that there are problems with the sound of digital audio and we need to look for them and solve them. No mention of how they manifest themselves, something which you would have thought might be slightly pertinent. Why do you suppose that was omitted?
On page 2 of the Reme Fourre article, a figure of 200 pico seconds jitter error is used to describe spurious tones that then appear at higher harmonics. Complicating is as the dynamic range
increases with higher bit D/A converters the problems of jitter just get worse. Malcolm Hawksford begins discussion of the problems of jitter distortion at 14.30 into the You tube video
https://www.youtube.com/watch?v=b5nzFejc2KA
Jitter being a distortion displaces timing to no longer sound like the original recording.
Jitter is manifested in inability firstly of the Analog to Digital Conversion, to preserve time correctly, and secondly in Digital to Analog Conversion, but
mostly where a transport and outboard D/A are implemented, namely the SPDIF or AES/EBU interface as outlined by Reme Fourre on page 2
https://www.stereophile.com/content/jitter-digital-interface-page-2
"Jitter in the digital interface
A logical question to ask is: Where does this clock jitter originate? The main source of jitter is the digital interface between a digital source such as a CD transport and the D/A converter. This interface is either the AES/EBU (Audio Engineering Society/European Broadcast Union) or S/PDIF (Sony/Philips Digital Interface Format) format. With the exception of voltage levels and line impedances, the two formats are very similar. Both carry two channels of digital audio data (up to 24 bits per sample per channel, though only 16 are currently used in consumer applications), control bits, parity bits, and checksum bits. The bit rate is 64 times the sampling frequency (64Fs), and the baud rate (the rate at which the line may change state) is 128Fs. For the CD's 44.1kHz sampling rate, the S/PDIF or AES/EBU interface has a bit rate of 2,822,400 bits per second and a baud rate of 5.6448MHz.
Two optical-fiber interfaces are also in use: TosLink and ST-type, both of which carry information optically instead of electrically. The TosLink interface, more properly called "EIAJ optical," uses a plastic optical fiber. AT&T ST-type optical uses a glass optical fiber. These two optical formats are merely transmission methods; they both use the AES/EBU or S/PDIF data formats.
The digital interface has two major functions: It carries the digital data and it carries the sampling clock. Both data and clock are transmitted on the same physical electrical conductors or optical fiber link: data are encoded using a pulse-modulation scheme, and the clock is embedded in the pulse edges. This system requires that the clock and data be separated in the digital processor, a function performed by the digital processor's input receiver. The "recovered" clock then serves as the processor's master clock. Consequently, jitter in the interface data stream produces clock jitter at the DAC. This is the mechanism by which transports affect a digital system's sonic performance. Moreover, the quality of the digital interface implementation greatly affects the amount of jitter in the recovered clock.
Interface jitter is similar to the periodic clock jitter described earlier. Because current input receiver implementations detect signal edges by comparing the input voltage to zero, jitter in the digital interface signal is defined as the sequence of the timing errors on the voltage zero crossings.
As with noise, jitter has multiple causes. One of them is electrical noise picked up on the interface cable. Noise causes the zero crossing points to shift slightly—which is, by definition, interface jitter. If the digital interface signal has an average slope of 20V/µs (a typical value in many S/PDIF implementations), just 10mV RMS of noise, for example, will introduce 500ps RMS of jitter. '
I looked at this very issue to disable SPDIF and replace it with 3 discrete codes between a CD transport and a Audio Synethesis DSM in 1997 , described here: http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
The benefits were improved bass response and stereo separation compared to SPDIF transfer that sounded muddled. I also did comparison with the best available turntable listening to each through Quad ESL63's
The source was a Three Blind Mice LP and identical Three Blind Mice CD - the benefit of the digital three code system was that it lacked any noise that the LP provided , but otherwise sounded
the same.
Very good points/questions, Martin. I’m with you on this subject, a non-problem looking for a solution. I know I may be unpopular for this view, but how many people who have this problems with the digital reproduction of music come from the analogue generation? When people have commented on my (digital) system as being very analogue sounding, I’ve not seen it as a compliment.
It is in fact the orthodox engineering view on the subject. The alternative point of view expressed by sites like HB project - that we can hear frequencies higher than 22Khz, that tiny amounts of jitter are audible, that pre and post ringing are audible, is not taken seriously.
A lot of research has been done over the past 100 years on what we are capable of hearing. Not from the perspective of hi-fi but as a general subject. When digital audio (CD) was being designed in the 1970s this already existing and accepted research was used to set the upper limit of frequency response at 22Khz. Since then nothing has changed as far as medical science is concerned. Someone mentioned bone conductance earlier in the thread. yes this does work - if the loudspeaker is up against your skull and the high frequencies are transmitted at sufficient levels (i.e high levels). None of this happens in reality.
Light Dependant Resistor
18-03-2019, 07:50
It is in fact the orthodox engineering view on the subject. The alternative point of view expressed by sites like HB project - that we can hear frequencies higher than 22Khz, that tiny amounts of jitter are audible, that pre and post ringing are audible, is not taken seriously.
A lot of research has been done over the past 100 years on what we are capable of hearing. Not from the perspective of hi-fi but as a general subject. When digital audio (CD) was being designed in the 1970s this already existing and accepted research was used to set the upper limit of frequency response at 22Khz. Since then nothing has changed as far as medical science is concerned. Someone mentioned bone conductance earlier in the thread. yes this does work - if the loudspeaker is up against your skull and the high frequencies are transmitted at sufficient levels (i.e high levels). None of this happens in reality.
What you are missing is that audio in the spectrum we hear is changed detrimentally, where frequency response is filtered. https://earthworksaudio.com/wp-content/uploads/2012/07/The-world-beyond-20kHz.pdf
No there was no proper research done rather it was found a upper limit of 20- 22.05 Khz suited the redbook standard to provide a music CD disc of 74 mins duration, as always market product comes first rather
than quality.
What you are missing is that audio in the spectrum we hear is changed detrimentally, where frequency response is filtered. https://earthworksaudio.com/wp-content/uploads/2012/07/The-world-beyond-20kHz.pdf
There's nothing in that article offering anything other than speculation to back that claim up. And the bloke does have a business selling extended bandwidth mics to the recording industry. of course recording at high sampling and bit rates makes perfect sense, it just makes no sense for playback.
If we are going to say that we need frequencies above 22Khz to be able to hear music properly, what about all the recordings, pop, rock, jazz, classical made with equipment that did not have that bandwith - i.e all the great recordings of history? is 'Kind Of Blue' now not worth listening too because the SQ is awful? Deep Purple 'In Rock'? - now worthless, sounds rubbish.
Sorry, but this makes absolutely no sense, either theoretically or practically. Even if we make a leap of faith and believe there would be a difference had these recordings been made with a bandwidth of 40Khz the difference, at best, would be infinitesimal as far as our perception is concerned.
Light Dependant Resistor
18-03-2019, 08:38
[QUOTE=Light Dependant Resistor;1070595]What you are missing is that audio in the spectrum we hear is changed detrimentally, where frequency response is filtered. https://earthworksaudio.com/wp-content/uploads/2012/07/The-world-beyond-20kHz.pdf
QUOTE]
There's nothing in that article offering anything other than speculation to back that claim up. And the bloke does have a business selling extended bandwidth mics to the recording industry. of course recording at high sampling and bit rates makes perfect sense, it just makes no sense for playback.
If we are going to say that we need frequencies above 22Khz to be able to hear music properly, what about all the recordings, pop, rock, jazz, classical made with equipment that did not have that bandwith - i.e all the great recordings of history? is 'Kind Of Blue' now not worth listening too because the SQ is awful? Deep Purple 'In Rock'? - now worthless, sounds rubbish.
Sorry, but this makes absolutely no sense, either theoretically or practically. Even if we make a leap of faith and believe there would be a difference had these recordings been made with a bandwidth of 40Khz the difference, at best, would be infinitesimal as far as our perception is concerned.
David Blackmers findings are backed by Malcolm Hawksford 6 mins into the video, namely calling for 96Khz sampling and as a result 45khz frequency response.
Likewise this paper identifies muted trumpet harmonics extending well beyond 20Khz http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
and shown in this graph 25585
Thankfully recording engineers usually preserves frequencies above 20khz, it is just sad our replay equipment does not presently properly reproduce what was recorded.
http://thehbproject.com/en
Respectfully disagree, I would not recommend that site on the whole. Some of what he says is correct but even a broken clock is right twice a day. Most of the issues he talks about as being a big deal make no difference to what we can hear.
As usual with audio mistakenly linking a cause to an effect can lead to some wrong (and possibly expensive) conclusions.
I like to see / read practical evidence supporting or disproving these issues and I agree with Martin on this one. The HB project guy needs to re-evaluate the way digital filters work, yes he makes some good points but the practical evidence shown here: http://archimago.blogspot.com/2018/01/audiophile-myth-260-detestable-digital.html#more shows that digital filters work perfectly well in a correctly bandwidth limited signal. An impulse test to prove they don’t is not valid, all that does is to test the response of the filter not how it will actually sound with normal program material because the impulse itself is not bandwidth limited.
It matters if there is enough of it to be audible I agree. The question should be does any DAC or CD player have enough of it to be audible? If you look on ASR site they have tested about 60 DACs and even the worst performing ones do not have jitter any where near audible levels.
Take a look at the research I linked to earlier where they tested the audibility of jitter. It's funny that even though I link to this stuff people think I am making it up myself. There's a whole world of research on this stuff out there, it's not some big mystery. These questions were asked and properly answered some time ago. Oh well.
Worked my way through the AES paper, the conclusion I took away was yes jitter can be heard but the author was surprised at the levels required, however, he also noted that it was material dependant and that with training the test participants improved their ability to identify jitter. If the listening participants where non-audio types then this maybe skewed the results, I’m totally guessing that if more keen audio participants where used the results may have been different. What I didn’t take from the paper was that CD players from decades ago had ‘solved the jitter issue’.
I like to see / read practical evidence supporting or disproving these issues and I agree with Martin on this one. The HB project guy needs to re-evaluate the way digital filters work, yes he makes some good points but the practical evidence shown here: http://archimago.blogspot.com/2018/01/audiophile-myth-260-detestable-digital.html#more shows that digital filters work perfectly well in a correctly bandwidth limited signal. An impulse test to prove they don’t is not valid, all that does is to test the response of the filter not how it will actually sound with normal program material because the impulse itself is not bandwidth limited.
Which bit do you agree with Martin on?
Worked my way through the AES paper, the conclusion I took away was yes jitter can be heard but the author was surprised at the levels required, however, he also noted that it was material dependant and that with training the test participants improved their ability to identify jitter. If the listening participants where non-audio types then this maybe skewed the results, I’m totally guessing that if more keen audio participants where used the results may have been different. What I didn’t take from the paper was that CD players from decades ago had ‘solved the jitter issue’.
No doubt, and that's precisely why, unlike Martin, I never take the conclusions of these technical studies/papers as Gospel, or at least allow them to influence my opinions to the extent that I use them to confirm my arguments in debates such as this.
What matters ultimately, is the conclusions YOU have reached, based on YOUR own research and experience, in reference to a specific subject, not what some technical paper says, which although interesting and perhaps in some ways relevant, does not automatically prove anything, especially, as you say, when it's likely the conclusions reached are based upon information gathered from participants not ideally qualified to perform their intended roles.
And in my experience, as you say, CD players from decades ago had not "solved the jitter issue", otherwise I wouldn't have had the clock in mine upgraded by Audiocom, subsequently with great success, and resulting in easily heard (significant) sonic improvements, along the lines you described earlier, after you'd carried out something similar with your gear.
Marco.
It is in fact the orthodox engineering view on the subject. The alternative point of view expressed by sites like HB project - that we can hear frequencies higher than 22Khz, that tiny amounts of jitter are audible, that pre and post ringing are audible, is not taken seriously.
Why, simply because of arrogance?
Well, it should be taken seriously, as that view, based on the conclusions of someone else's research and experience, is as valid as the "orthodox engineering view" you choose to accept.
Marco.
Note in the article there is no mention of what jitter sounds like and what levels it has to be at before you can hear it.
That is explained in the video Gaz linked to, entitled 'Why jitter matters'. Jitter manifests itself by creating timing anomalies in music, which to some discerning listeners is audible. The chap also explains the levels involved, where jitter becomes audible, in terms of chips he's fitted and tested. Watch the video again and you'll see.
You have a bad habit of highlighting only information that appears to back up your argument, whilst ignoring the rest!;)
Marco.
Which bit do you agree with Martin on?
Recommending that site as a whole, as per my post above.
here’s a bit more on impulse and filter ringing. http://archimago.blogspot.com/2016/07/musings-digital-interpolation-filters.html
When you move to bit perfect accurately clocked file based audio you can hear what the effect is of removing jitter. Technical journalists and articles written about the subject can tell you this but the proof is actually demonstrating it to yourself and actually hearing the results.
You can read lots of opposed claptrap and theoretical technical arguments but it is only when you undertake actually evaluation yourself through listening does any of this make sense.
A lot of the work in digital audio at the moment seems to be around Power supplies and for good reason. Their effect on DACs and anything in the audio chain is audible.
Suggesting jitter had been solved is a nonsense because even with the very best equipment and accurate clocking their are variables in quality of parts used.
Timing is everything and the reconstruction of the original digital recording can so easily be compromised, especially by CD players unless huge effort and expense has been taken to resolve the issues. A spinning disc read by a laser at 500 mph with small scratches and imperfections is not best way to extract digital audio!
Hi Jason,
I know I may be unpopular for this view, but how many people who have this problems with the digital reproduction of music come from the analogue generation? When people have commented on my (digital) system as being very analogue sounding, I’ve not seen it as a compliment.
Well, in terms of your first point, I'm from what you call the "analogue generation" (so is Martin), but also fully embrace, and indeed enjoy, digital audio equipment and digitally reproduced music, having both a superb Sony CDP/DAC, from the era Martin mentions, where the players and DACs he claims had already "solved the jitter issue", yet in my case benefitted significantly from having clock upgrades carried out, plus a modern streaming system, with access to over 6TB of music files - both of which I listen to regularly and enjoy immensely.
The sound either produces isn't 'perfect', but then nothing is, but I don't sit there worrying about the effects of jitter, produced by my digital equipment, simply because it sounds great, regardless of whatever small amounts of jitter are present - and present I believe they are, to varying degrees, not just in my gear, but ALL digital equipment, vintage and modern. In effect though, it doesn't spoil my enjoyment of the music it makes.
However, if someone were to build a box that successfully removed all jitter from my digital equipment, I'd be very confident of hearing the difference, just as I did when clock upgrades were applied to my 1987 Sony DAS-R1 DAC. Therefore, it's not a case of a non-problem looking for a solution, but rather an existing problem (jitter) only becoming an issue if you're aware of it and can hear it, and so only at that point does it need a solution!:)
In terms of people saying that your digital system sounds "very analogue", well, that's simply a subjective observation that you can take how you like. It could be seen as a compliment though, if your digital system displays little of the traits (a hard, bright sound), which often afflicts digital equipment and causes listener fatigue, which as long as it doesn't make your system sound dull and boring (which I'm sure it doesn't), I'd take as a compliment:cool:
Marco.
You can read lots of opposed claptrap and theoretical technical arguments but it is only when you undertake actually evaluation yourself through listening does any of this make sense.
I fully agree. It's also what Martin needs to do before I can take his views on the matter more seriously, as quoting the contents of technical studies, to supposedly prove a point, conducted by goodness knows who [in terms of the 'guinea pigs' employed to produce the crucial test information], IMO, doesn't add any real value to the discussion.
If he came back and said that he's had the clock upgraded in his DAC, and couldn't hear any difference, or the opposite, then it'd sit up and take notice, as I trust his ears. The fact is, we can all search the Internet and find any number of 'technical studies', which on the surface appear to confirm our views on audio, but that doesn't necessarily mean that our efforts have been successful.
Timing is everything and the reconstruction of the original digital recording can so easily be compromised, especially by CD players unless huge effort and expense has been taken to resolve the issues. A spinning disc read by a laser at 500 mph with small scratches and imperfections is not best way to extract digital audio!
Nope, which is why all else being equal, especially in terms of the partnering DAC, streaming music files digitally, minus the use of discs and the physical interface of a CDP, can pay dividends.
Marco.
As you may have gathered Marco I am not a digital guy but I can appreciate what I hear whether its analogue or digital and some of the digital audio I have heard recently has been mighty impressive. I have then asked the reasons why and tried to understand technically what affects digital audio but it is sometimes difficult getting a clear straight answer. And with digital audio there are many ways to cut the cake!
Sorting Fact and Realities from the mumbo jumbo is difficult so I have simply followed trust worthy advice and experimented myself. I am not a flat earther and always open to suggestions in audio and if it sounds good then it is something I want to investigate. It may not always work out in your own system but unless you try I don't think you can argue whether its good or bad?
Like you have always advocated your own ears tell you everything. Technical information only gives you part of the story.
AJSki2fly
18-03-2019, 12:55
Hi Jason,
The sound either produces isn't 'perfect', but then nothing is, but I don't sit there worrying about the effects of jitter, produced by my digital equipment, simply because it sounds great, regardless of whatever small amounts of jitter are present - and present I believe they are, to varying degrees, not just in my gear, but ALL digital equipment, vintage and modern. In effect though, it doesn't spoil my enjoyment of the music it makes.
Marco.
Marco,
How do you know it isn't 'perfect', what are your terms of reference, your system may in fact be nearing audio perfection but we and you may not know it. :eyebrows: As you say it sounds great, so isn't that 'perfect' enough? :D
Cheers
Adrian
As you may have gathered Marco I am not a digital guy but I can appreciate what I hear whether its analogue or digital and some of the digital audio I have heard recently has been mighty impressive. I have then asked the reasons why and tried to understand technically what affects digital audio but it is sometimes difficult getting a clear straight answer. And with digital audio there are many ways to cut the cake!
Sorting Fact and Realities from the mumbo jumbo is difficult so I have simply followed trust worthy advice and experimented myself. I am not a flat earther and always open to suggestions in audio and if it sounds good then it is something I want to investigate. It may not always work out in your own system but unless you try I don't think you can argue whether its good or bad?
Like you have always advocated your own ears tell you everything. Technical information only gives you part of the story.
We're on the same page, dude:)
Basically, I'll ultimately trust my own ears and those of others that from experience I consider as good, such as yours or Martin's - not those of a complete stranger, whose hearing acuity/mental capacity ultimately forms the basis of the conclusions reached in some technical study, and who may be deaf to real things I can hear and consider as important.
That's why I'd never use such material as supporting evidence to form or back up any of my views on audio.
Marco.
Marco,
How do you know it isn't 'perfect', what are your terms of reference, your system may in fact be nearing audio perfection but we and you may not know it. :eyebrows: As you say it sounds great, so isn't that 'perfect' enough? :D
Ha... Adrian, I work from the assumption that *nothing* in life is perfect, otherwise we'd never have anything better to look forward to!;)
More seriously though, I know that I could improve my system (in certain areas), and afterwards almost certainly achieve a real improvement, so using that logic, it can't be perfect. Besides, this hobby of ours, by its very nature, is imperfect in many ways, so to kid oneself on otherwise would be rather foolish.
Marco.
We're on the same page here, dude:)
Marco.
:):thumbsup:
AJSki2fly
18-03-2019, 15:17
Ha... Adrian, I work from the assumption that *nothing* in life is perfect, otherwise we'd never have anything better to look forward to!;)
More seriously though, I know that I could improve my system (in certain areas), and afterwards almost certainly achieve a real improvement, so using that logic, it can't be perfect. Besides, this hobby of ours, by its very nature, is imperfect in many ways, so to kid oneself on otherwise would be rather foolish.
Marco.
So what you are telling me is my system is imperfect in essence, without having heard it :eyebrows: :D
Yesh, shweety, the same as I can state with relative certainty that your cock isn't 12" long, despite not having seen it!:eek::D:lol:
Marco.
So what you are telling me is my system is imperfect in essence, without having heard it :eyebrows: :D
Since no system is perfect (and never will be), it follows that all systems are to a greater or lesser extent imperfect - yours, mine and Marco's.
AJSki2fly
18-03-2019, 16:25
Quote Originally Posted by AJSki2fly View Post
So what you are telling me is my system is imperfect in essence, without having heard it
Since no system is perfect (and never will be), it follows that all systems are to a greater or lesser extent imperfect - yours, mine and Marco's.
No Barry, I think you have it all wrong, my system is in fact basically perfect because I listen to it and it is music to my ears, so as your system and Marco's are not the same then by definition they must be imperfect. :lol:
AJSki2fly
18-03-2019, 16:27
Yesh, shweety, the same as I can state with relative certainty that your cock isn't 12" long, despite not having seen it!:eek::D:lol:
Marco.
That depends on the time of day and temperature. :lol:
No Barry, I think you have it all wrong, my system is in fact basically perfect because I listen to it and it is music to my ears, so as your system and Marco's are not the same then by definition they must be imperfect. :lol:
By that logic, and using your criteria, Marco and I have perfect systems as well. :lol:
AJSki2fly
18-03-2019, 16:37
Quote Originally Posted by AJSki2fly View Post
No Barry, I think you have it all wrong, my system is in fact basically perfect because I listen to it and it is music to my ears, so as your system and Marco's are not the same then by definition they must be imperfect.
Barry Quote
By that logic, and using your criteria, Marco and I have perfect systems as well. :lol:
Are but only if you have perfect ears like mine, and therefore as no one is the same yours cannot be perfect.:eyebrows::lol:
How do you know we don't have perfect hearing? I'm perfect in every way!
"Sophistry, sophistry - all is sophistry."
AJSki2fly
18-03-2019, 16:56
How do you know we don't have perfect hearing? I'm perfect in every way!
"Sophistry, sophistry - all is sophistry."
That is the point, what may sound perfect to you may not sound perfect to me or Marco or anyone else for that matter, so either all our systems are perfect to us as individuals or they are imperfect as no one can have perfect hearing (or so I believe).:D
Timing is everything and the reconstruction of the original digital recording can so easily be compromised, especially by CD players unless huge effort and expense has been taken to resolve the issues. A spinning disc read by a laser at 500 mph with small scratches and imperfections is not best way to extract digital audio!
Actually the beauty of doing things digitally is that it doesn't matter. You can check your work beforehand to ensure that what is output is correct. Yes if the disc is damaged past a certain point it won't play or will drop out. The trick there is not to damage your discs. Not hard, we manage to do it with vinyl which is far more fragile.
I don't understand why you think reading the file from a hard drive as opposed to a disc makes any difference at all. Hard drives spin pretty fast. Unless it is a solid state job. And why does the fact that it is spinning matter? If you can provide some back up to these assertions I'm happy to take it on board.
[QUOTE=Macca;1070599]
David Blackmers findings are backed by Malcolm Hawksford 6 mins into the video, namely calling for 96Khz sampling and as a result 45khz frequency response.
Likewise this paper identifies muted trumpet harmonics extending well beyond 20Khz
Thankfully recording engineers usually preserves frequencies above 20khz, it is just sad our replay equipment does not presently properly reproduce what was recorded.
some instruments have harmonics out to 100Khz. This doesn't matter because we can't hear them! 22Khz is the upper limit even best case for a child. Once your past thirty 15Khz max is good going. That is the same whether you are listening to a recording of the trumpet or the trumpet-player is standing in your living room. All the things we like about the sound of instruments happens within our range of hearing. And as I said before, 99% of recordings, and all the great recordings, do not contain any frequencies above about 20Khz to begin with because the mics could not capture them.
So not only are we physically incapable of hearing it, it is not there to be heard in the first place.
https://66.media.tumblr.com/tumblr_m1voxst73E1qcay1ao1_500.gif
Yeah there's no real answer to that is there? I'm surprised anyone still clings to the belief that frequencies we can't hear matter. The evidence and indeed the logic are pretty overwhelming.
It's a shame we can't bring Miles Davis et al back to life so they could re-record 'Kind Of Blue' with mics that can capture 100Khz then maybe it wouldn't sound so crap and people might start rating it as a good album.
walpurgis
18-03-2019, 18:46
Yeah there's no real answer to that is there? I'm surprised anyone still clings to the belief that frequencies we can't hear matter.
They do at the recording stage. Harmonic interplay is part of the structure of music and the supersonic element influences what is heard lower down. Once recorded though, the structure of the music is complete and filtering off above the audio band will affect nothing we hear.
I'm surprised anyone still clings to the belief that frequencies we can't hear matter...
So how do super-tweeters work, then?;)
In any case, any chance you could answer some of the posts I've directed at you today, all of which so far have been ignored!:ner:
Marco.
AJSki2fly
18-03-2019, 19:23
So how do super-tweeters work, then?;)
Marco.
Well as according to Martin there is no musical content above 20khz because mics generally used in recording studios do not go above 20Khz (I would agree unless they are using Bruel & Kear high end mics capable of over 20khz, there are a few other manufacturers) then you must be hearing some form of invented sound from the results of digital mathematical interpolations in the DAC processing. :D
So how do super-tweeters work, then?;)
In any case, any chance you could answer some of the posts I've directed at you today, all of which so far have been ignored!:ner:
Marco.
If you can set your super tweeters so they only output above, say 20Hz, switch off the main speakers and play some music, you won't hear a thing. In reality super tweeters have output way down into the audible range. That's how they work. It's just adding an extra tweeter to the speaker. There will be some cancellation but on the whole you'll shelve up the whole top end from maybe as low as 5Khz all the way up. So there's more air and space! But no mystery.
What other posts are you talking about?
Dont hear anything from the bapture tweeters.
Originally Posted by Jimbo View Post
Timing is everything and the reconstruction of the original digital recording can so easily be compromised, especially by CD players unless huge effort and expense has been taken to resolve the issues. A spinning disc read by a laser at 500 mph with small scratches and imperfections is not best way to extract digital audio!
Actually the beauty of doing things digitally is that it doesn't matter. You can check your work beforehand to ensure that what is output is correct. Yes if the disc is damaged past a certain point it won't play or will drop out. The trick there is not to damage your discs. Not hard, we manage to do it with vinyl which is far more fragile.
I don't understand why you think reading the file from a hard drive as opposed to a disc makes any difference at all. Hard drives spin pretty fast. Unless it is a solid state job. And why does the fact that it is spinning matter? If you can provide some back up to these assertions I'm happy to take it on board.
Maybe it’s me but I’m struggling to follow your reasoning. Precise timing in digital circuits is imperative to reliable data transmission and critical in audio circuits especially when converting to digital and back into analogue. It’s also key for reading optical disk as rotation of the disk and synconisation of the read optics is critical, it’s a real time operation we can’t back track and re-read a data block if it was misread or found corrupt. Reading from a HDD is not time critical as we can go back and retry and apply CRC error correction or reallocate a bad block if necessary. Data is buffered in memory and can be clocked out as and when the end device is ready to receive it. The two operations are fundamentally different.
If you can set your super tweeters so they only output above, say 20Hz, switch off the main speakers and play some music, you won't hear a thing. In reality super tweeters have output way down into the audible range. That's how they work. It's just adding an extra tweeter to the speaker. There will be some cancellation but on the whole you'll shelve up the whole top end from maybe as low as 5Khz all the way up. So there's more air and space! But no mystery.
What other posts are you talking about?
Scroll back and note the times today I've quoted you and replied, directing some questions, but you've not responded in return... At a guess, without looking, there must be at least four examples of such. Anyway, it's no biggie; I was really just pulling your leg;)
As for the rest, my point was simply not to underestimate/dismiss out of hand the influence of things that, on the surface, don't appear to matter. It can be a bad mistake to make.
Marco.
walpurgis
18-03-2019, 20:26
Dont hear anything from the bapture tweeters.
Well if you're anything like me Grant, you won't hear much above 10kHz. :)
Maybe it’s me but I’m struggling to follow your reasoning. Precise timing in digital circuits is imperative to reliable data transmission and critical in audio circuits especially when converting to digital and back into analogue. It’s also key for reading optical disk as rotation of the disk and synconisation of the read optics is critical, it’s a real time operation we can’t back track and re-read a data block if it was misread or found corrupt. Reading from a HDD is not time critical as we can go back and retry and apply CRC error correction or reallocate a bad block if necessary. Data is buffered in memory and can be clocked out as and when the end device is ready to receive it. The two operations are fundamentally different.
I agree. Nevertheless the number of times that a cd drops out or fails to read entirely is so few that it isn't worth worrying about. Otherwise the waveform is reproduced perfectly and as I said previously, and as the AES paper I linked to demonstrates, jitter in a cd player (or a DAC) is not sufficiently high to be audible. The ability to re-read a data block or buffer to reduce timing errors therefore confers no advantage over the disc in terms of sound quality. Although I accept that technically it is a superior way of doing things.
Problems that could theoretically cause poor sound as people describe - harsh, hard, lacking focus and soundstage - like distortion from power supply modulation never seem to get talked about and that could equally be a problem in a server-streamer-DAC set up as in a cd player.
Dont hear anything from the bapture tweeters.
But didn't you report you could hear the effect of adding them?
But didn't you report you could hear the effect of adding them?Yup.
Effect isn't sound..
That was my point - you couldn't hear anything coming from them directly, but their addition had an effect on bass perception and presentation.
I can't hear anything above 10kHz, yet I can hear the effect on a violin where everything above 20kHz is filtered out.
I agree. Nevertheless the number of times that a cd drops out or fails to read entirely is so few that it isn't worth worrying about. Otherwise the waveform is reproduced perfectly and as I said previously, and as the AES paper I linked to demonstrates, jitter in a cd player (or a DAC) is not sufficiently high to be audible. The ability to re-read a data block or buffer to reduce timing errors therefore confers no advantage over the disc in terms of sound quality. Although I accept that technically it is a superior way of doing things.
Problems that could theoretically cause poor sound as people describe - harsh, hard, lacking focus and soundstage - like distortion from power supply modulation never seem to get talked about and that could equally be a problem in a server-streamer-DAC set up as in a cd player.
That’s not what the AES paper states, it was found to be audible but at higher rates than expected and it was determined material dependant. Plus training the listeners improved identification of jitter. If the situation was as you describe all cd transports would sound the same, the effects of improving master clocks would have no effect and HDD playback is not better than that of a CD player. I’ll repeat one of Marcos questions have you actually experimented with improving power supplies and master clocks in CD players or HDD based streamers or are you just going by theory?
That’s not what the AES paper states, it was found to be audible but at higher rates than expected and it was determined material dependant. Plus training the listeners improved identification of jitter. If the situation was as you describe all cd transports would sound the same, the effects of improving master clocks would have no effect and HDD playback is not better than that of a CD player. I’ll repeat one of Marcos questions have you actually experimented with improving power supplies and master clocks in CD players or HDD based streamers or are you just going by theory?
The AES paper demonstrates that to make jitter audible, even with the trained listeners, the level had to be way above that generated by any cd player or DAC. I don't see how it can be argued otherwise.
As to your second point I'd consider experimenting with a clock upgrade as measuring the level of jitter then upgrading the clock, then measuring again to see if the jitter level has been reduced.
Then listen to the player against an unmodded version and swap between them blind to see if you can distinguish any difference to a degree of confidence.
If after the upgrade measured jitter was reduced and if you can pick a difference between the players blind to 90% confidence then you might credibly say that the reduction in jitter could be changing the sound. Although there is a possibility that depending on what the clock upgrade entailed other factors could be at play.
That would be experimenting with a clock upgrade. So no, I haven't done that although would be interesting to hear from anyone who has.
Just sending the player off to have it done, getting it back a week or so later and thinking it sounds better is not experimenting with a clock upgrade and isn't going to convince anyone but the person who does it.
Improving power supplies you won't get any argument from me, I agree that can make a difference
Just sending the player off to have it done, getting it back a week or so later and thinking it sounds better is not experimenting with a clock upgrade and isn't going to convince anyone but the person who does it.
It still acts as valid personal experience, from which to make an informed opinion on the matter, rather than simply worshiping at the altar of theory!;)
What would matter to you more, the contents of an AES paper, or the results of what your own ears told you with a clock upgrade, if they contradicted the AES paper? And yes, I'd like an answer to that one please...
And in terms of not convincing anyone but the person who does it, why so?
If I've got a mate who's system I know almost as well as mine, including the CD player, and he's had the clock in the player's DAC upgraded, with no other changes whatsoever carried out to his system, and next time I visit, CDs consistently sound better, with a variety of familiar music, then I'm going to think that the clock upgrade has effected a genuine sonic improvement.
Why wouldn't I?
Marco.
Primalsea
18-03-2019, 22:24
I posed a few questions related to sampling rates a while back and was sent in the direction of some good info. What I didn’t appreciate at that time was that while it was a new discussion for me for some it was an old reoccurring discussion that held some ancient pain and truma for others.
If I have understood what I have since read correctly sampling at a higher rate does reduce an effect due to filter pre and post ringing that smears an impulse transient signal over a longer period of time than the actual original signal. However, while this is important for some industries the actual benefits for audio are claimed to be inaudible, as they are way below the threashold of what we can discern.
IMO which I have admittedly just made up we are talking about very small errors that are completely masked by further filters anyway (d/a anti aliasing, mechanical and electrical filters in the speakers, the air, more mechanical filters - our ears - and then whatever our brain then does) that trying to reduce these arguably inaudible errors is the wrong battle to fight to the point where keeping our ears clean and healthy and learning skills like interval recognition would be far more effective upgrades.
Digital sound quality, all sounds the same and anythings as good as you’ll ever need...not.
https://youtu.be/WjMwGWdqHVQ
The AES paper demonstrates that to make jitter audible, even with the trained listeners, the level had to be way above that generated by any cd player or DAC. I don't see how it can be argued otherwise.
As to your second point I'd consider experimenting with a clock upgrade as measuring the level of jitter then upgrading the clock, then measuring again to see if the jitter level has been reduced.
Then listen to the player against an unmodded version and swap between them blind to see if you can distinguish any difference to a degree of confidence.
If after the upgrade measured jitter was reduced and if you can pick a difference between the players blind to 90% confidence then you might credibly say that the reduction in jitter could be changing the sound. Although there is a possibility that depending on what the clock upgrade entailed other factors could be at play.
That would be experimenting with a clock upgrade. So no, I haven't done that although would be interesting to hear from anyone who has.
Just sending the player off to have it done, getting it back a week or so later and thinking it sounds better is not experimenting with a clock upgrade and isn't going to convince anyone but the person who does it.
Improving power supplies you won't get any argument from me, I agree that can make a difference
The AES paper demonstrates that to make jitter audible, even with the trained listeners, the level had to be way above that generated by any cd player or DAC. I don't see how it can be argued otherwise.
As to your second point I'd consider experimenting with a clock upgrade as measuring the level of jitter then upgrading the clock, then measuring again to see if the jitter level has been reduced.
Then listen to the player against an unmodded version and swap between them blind to see if you can distinguish any difference to a degree of confidence.
If after the upgrade measured jitter was reduced and if you can pick a difference between the players blind to 90% confidence then you might credibly say that the reduction in jitter could be changing the sound. Although there is a possibility that depending on what the clock upgrade entailed other factors could be at play.
That would be experimenting with a clock upgrade. So no, I haven't done that although would be interesting to hear from anyone who has.
Just sending the player off to have it done, getting it back a week or so later and thinking it sounds better is not experimenting with a clock upgrade and isn't going to convince anyone but the person who does it.
Improving power supplies you won't get any argument from me, I agree that can make a difference
I hope my last comment as I fear we are going to go around in circles. Firstly the AES paper makes no mention of jitter being fixed in CD players, in fact it talks about DVD players (semantics maybe) TV recievers etc. Again it does not state *way* above....it states higher than expected ....but still audible under certain conditions. So you’ve never experimented with clock upgrades and have no first hand experience of them and the benefit they can bring...and who said anything about sending your player off, it’s possible to fit an upgraded clock yourself if you have some soldering skill and if not I’m sure there’s somebody on the forum who would help. You are seemingly dismissing the practical experience of others who have tried this, myself on 4 occasions now, as irrelevant yet you are happy to base your argument on a 20 year old paper as gospel with no experience yourself. But you are happy with improvements to power supplies which, funnily enough, are known to influence periodic jitter and improve playback...
You are seemingly dismissing the practical experience of others who have tried this, myself on 4 occasions now, as irrelevant yet you are happy to base your argument on a 20 year old paper as gospel with no experience yourself.
Yup, and unfortunately that stance isn't unusual for Martin, with subjects that challenge the veracity of his (sometimes) entrenched views. I just wish he would concentrate more on basing his opinions, on certain aspects of audio, on the results of his practical experience in those areas, rather than the contents of technical papers, as the latter is no substitute for the former.
Marco.
I can't see how you can have a reasoned valid view on something unless you have practical experience, especially concerning how something will sound?
AJSki2fly
19-03-2019, 07:22
Ok to try and put some reality into this I may be wrong but I seem to remember various manufacturers in the 80’s and early 90’s doing various things to try and improve playback quality of CD’s. In fact these were often referred to in HiFi mags, and were reviewed as an improvement over the manufactures same previous product. One manufacturer that comes to mind is Meridian, if I recall correctly they went through some intense development through the 506 508 CD players that involved oversampling, sampling rates, re-clocking using more accurate chip clocks and various other technical changes to improve CD replay SQ. at the time the last iteration of the Meridian 508 24bit player was considered by a many in the hifi press one of the best CD players to own, there were only one or possibly two others at the time that got close if I recall correctly.
I purchased a 508 24 bit in the late 80’s early 90’s after auditionaing and listening to at least ten other CD players and each time I auditioned I did with it with the same pieces of music and was able to compare side by side and I would insist the dealer switched then blind. In fact I remember spending hours doing this, much to my wife’s annoyance. I did it because at the time spending £1850 on a CD player was a lot of money and a leap of faith.
So my point is this the technical improvements already mentioned undertaken by Meridian and a few others did improve CD playback SQ, so for me it is quite clear that addressing error correction, clocking issues, and anything else that may impact SQ including jitter has to be good.
I can still clearly demonstrate just how good my 508 CD player is to anyone. By accident I discovered a CD that did this. I put on Hawkwind space ritual cd 1 only to here what sounded like a mains loop coming from my hifi. I did everything to eradicate it swapping leads changing mains plugs running dedicated earths etc all to no success. I was confounded, then I put Cd 2 on and the earth loop sound went, suddenly the penny dropped, the Meridian 508 was replaying the earth loop that was actually on the recording, but why had I never heard this before. So up to the loft and out came my old CD player and connected it up to amp and in went cd 1 space ritual and no earth hum. I got my friend to bring his CD player, actually 5 years newer than my 508, and again no earth hum replayed.
So for me the above proves that the more that is done to improve the accuracy of digital data retrieval and how it is processed then the more likely are you to get a more accurate replay close to the original recording. Whether this is CD, digital data files stored on whatever or streamed digital data it still applies in my opinion.
I am certainly no digital expert and I certainly do not fully understand some of the articles recently posted on here in relation to jitter and other topics, but it is clear to me from experience that improvimg approach and technique in the digital realm has improved the resulting sound quality, certainly at the audiophile end of the spectrum. Sadly IMO it has also dumbed down SQ when compression has been used, and moved away from CD quality, to save storage space and enable streaming in large volumes to many end users.
I hope this is not contentious for too many of you, it’s just an observation I thought I would share to try and put a practical experience into play, without getting into arguments quoting technical papers and their validity.
Firebottle
19-03-2019, 08:11
Good post Adrian :thumbsup:
Primalsea
19-03-2019, 08:31
I do believe that your listening experience is as much about you as the equipment you listen to. I wouldn’t be surprised if what you know and understand about the engineering behind hifi effects your listening experience. If you intimately understand the science and engineering of digital audio and accept whar are sometimes sweeping statements as to what is below the limits of perception are you preconditioned to not hearing differences and then vice versa.
AJSki2fly
19-03-2019, 09:16
I do believe that your listening experience is as much about you as the equipment you listen to. I wouldn’t be surprised if what you know and understand about the engineering behind hifi effects your listening experience. If you intimately understand the science and engineering of digital audio and accept whar are sometimes sweeping statements as to what is below the limits of perception are you preconditioned to not hearing differences and then vice versa.
I do not think that I am necessarily pre-conditioned as you imply. As I said I did many auditions and in at least 5 different hifi shops and often repeated listening to the same pieces of equipment side by side and as blind as I could with the dealer in control of which was out on. I just noted the track and sequence and my thoughts on how it sounded, at the end end he told me with CD related to which track. I then reviewed what I had written.
As I said I can clearly demonstrate with at least one CD how a 508 CD player it is better than another manufactures, and I expect there are others out there now that are better than a 508 as design and digital manipulation has moved and and hopefully improved.
Also I do not actually think I "intimately understand the science and engineering of digital audio", I have a bit of general information about what it is and how it is done, but I am certainly not an expert and do not fully understand the impacts that changes to how musical data processing affects how were hear it or the end result. I just rely on my ears as to whether it sound OK to me or not. I suspect that the majority on this forum are the same, although I do know there are a few who are very knowable in this area.
Primalsea
19-03-2019, 10:42
I think that everyone is effected by pre-condition to some degree but it is not impossible overide it. I agree that you need just listen to a change over a period of time and keep swapping back and then listen again for a few days and then swap back again. For me are differences that I can hear and discern and there are differences that I can sense but not necessarily clearly say why. Sometimes I can change something in my hifi and over a few days I just don’t get the same satisfaction as I did before. Recently I came to the conclusion that I prefer it when I have turned the upsampling on my music server off. I can’t say exactly why just that I enjoy listening more that way.
AJSki2fly
19-03-2019, 13:31
Yes we all have pre-conception or an anticipation of what something we are going to try or change may do for us, in this case a piece of of his or a change to it. I think that if you try and remain open minded and think it may be better or worse or in my view I may like but someone else may dislike it, then you have a chance of improving what and how you listen.
I have recently made a series of changes to my turntable to try and improve its musical rendition, mainly trying to remove feedback type resonances and get better isolation. So changes worked and others didn't. But I took my time and listened carefully and for a period of time, not a few tracks but for many days until I decided there was or was an improvement to what I was hearing in the end rendition and across various types of music. The key is patience and let you ears and brain decide whether you like something or not.
most things i try have made little if any difference; some backwards... But occasionally you move forward. Mine is where its likely to remain ability wise and not at all sure flinging money at it will improve it much if at all.
AJSki2fly
19-03-2019, 14:21
most things i try have made little if any difference; some backwards... But occasionally you move forward. Mine is where its likely to remain ability wise and not at all sure flinging money at it will improve it much if at all.
Yes I quite agree, sometime changing a key part of a system amp, speakers or input does not necessarily introduce and improvement, sometimes just something that sounds different.
I am facing a similar issue on the turntable front I have tweaked and tuned it probably as far as I can to get the best out of it to my ears anyway. So the question is now just how good is it really, maybe I will never know unless I stick another one next to it and am able to listen back to back. If I did that would what I chose be better than my existing set up or just different? I think that’s very hard to establish. What I do believe is that to get small gains in audio improvement, whatever that might be real or imagined, becomes very quickly exponentially costly.
Light Dependant Resistor
19-03-2019, 14:28
I have written to IASA below, who are " The International Association of Sound and Audiovisual Archives (IASA) is a professional association concerned with the care, access and long-term preservation of the world’s sound and moving image heritage. Through its active worldwide membership and training initiatives, IASA supports and advocates the development of best professional standards and practice amongst organisations and individuals which share these purposes "
Dear IASA
I am interested in becoming a member of IASA. however wish to ask before doing so, Is there technical ability yet or endeavour, to raise the audio bandwidth of CD to say 45khz with Nyquist /Shannon sampling then at 96khz - allowing a 6 khz margin for filtering ?
The benefits or suggestions of benefit to a better audio experience have been raised in the past by Dr Malcolm Hawksford, Paul McGowan of PS audio and David Blackmer founder of DBX
I look forward to your reply
Cheers / Chris
It still acts as valid personal experience, from which to make an informed opinion on the matter, rather than simply worshiping at the altar of theory!;)
What would matter to you more, the contents of an AES paper, or the results of what your own ears told you with a clock upgrade, if they contradicted the AES paper? And yes, I'd like an answer to that one please...
And in terms of not convincing anyone but the person who does it, why so?
If I've got a mate who's system I know almost as well as mine, including the CD player, and he's had the clock in the player's DAC upgraded, with no other changes whatsoever carried out to his system, and next time I visit, CDs consistently sound better, with a variety of familiar music, then I'm going to think that the clock upgrade has effected a genuine sonic improvement.
Why wouldn't I?
Marco.
Okay let's look at the Audicom clock mod. Is it just the clock that they change out or is there more to it?
https://i.imgur.com/bEvsjJ1.jpg
That's a lot of gubbins. I've got CD players with less gubbins than that in total. So what does it include?
Not only are three separate high precision regulators on the PCB, a large number of Sanyo Os-Con's and Black Gate's latest Capacitors support the Circuitry. The Board uses more than just the usual two layers and everything has been optimised to the highest degree possible. This extends to the Wiring on the clock output that is high purity Silver, chosen over coax cable after extensive tests.
Now I know we both agree that changing the value of a component, or even keeping the value the same and changing it for one of a different material construction, can change the sound. So given that this is not simply swapping the crystal out for a better one how can you say for certain that it is reduced jitter that is responsible for the sound of the player being different after the mod? This looks like a whole raft of changes and boutique components.
Given that you have no before and after jitter measurements and given the findings of the AES study that shows jitter has to be at high levels before we can hear it, and given that Sony were unlikely to have overlooked jitter reduction in their flagship DAC and transport combo, can you really still say for certain that the differences you heard were due to a reduction in jitter? Or would you now agree that there are more variables involved than you had previously considered?
Digital sound quality, all sounds the same and anythings as good as you’ll ever need...not.
Careful with that strawman Eugene. I mean Gaz.
Careful with that strawman Eugene. I mean Gaz.
No it's not... :)
No it's not... :)
Sorry, I'm not allowed to argue with you unless you pay.
Sorry, I'm not allowed to argue with you unless you pay.
That's the Spirit... :D
Okay let's look at the Audicom clock mod. Is it just the clock that they change out or is there more to it?
https://i.imgur.com/bEvsjJ1.jpg
That's a lot of gubbins. I've got CD players with less gubbins than that in total. So what does it include?
Not only are three separate high precision regulators on the PCB, a large number of Sanyo Os-Con's and Black Gate's latest Capacitors support the Circuitry. The Board uses more than just the usual two layers and everything has been optimised to the highest degree possible. This extends to the Wiring on the clock output that is high purity Silver, chosen over coax cable after extensive tests.
Now I know we both agree that changing the value of a component, or even keeping the value the same and changing it for one of a different material construction, can change the sound. So given that this is not simply swapping the crystal out for a better one how can you say for certain that it is reduced jitter that is responsible for the sound of the player being different after the mod? This looks like a whole raft of changes and boutique components.
Given that you have no before and after jitter measurements and given the findings of the AES study that shows jitter has to be at high levels before we can hear it, and given that Sony were unlikely to have overlooked jitter reduction in their flagship DAC and transport combo, can you really still say for certain that the differences you heard were due to a reduction in jitter? Or would you now agree that there are more variables involved than you had previously considered?
The electrical components used on the board combine to perform the function of the clock upgrade, and so their effect can't be assessed individually. It's cumulative, thus the board acts as one part (replacing the existing 'gubbins'), and so that's how its effect is assessed.
Obviously I can't say for certain if the clock upgrade factually reduced jitter, as I didn't measure it (although Mark may have on his test bench), but since you're a fan of technical studies, there are plenty around*, documenting evidence of clock upgrades reducing jitter in the digital domain, for me to believe that they're intrinsically linked, notwithstanding numerous testimonials from people, such as Neal, who've had first-hand experience of carrying out such upgrades themselves, and of course my own.
In that respect, it would certainly have been interesting to have taken some pre and post clock upgrade jitter measurements, and I'll ask Mark if he did, and what his views are on the matter, as I'd be confident that results would've backed up what my ears clearly indicated. The bottom line though, is that the clock upgrade carried out to my Sony DAC fundamentally improved its sonic performance, which demonstrated that there was mileage to be had in that area, even with 'flagship' gear, as indeed there will almost certainly be with any of your CDPs.
* Here's a nice wee paper for you to read: http://digitalassets.lib.berkeley.edu/techreports/ucb/text/EECS-2007-96.pdf ;)
Marco.
From direct experience:
Know that upgrading the clock in my Oppo 203 Blu-ray player noticeably improved everything, including Redbook CD playback.
Video improvement was off the scale, but CDs sounded much better too.
The electrical components used on the board combine to perform the function of the clock upgrade, and so their effect can't be assessed individually. It's cumulative, thus the board acts as one part (replacing the existing 'gubbins'), and so that's how its effect is assessed.
Obviously I can't say for certain if the clock upgrade factually reduced jitter, as I didn't measure it (although Mark may have on his test bench), but since you're a fan of technical studies, there are plenty around*, documenting evidence of clock upgrades reducing jitter in the digital domain, for me to believe that they're intrinsically linked, notwithstanding numerous testimonials from people, such as Neal, who've had first-hand experience of carrying out such upgrades themselves, and of course my own.
In that respect, it would certainly have been interesting to have taken some pre and post clock upgrade jitter measurements, and I'll ask Mark if he did, and what his views are on the matter, as I'd be confident that results would've backed up what my ears clearly indicated. The bottom line though, is that the clock upgrade carried out to my Sony DAC fundamentally improved its sonic performance, which demonstrated that there was mileage to be had in that area, even with 'flagship' gear, as indeed there will almost certainly be with any of your CDPs.
.
no - paper you linked to does not document a clock upgrade reducing jitter. That aside I am not disputing that a clock upgrade could reduce jitter. My points are 1) Jitter was almost certainly inaudible to begin with so even if you reduce it further you will not gain any audible benefit. 2) That the process of modifying the DAC could introduce changes in the sound unconnected with the level of jitter.
Without more controls there is no way to say that the reduction in jitter created the subjective improvement in sound quality.
I think Mark is unlikely to have made any measurements but it would be interesting if he has.
Both you and Gaz are far overestimating the effect jitter has on the sound and our ability to hear those effects. And are therefore erroneously concluding that the beneficial effects of the modifications you made are due to reduced jitter. The vast bulk of the evidence indicates that whatever improved the sound it was not a reduction in jitter.
Here there are some files with various amounts of jitter added http://archimago.blogspot.com/2018/08/demo-musings-lets-listen-to-some-jitter.html
If you can't be arsed to read it all and listen to the files here's the conclusion (my bold):
For me, even with a synthetic sweep, it took >100ns of cumulative sinusoidal jitter in the frequencies used (+/-225Hz, +/-1125Hz, +/-2475Hz distribution with decreasing amounts for each as above) before I could hear distortion! This is obviously a level of sideband distortion significantly beyond what we see with CD/DAC measurements. As I said a couple weeks ago, finding jitter distortion is certainly not impossible out in the wild... But it will be like looking for the giant panda. You'll only rarely find it when you know where to look.
Over the years, I have discussed my suspicions about devices like "regenerating" USB hubs, "purifiers", passive filters, doubts about digital cables of normal length supposedly capable of reducing jitter, and I've read articles from folks who seem to find (and hear) jitter everywhere! The designers and manufacturers have had every opportunity to demonstrate that these devices are able to do something to improve sound quality. The fact that they don't show objective results for their products and in fact essentially ignores calls to produce evidence when confronted is telling
I'm getting bored now, mate [big yawn], as this is going round in circles. Come back to me when you've got some practical experience with clock upgrades, and we'll talk then;)
Marco.
Not so sure about jitter. Jitter sounds like tape wow and flutter. We've all owned or heard a cheap cassette deck (except maybe the youngsters) so we all know what that sounds like. The wobble on sustained notes. Who's ever heard a digital source sound like that? (unless it's actually on the recording like on the Boston album 'Third Stage' for example).
Even that cheapo DVD player you found in a skip has better jitter performance than any RTR deck or turntable by quite a large margin. There needs to be a lot of jitter before it is possible to hear it. http://www.ultrahighendreview.com/uploads/documents/Benjamin_and_Gannon_on_Jitter.pdf
Going back to one of your earlier posts and you state Jitter sounds like tape "wow and flutter." This shows a complete lack of understanding of what jitter sounds like and what it is. Because of this statement of *bollocks* then I feel all that you have written about this subject and indeed all your comments on digital audio need to be considered with a high deal of suspicion as to their validity.
I am by no means an expert on digital audio but I do know some of the facts which not only have I read and studied, more importantly I have demonstrated practically to myself through listening experience which is what I really think you really ought to try. Before you cast around saying that other people either on here or people I have suggested are flawed in their arguments you should take or should I say LISTEN to digital audio yourself and do the practical experimentation.
You can evaluate clocking and separate power supplies yourself at modest expense nowadays and you will see what their effect is on jitter. You can start with an old antique CD player if you like but far better investigate FBA audio as it gives a far better SQ and is easy to work on.
I'm getting bored now, mate [big yawn], as this is going round in circles. Come back to me when you've got some practical experience with clock upgrades, and we'll talk then;)
Marco.
Sorry Martin, have to agree with Marco here.
In my own case there were very noticeable improvements after fitting an upgraded clock to my Oppo.
Subjective...sure.
But importantly measurable too: the new clock specifically designed to lower the amount of measurable jitter.
So it’s plainly cause and effect, not me reading something into it that’s not there.
Just sayin’ :thumbsup:
I see Martin is following his usual path of denial :)
They are reasons that the power supply to a clock is also as important as the quality of the clock itself imho
For example the latest fad for fetmo clocks is quite something, uber low jitter sepc's the best yet etc. However EVERYTHING in the clock signal path way has an effect, personally I use low phase noise clocks for me they produce a far more natural sound, however the correct ultra quiet power supply rail to the clock is AS important to reaping these benefits.
Having fitted over 200 aftermarket clocks in the last 15 years, NO one has returned the units and decided it has made no difference what so ever.
So according to Martin's logic either they are over 200 people fooling themselves that it does make a different or like naim owners convincing themselves it does sound better than the rest of the audio world products :lol:
Or a lot of people are embarrassed but are too afraid to speak up?
Jitter is important in the context of the individual piece of equipment and how it is implemented, however the sum of the parts make the whole.
You can measure the before and after clock output frequency signals when you change a clock whether this relates directly to improved sound quality is as debatable as a class 'a' monster power amp is the epitome of quality audio play back.
I see Martin is following his usual path of denial :)
They are reasons that the power supply to a clock is also as important as the quality of the clock itself imho
For example the latest fad for fetmo clocks is quite something, uber low jitter sepc's the best yet etc. However EVERYTHING in the clock signal path way has an effect, personally I use low phase noise clocks for me they produce a far more natural sound, however the correct ultra quiet power supply rail to the clock is AS important to reaping these benefits.
Having fitted over 200 aftermarket clocks in the last 15 years, NO one has returned the units and decided it has made no difference what so ever.
So according to Martin's logic either they are over 200 people fooling themselves that it does make a different or like naim owners convincing themselves it does sound better than the rest of the audio world products :lol:
Or a lot of people are embarrassed but are too afraid to speak up?
Jitter is important in the context of the individual piece of equipment and how it is implemented, however the sum of the parts make the whole.
You can measure the before and after clock output frequency signals when you change a clock whether this relates directly to improved sound quality is as debatable as a class 'a' monster power amp is the epitome of quality audio play back.
Agreed re: phase noise, more of an effect I think than a clock that is still super accurate a thousand years from now.
Power supply quality is vital too.
of course...assuming IF any of this stuff makes a difference ;)
Bigman80
21-03-2019, 16:20
I see Martin is following his usual path of denial :)
They are reasons that the power supply to a clock is also as important as the quality of the clock itself imho
For example the latest fad for fetmo clocks is quite something, uber low jitter sepc's the best yet etc. However EVERYTHING in the clock signal path way has an effect, personally I use low phase noise clocks for me they produce a far more natural sound, however the correct ultra quiet power supply rail to the clock is AS important to reaping these benefits.
Having fitted over 200 aftermarket clocks in the last 15 years, NO one has returned the units and decided it has made no difference what so ever.
So according to Martin's logic either they are over 200 people fooling themselves that it does make a different or like naim owners convincing themselves it does sound better than the rest of the audio world products :lol:
Or a lot of people are embarrassed but are too afraid to speak up?
Jitter is important in the context of the individual piece of equipment and how it is implemented, however the sum of the parts make the whole.
You can measure the before and after clock output frequency signals when you change a clock whether this relates directly to improved sound quality is as debatable as a class 'a' monster power amp is the epitome of quality audio play back.
But we all know it is :lol:
Sorry Martin, have to agree with Marco here.
In my own case there were very noticeable improvements after fitting an upgraded clock to my Oppo.
Subjective...sure.
But importantly measurable too: the new clock specifically designed to lower the amount of measurable jitter.
So it’s plainly cause and effect, not me reading something into it that’s not there.
Just sayin’ :thumbsup:
No need to apologise. I think it is good to have a debate on the subject and I don't take it personally although the attempts to discredit me rather than the independent evidence and information I have put forward are somewhat tiresome. As for those telling me I have no experience because I have not paid some bloke to put some different parts into a cd player - that's quite funny.
I spent too much wasted time and money in this hobby listening to the one-sided argument put forward by the magazines who I now know were wrong way more often than they were right, but in the absence of any conflicting point of view I accepted unquestioningly. Now at least with forums you can read more than one point of view and make your own mind up or even go and research yourself.
Tony - Some people still not getting the fact that I am not arguing that they did not hear an improvement with their upgrades but that they have offered no evidence that the improvement was due to decreased jitter whereas I have presented evidence showing that that it is highly unlikely to be the reason. It may be that is wrong but no-one has put forward anything that even remotely convinces me otherwise so far. 'No-one has sent one back' will probably not stand up at the AES and it won't with me either. You need to demonstrate cause and effect to a much higher standard. I have explained why already.
For my own part I have owned many and still own quite a few CD players. (Twenty-odd). They range from budget all the way up to flagship, with all forms of processing - multibit, MASH, Bitstream, upsampling and across many manufacturers and ranging from 1984 to 2008 in date of origin. I have also had a number of DACS and have tested them using various CD players, DVD players and BD players as transports.
My untested, uncontrolled subjective opinion (since that seems to be the gold standard of proof in this discussion despite the title of the thread) after 30 years of trying to get a high quality sound out of digital is that as far as sound quality (not presentation) is concerned, the amplification is by far the most influential factor. Over whatever topology, cost, age, make, spec or whatever of the CD player. None of my Cd players exhibits any sort of hash, grain, harshness or glare through my present system. They do not all sound the same but some are better than others. Why?
Why indeed? - and of course once we ask this question it is where we leave the realm of subjectivity - where all we need to do is listen and decide which one we like best - and move into the realm of objectivity. And that is where the 'An audio reviewer said it was because of this' or 'A bloke trying to sell me stuff said this was the reason' should not get put forward as reasoned argument. If really you want to know why you need to get into the realm of measurements, controlled testing and so forth. If you don't fancy that that's fine - just stick with 'I had a listen and it was better' and leave it at that.
Going back to one of your earlier posts and you state Jitter sounds like tape "wow and flutter." This shows a complete lack of understanding of what jitter sounds like and what it is. Because of this statement of *bollocks* then I feel all that you have written about this subject and indeed all your comments on digital audio need to be considered with a high deal of suspicion as to their validity.
.
So what does jitter sound like then?
I'm sorry but I have tried and failed to find the site where you can compare music files with no jitter and with lots of jitter. I don't think I dreamed it! But on those files at least you will have to trust me it sounds exactly like tape wow. However in searching for them I did end up reading another fairly large amount of jitter-related material and it seems that it is very hard to pin down what jitter sounds like. Obviously there are different types of jitter which confuses the matter. Anyway I found the following opinions from various 'boffins':
It sounds like tape wow and flutter
It masks fine details
It makes the sound harsh and hard
It is impossible to pin down what jitter will sound like as its subjective effect on the sound will be system dependant.
of course all of these statements are based on jitter being at audible levels to begin with.
So what does jitter sound like then?
I'm sorry but I have tried and failed to find the site where you can compare music files with no jitter and with lots of jitter. I don't think I dreamed it! But on those files at least you will have to trust me it sounds exactly like tape wow. However in searching for them I did end up reading another fairly large amount of jitter-related material and it seems that it is very hard to pin down what jitter sounds like. Obviously there are different types of jitter which confuses the matter. Anyway I found the following opinions from various 'boffins':
It sounds like tape wow and flutter
It masks fine details
It makes the sound harsh and hard
It is impossible to pin down what jitter will sound like as its subjective effect on the sound will be system dependant.
of course all of these statements are based on jitter being at audible levels to begin with.
Jitter can manifest itself in a number of ways audibly, these include, sibilance, narrow stereo image and flat soundstage, poor bass(common) and sharp tinny brass instruments.
I've heard all of those with digital and a good few with analogue. Couldn't tell you why tho.
I can say the better quality boards you put in a pi for instance improves things steadily.
Whether that is jitter at the base of it I don't know. Even if jitter is measurably lower doesn't gaurentee that was the main reason of course.
Personally I dont worry too much as long as its better
I've heard all of those with digital and a good few with analogue. Couldn't tell you why tho.
I can say the better quality boards you put in a pi for instance improves things steadily.
Whether that is jitter at the base of it I don't know. Even if jitter is measurably lower doesn't gaurentee that was the main reason of course.
Personally I dont worry too much as long as its better
All I want to know is how to improve digital sound to get it as good as possible and therefore try and identify the problem areas and come up with a solution that will fix them. Each step I have taken has given sound improvements, the first being to put the CD player in the bin!:lol:
Light Dependant Resistor
22-03-2019, 01:06
All I want to know is how to improve digital sound to get it as good as possible and therefore try and identify the problem areas and come up with a solution that will fix them. Each step I have taken has given sound improvements, the first being to put the CD player in the bin!:lol:
Essential ( I think :eyebrows: ) is to retrieve the CD player from the bin.
Begin by asking yourself what is right or wrong with the audio your CD player provides, what make and model is it ?
a easy path usually is to update the analogue stage opamps, but before that download a schematic and have a look at what style of opamp is needed
which might be what is called DIP size or soic . I recently did the op amps in my Sony CDP555ES and replaced with burr Brown opa2107
which for the Sony I could not ask for better audio performance.
If your player has DIP size opamps already then it will be much easier to change, if soic yes it can be done but requires much care and skill as soic are very small indeed.
Essential ( I think :eyebrows: ) is to retrieve the CD player from the bin.
Begin by asking yourself what is right or wrong with the audio your CD player provides, what make and model is it ?
a easy path usually is to update the analogue stage opamps, but before that download a schematic and have a look at what style of opamp is needed
which might be what is called DIP size or soic . I recently did the op amps in my Sony CDP555ES and replaced with burr Brown opa2107
which for the Sony I could not ask for better audio performance.
If your player has DIP size opamps already then it will be much easier to change, if soic yes it can be done but requires much care and skill as soic are very small indeed.
I was thinking some of the finer CD players had discrete amp circuits? Which i suppose could really open up the possibility of upgrading?
Russell
Essential ( I think :eyebrows: ) is to retrieve the CD player from the bin.
Begin by asking yourself what is right or wrong with the audio your CD player provides, what make and model is it ?
a easy path usually is to update the analogue stage opamps, but before that download a schematic and have a look at what style of opamp is needed
which might be what is called DIP size or soic . I recently did the op amps in my Sony CDP555ES and replaced with burr Brown opa2107
which for the Sony I could not ask for better audio performance.
If your player has DIP size opamps already then it will be much easier to change, if soic yes it can be done but requires much care and skill as soic are very small indeed.
Cheaper and far more effective bang for bucks is to buy a RPi with Allo Boss DAC, from what I have heard using this with FBA sounds better than any CD player.
Bigman80
22-03-2019, 08:33
Here's the best explanation of Jitter I've found and it's by the Guru, Hans!
From 4:56 is the most relevant.
https://youtu.be/grzoqEb2KMk
Firebottle
22-03-2019, 09:30
'French cauliflower' - brilliant.
Bigman80
22-03-2019, 09:38
'French cauliflower' - brilliant.
Yeah, that tickled me too. I still wouldnt eat them lol
walpurgis
22-03-2019, 09:43
Here's the best explanation of Jitter I've found and it's by the Guru, Hans!
From 4:56 is the most relevant.
https://youtu.be/grzoqEb2KMk
Nice explanation.
He is the man. He has covered so many areas of digital audio on the HB project and is an impartial journalist.
Bigman80
22-03-2019, 10:00
Essential ( I think :eyebrows: ) is to retrieve the CD player from the bin.
Begin by asking yourself what is right or wrong with the audio your CD player provides, what make and model is it ?
a easy path usually is to update the analogue stage opamps, but before that download a schematic and have a look at what style of opamp is needed
which might be what is called DIP size or soic . I recently did the op amps in my Sony CDP555ES and replaced with burr Brown opa2107
which for the Sony I could not ask for better audio performance.
If your player has DIP size opamps already then it will be much easier to change, if soic yes it can be done but requires much care and skill as soic are very small indeed.
I had a Sony CDP555ES. It wasn't something i think would compete with the Pi+Allo combination i have now. What i heard from it was pretty typical of every CD player i have heard. I imagine the BBOA made the tone better but can the opamp help with Jitter?
One you get away from all of those moving mechanical parts in CD players, everything gets significantly better IMO.
Macca made a point that digital is far better when you have sufficient quality amplification. I would totally agree with that. Analogue has sounded great on every amp i have had but digital has struggled to lay a scratch on the vinyl in comparison. Now I am running the Krell and removed as much of the mechanical aspect of digital replay, the sound easily compares with the vinyl and considering the vinyl front end is worth around £4K and the Raspberry Pi, Allo Dac and LPSU cost me under £200, its absolute sickening. I am seriously considering chucking my analogue set up on ebay while it still has value, but i'd never sell my vinyl so i may as well keep it. That's probably the only reason it's still here.
I have never been concerned previously, that Vinyl would be killed off. CD has still got it's finger nails on the edge of the cliff in terms of sales but i will say this with utter confidence, As soon as someone makes the Pi + DAC a plug and play device with this level of quality, it WILL take over. The only thing holding it back is the lack of support when setting one up. Acronyms like NAS, J25 blah blah blah are not user friendly. How DO you set up a NAS? Do you pay? where do you get the files from? How do i connect a pi to Volumio? How do i create an image file? What is mounting a file? Do i plug the pi into the Router? What CAT cable should i use? Is wifi ok? .........AND MORE. That is the barrier to the Pi being in every living room in the country. Thats also why CD players and Vinyl still get played. They do all the work for you. Connect it up, switch it on drop the needle or put a disc in and you are good to go.
This, in my opinion, is the only barrier to world domination for the new generation of streamers.
Bigman80
22-03-2019, 10:07
Nice explanation.Hans is very good indeed. Jimbo highlighted him a while back but I have only been watching his stuff recently. Very knowledgeable and comes across as very unbiased.
Martin, I said earlier that the discussion was becoming circular and boring, so that's why I've ignored the rest of your previous post, as responding would simply continue a currently (pointless) circular argument. However, I'll highlight this bit, as it's important, and something that you apparently don't seem to 'get':
I think it is good to have a debate on the subject and I don't take it personally although the attempts to discredit me rather than the independent evidence and information I have put forward are somewhat tiresome. As for those telling me I have no experience because I have not paid some bloke to put some different parts into a cd player - that's quite funny.
Funny or not, the fact is you have no direct personal experience of assessing the effect of a clock upgrade, in any of your digital equipment, in the context of your system, and no amount of research/paper reading on the subject can make up for that.
Therefore, your stance on this matter is akin to that of a cable cynic, who's only ever used freebie interconnects, telling everyone that anything better doesn't make a difference because 'science says so', which simply doesn't add any value to the discussion.
Like most folk here, I'm only really interested in the opinions people form, based on their practical listening experience, not what they've learned from reading technical papers, as anyone can do that, and thus turn the discussion of audio into a form of scientific study.
*That* [last bit] is fundamentally NOT what AoS is about, and indeed flies completely in the face of our ethos!:doh:
AoS is primarily about members sharing their personal experiences of using hi-fi stuff, whatever it may be, and relating their *subjective* findings to others, in the hope that it will be of some benefit, so that they can *learn from each other*, not from books or technical studies, and subsequently use what is learned that way to improve their systems and enjoyment of music.
That is why this forum exists and has a predominantly subjectivist, 'ears first' ethos, and in turn is why the results of people's listening experiences will ultimately always be what matters most.
In that respect, as far as you're concerned, I care about what YOU think, based on what YOU have heard, and the conclusions YOU have reached on that basis, not what you've read about, and more importantly, I trust *your* ears, not the ears/hearing acuity of those whose input has been used to compile a CES paper.
In that respect, I guarantee that if you were to try a clock upgrade for yourself, and report your experiences here, whether positive or negative, people would sit up and take notice, as they'd be much more interested in your findings there, than you spouting technicalities, which neither you nor most here fully understand.
So why not give it a go?:)
Marco.
orchardaudio
22-03-2019, 10:36
Fact is:
In the digital domain 1-bit equates to 6dB of SNR so:
16-bit = 96dB
24-bit = 144dB
32-bit = 192dB
In the analog domain however even 24-bits cannot be realized, there are no DACs that have a SNR of 144dB.
Hence 32 bits will not yield any better results than 24, because of limitations in analog electronics.
although i often upscale to 24/96 and occasionally higher, I have to say Im not at all sure there are any benefits. DSD is another matter, and im still exploring this. Tidal masters is again another thing, and not as obvious as it says. Sometimes its beneficial sometimes its worse..
Bigman80
22-03-2019, 10:45
although i often upscale to 24/96 and occasionally higher, I have to say Im not at all sure there are any benefits. DSD is another matter, and im still exploring this. Tidal masters is again another thing, and not as obvious as it says. Sometimes its beneficial sometimes its worse..I spent some time with Tidal, I hated it.
I spent some time with Tidal, I hated it.
I dont find it s basic 16 bit stuff any different really to qoboz tbh.(mostly) Still using both, but will have to make a choice sometime soon lol or i will end up paying 2 subs
Bigman80
22-03-2019, 10:54
I dont find it s basic 16 bit stuff any different really to qoboz tbh.(mostly) Still using both, but will have to make a choice sometime soon lol or i will end up paying 2 subsI can't accept having to keep a sub going to have access to music. I've set up a NAS at home and am ripping CDs or downloading FLAC files. I've filled a 500gb hard drive and I'm now starting to look for a 1TB one. I'll eventually set up a proper NAS with built in storage.
I can't accept having to keep a sub going to have access to music. I've set up a NAS at home and am ripping CDs or downloading FLAC files. I've filled a 500gb hard drive and I'm now starting to look for a 1TB one. I'll eventually set up a proper NAS with built in storage.
its a point of view certainly. I have had one for a few years. Tried most. Ive got about 800 albums stored of my own but it is a great way to find new stuff.
Bigman80
22-03-2019, 11:00
its a point of view certainly. I have had one for a few years. Tried most. Ive got about 800 albums stored of my own but it is a great way to find new stuff.I see the appeal mate, but it's not for me. I usually have a browse through YouTube and see what's new. Then I'll either Rio their cd or download.
I like to know I have a copy. If Tidal goes down, you've got no music!!!!
I spent some time with Tidal, I hated it.
Lol - I love it! What did you hate about it?:)
Marco.
I can't accept having to keep a sub going to have access to music. I've set up a NAS at home and am ripping CDs or downloading FLAC files. I've filled a 500gb hard drive and I'm now starting to look for a 1TB one. I'll eventually set up a proper NAS with built in storage.
Ah, so does that answer my previous question? If so, then fair enough.
However, I've got 6TB worth of music, spread over three HDs, not to mention about 2500 CDs, and I still find having access to an 'online jukebox' handy!
I don't use it for 'serious' listening, as for me NO online streamed music is of sufficient quality to replace that available from local streaming of such at home, or with CDs, but in terms of discovering new music or checking out something first before buying or downloading it, Tidal (and other services like it) is invaluable:)
Marco.
Bigman80
22-03-2019, 11:11
Lol - I love it! What did you hate about it?:)
Marco.I had to use bubbleupnp which was horrible so didn't help but what bugged me was usually there was 3/4 versions of one album. I then spent 20 minutes trying to pick the one that sounded best. When I turned it all off and come back to it later, I then couldn't find the same bloody version again.
It's almost too much choice.
My HDD has the best versions (subjective best) of the album's I want so there's no messing about.
I was spending an age looking for music to listen too by artists I knew and I didn't sound any better than using my home NAS setup which is a laptop and HDD.
I just didn't see the point in paying £20 a month for all that aggro
Fair enough, mate. See my previous post. I use Tidal (and other such services) in a very different way:)
Marco.
Light Dependant Resistor
22-03-2019, 11:26
Fact is:
In the digital domain 1-bit equates to 6dB of SNR so:
16-bit = 96dB
24-bit = 144dB
32-bit = 192dB
In the analog domain however even 24-bits cannot be realized, there are no DACs that have a SNR of 144dB.
Hence 32 bits will not yield any better results than 24, because of limitations in analog electronics.
I find it beneficial to never create boundaries or say no, as its just a matter of applying the right principles to find a solution.
As example 24 bit recording and playback was possible in November of 1998 and remains possible with digital - you will
though need a type 4 DBX compander to assist.
Another example think beyond the ubiquitous bridge rectifier and deploy mosfets thyristors and current regulation within the bridge
there are endless possibilities when you do not include, the word limitations and no.
Martin, I said earlier that the discussion was becoming circular and boring, so that's why I've ignored the rest of your previous post, as responding would simply continue a currently (pointless) circular argument. However, I'll highlight this bit, as it's important, and something that you apparently don't seem to 'get':
Funny or not, the fact is you have no direct personal experience of assessing the effect of a clock upgrade, in any of your digital equipment, in the context of your system, and no amount of research/paper reading on the subject can make up for that.
Therefore, your stance on this matter is akin to that of a cable cynic, who's only ever used freebie interconnects, telling everyone that anything better doesn't make a difference because 'science says so', which simply doesn't add any value to the discussion.
Like most folk here, I'm only really interested in the opinions people form, based on their practical listening experience, not what they've learned from reading technical papers, as anyone can do that, and thus turn the discussion of audio into a form of scientific study.
*That* [last bit] is fundamentally NOT what AoS is about, and indeed flies completely in the face of our ethos!:doh:
AoS is primarily about members sharing their personal experiences of using hi-fi stuff, whatever it may be, and relating their *subjective* findings to others, in the hope that it will be of some benefit, so that they can *learn from each other*, not from books or technical studies, and subsequently use what is learned that way to improve their systems and enjoyment of music.
That is why this forum exists and has a predominantly subjectivist, 'ears first' ethos, and in turn is why the results of people's listening experiences will ultimately always be what matters most.
In that respect, as far as you're concerned, I care about what YOU think, based on what YOU have heard, and the conclusions YOU have reached on that basis, not what you've read about, and more importantly, I trust *your* ears, not the ears/hearing acuity of those whose input has been used to compile a CES paper.
In that respect, I guarantee that if you were to try a clock upgrade for yourself, and report your experiences here, whether positive or negative, people would sit up and take notice, as they'd be much more interested in your findings there, than you spouting technicalities, which neither you nor most here fully understand.
So why not give it a go?:)
Marco.
Okay repeating myself again but what can I do. I have not said that the clock upgrades did not offer a subjective improvement.
If I get the clock upgraded on one of my players and I decide it sounds better, how do I know simply from listening that is due to reduced jitter and not some other factor?
Anyone can answer not just Marco. I would really like to know.
Martin, ask yourself what other factors an upgraded clock could possibly offer...and ask yourself what all these clock upgrades are advertised as offering, like Ticord and Tent etc.
The discussion, Martin, isn't just about jitter. It's about digital audio in general, and some supposed "facts and realities" in relation to that, which *in reality* aren't facts [certainly indisputable ones], but more people's views on the subject, based on their personal experiences, which is why the latter is so important;)
None of us here can prove beyond question whether clock upgrades to digital equipment reduces jitter, but the feeling is that if you've had direct experience of carrying out such an upgrade on your own equipment, then you'll both have a better idea of the sonic effect and subsequent subjective impact on jitter, than you would from simply from having read about it in a technical paper or otherwise.
As I've said, ultimately what matters most on AoS are people's subjective experiences and opinions, not what's written in technical papers:)
Marco.
Martin, ask yourself what other factors an upgraded clock could possibly offer...and ask yourself what all these clock upgrades are advertised as offering, like Ticord and Tent etc.
I posted a picture of a clock mod earlier. Who is to say that simply swapping the resistors and capacitors for a different brand (aside from any changes to the circuit itself) could not be reducing noise of some other kind? Or enhancing leading edges a little more, or some other pleasing effect on the sound completely unrelated to jitter?
What the manufactures say that they do is neither here nor there. They can claim anything they like especially if no-one ever asks for any proof of the claims.
But enough - I think the AES paper is enough evidence to make this question important, you, for some reason, don't. I'll leave it there. It's been an interesting discussion anyway.
I posted a picture of a clock mod earlier. Who is to say that simply swapping the resistors and capacitors for a different brand (aside from any changes to the circuit itself) could not be reducing noise of some other kind? Or enhancing leading edges a little more, or some other pleasing effect on the sound completely unrelated to jitter?
What the manufactures say that they do is neither here nor there. They can claim anything they like especially if no-one ever asks for any proof of the claims.
But enough - I think the AES paper is enough evidence to make this question important, you, for some reason, don't. I'll leave it there. It's been an interesting discussion anyway.
I don’t because I as others, have found audible benefits from reclocking and implementing improved clock upgrades. The dogma you’ve shown and absolute closed mind ‘fact’ statements you’ve made based on theory supposedly supported by a 20year old AES paper hold no water when put to the practical test of trying it for yourself. Have you considered that the basis of the paper may actually be wrong? Reality trumps theory in every case.
Do you know Guido Tent? He worked for Philips on consumer electronics back in the ‘90s, an extremely talented engineer in audio electronics and DACs in particular. You may do well to read his web site and try one of his clocks, I had the very fortunate opportunity to talk to him many years ago and if he as a designer says this works like this and will improve the sound of your CD player by reducing jitter then you would be stupid not to take notice. That was when he was still at Philips.
As to your first paragraph, you are obviously not technically minded as Marco answered that neatly in an earlier post. As you say enough I can’t be arsed anymore with your entrenched view.
More personal attacks. It's really all you have in the way of argument isn't it? Why would it make a difference that the paper is 20 years old? Has human hearing evolved in that short time? The fact that you would even use the age of the research as an argument demonstrates that you are clutching at straws. I won't be responding on this topic any more so feel free to have the last word.
https://i.imgflip.com/yzyu3.jpg
To change tack and go back to basics...
First let me state that with a good DAC I'm very happy with 16/44.1.
I often see it stated that red book sampling rate is perfectly good for reproducing a sine wave. That's dandy but we don't listen to pure sine waves. The waves we listen to have incredibly varied amplitude. I can freehand link up something like 12 dots to make a decent fist of a sine wave. If I tried to do the same with a wave of varying amplitude how many dots do I need to accurately show the direction changes? A lot more for sure. I'm not saying that 16/44.1 can't do this, clearly it can or it gets very close. It's that statement about accurately reproducing a sine wave which seems like the wrong test.
Can 16/44.1 accurately reproduce a real-world jagged curve in all cases?
Firebottle
26-03-2019, 11:49
Can 16/44.1 accurately reproduce a real-world jagged curve in all cases?
Yes but only including frequencies below the nyquist limit of 22KHz.
Yes but only including frequencies below the nyquist limit of 22KHz.
That's easily said but I'm looking for something deeper. An infinite number of samples would provide a perfect curve as there would be no curve extrapolations required. Clearly we don't have an infinite number of samples at any frequency so what are the limiting factors? I'm sure that in at least say 99.5% or 99.9% of the time the curve is accurate (or is that accurate enough, but who decides?). Are there dynamic situations where there can be a tiny error which may or may not be audible?
is it like measuring the coastline? the smaller the ruler the more accurate? ie the coastline becomes longer
Don’t forget the analogue signal is bandwidth limited to 20khz, assuming the input signal is not allowed to clip, 44.1Khz will sample everything correctly...the only contentious issue will be the quality/type of the input brick wall filter. If you believe that we can hear or sense information above 20khz then there’s is an argument that 44.1khz doesn’t cut it and we need sampling greater than that....in either case the input signal still needs to be bandwidth limited.
What I'm trying to get is how do we know that between the samples at say time points 4 and 5 there isn't a "squiggle" we're missing? We rely on Nyquist and Shannon theorems but there may be a point where they break down. Agreed it's probably not of practical importance but unless we know the point at which they break down we can't be sure.
https://upload.wikimedia.org/wikipedia/commons/thumb/5/50/Signal_Sampling.png/300px-Signal_Sampling.png
Because the signal is bandwidth limited. It can break down if the signal is allowed to clip but that’s poor mastering, it shouldn’t be allowed to happen. The sampling is continuous not stepped or point in time so there are no gaps.
Because the signal is bandwidth limited. It can break down if the signal is allowed to clip but that’s poor mastering, it shouldn’t be allowed to happen. The sampling is continuous not stepped or point in time so there are no gaps.
So from 20Hz to 22.5kHz it's not actually sampling, it is in fact recording everything. Sampling suggests taking readings at set points.
My bad, poor choice of words, the samples are discrete points in time of a bandwidth limited continuous signal. If you wish to capture a signal which lays between your example of sample 4 and 5 IE: above 1/2 Nyquest frequency you need to sample at a higher rate...
Primalsea
26-03-2019, 20:19
To change tack and go back to basics...
First let me state that with a good DAC I'm very happy with 16/44.1.
I often see it stated that red book sampling rate is perfectly good for reproducing a sine wave. That's dandy but we don't listen to pure sine waves. The waves we listen to have incredibly varied amplitude. I can freehand link up something like 12 dots to make a decent fist of a sine wave. If I tried to do the same with a wave of varying amplitude how many dots do I need to accurately show the direction changes? A lot more for sure. I'm not saying that 16/44.1 can't do this, clearly it can or it gets very close. It's that statement about accurately reproducing a sine wave which seems like the wrong test.
Can 16/44.1 accurately reproduce a real-world jagged curve in all cases?
I think what you are asking is something that I have wondered about in the past. This is what happens if a sharp transient initiates between sampling points? The transient will not be detected until the next sample point meaning that during playback it will be played slightly later than it actually occured in real life?
Although it is easy to visualise this “change of direction” between the sample points This does not actually happen in real life with recorded sound, due to something called pre and post ringing. This is a gradual build up and fading away of a response to a transient or impulse that happens in electronic filters as well as microphones and our ear drums, which are essentially mechanical filters. Increasing the sample rate does improve the response to these transients of the electronic filters used in sampling and playback by reducing the pre/post ringing duration but it is argued that this improvement is not audiable and I think this is where the contention lies. Possibly with exceptionally well performing hifi equipment someone who knows their system and the music well and is skilled and experienced in listening can detect this, maybe. If they are able to decern what are essentially very small differences in timing then possibly they can hear a difference of a more accurate clock.
I’m not sure that I have explained this very eloquently or completely accurately but maybe someone else can do a better job? I do openly admit that I am out of my depth somewhat.
That is indeed what I was getting but if I am correctly interpreting what Neal said, there are no gaps between samples. If so I don't get why sample graphs are drawn with gaps...or they should be blocks rather than lines but that would imply inaccuracy. Something doesn't stack up.
Light Dependant Resistor
27-03-2019, 00:34
I think what you are asking is something that I have wondered about in the past. This is what happens if a sharp transient initiates between sampling points? The transient will not be detected until the next sample point meaning that during playback it will be played slightly later than it actually occured in real life?
Although it is easy to visualise this “change of direction” between the sample points This does not actually happen in real life with recorded sound, due to something called pre and post ringing. This is a gradual build up and fading away of a response to a transient or impulse that happens in electronic filters as well as microphones and our ear drums, which are essentially mechanical filters. Increasing the sample rate does improve the response to these transients of the electronic filters used in sampling and playback by reducing the pre/post ringing duration but it is argued that this improvement is not audiable and I think this is where the contention lies. Possibly with exceptionally well performing hifi equipment someone who knows their system and the music well and is skilled and experienced in listening can detect this, maybe. If they are able to decern what are essentially very small differences in timing then possibly they can hear a difference of a more accurate clock.
I’m not sure that I have explained this very eloquently or completely accurately but maybe someone else can do a better job? I do openly admit that I am out of my depth somewhat.
What actually happens is digital being sampled has 1's and 0's , so a error including jitter distortion interprets wrongly a 1 as a 0, ( naughty digital !!! ) and in the process changes the intended signal usually irreversibly,
however two mechanisms attempt to correct this, error correction - and interpolation, which are both attempts at correction. The better solution is to try to avoid any mechanism for jitter to occur in the first place.
Light Dependant Resistor
27-03-2019, 04:40
Which leads to an interesting point, instead of quickly making information up via interpolation, and sometimes still getting the data wrong, CD players were initially able to
very slowly read all available data into a memory, there would then be a much better chance of data being read correctly when playing if the memory was referenced.
The thing is, would we patiently wait for them to effectively buffer their content ? which might take a while.
When Red book came out memory was silly expensive and pretty small scale. You couldn't save enough data to make it worthwhile.
Also the system they use is pretty good if not absolutely perfect. (nothing is) and cheap to implement. CD was meant to take over from tapes anyway
That is indeed what I was getting but if I am correctly interpreting what Neal said, there are no gaps between samples. If so I don't get why sample graphs are drawn with gaps...or they should be blocks rather than lines but that would imply inaccuracy. Something doesn't stack up.
Basically for convieneince. The correct way is a lollypop graph, the stair step graphs that we've been fed with for decades are technically incorrect and confusing, there are no gaps in a sampled, correctly bandwidth limited signal. This is still a great video from Xiph.org FF to 6mins, explains it all....and at 20:55 https://www.xiph.org/video/vid2.shtml you'll need chrome or firfox to view.
Basically for convieneince. The correct way is a lollypop graph, the stair step graphs that we've been fed with for decades are technically incorrect and confusing, there are no gaps in a sampled, correctly bandwidth limited signal. This is still a great video from Xiph.org FF to 6mins, explains it all....and at 20:55 https://www.xiph.org/video/vid2.shtml you'll need chrome or firfox to view.
The video helps greatly, thanks Neal. I suppose I'll just have to take the scientific view that the space between two lollipop heads can ever show any variance from a perfect curve. Whether it can or not I expect in reality is immaterial or 99.9% likely to be so. Other factors such as jitter are of far greater importance.
I still can't help having a nagging doubt in the back of my mind. The digital conversion we're told is perfect but is it really 100.00% perfect? Maybe it's 99.99% perfect but is a digital error of say 0.01% better or worse than an analogue error of 0.1% or even 1%. The analogue error maybe euphonic or habituated so even though it's an order of magnitude (or more) that any remotely possible digital error the aural effect may differ, some people may prefer one or the other. It's all a bit academic, even I hold my hands up and admit my digital playback is superb and every "bit" as good an my analogue. Vinyl has its issues with noise and dirt and computer audio has it's issues too as it's not as stable as I'd hope (there's often some connectivity issue or update).
Stratmangler
27-03-2019, 09:20
computer audio has it's issues too as it's not as stable as I'd hope (there's often some connectivity issue or update).
If you have stability issues then you need to explore other ways of doing things.
My Squeezebox setup is very, very stable, but there again, I'm not trying to reinvent the wheel all the time, and I'm not drawn in by the latest in visual rendering.
My server is local, so doesn't have to go to the www. whenever I want to play local material (which is not the case for some player distros), and all of the connections for critical equipment are wired.
If you have stability issues then you need to explore other ways of doing things.
My Squeezebox setup is very, very stable, but there again, I'm not trying to reinvent the wheel all the time, and I'm not drawn in by the latest in visual rendering.
My server is local, so doesn't have to go to the www. whenever I want to play local material (which is not the case for some player distros), and all of the connections for critical equipment are wired.
Stability or lack of is one of the reasons I moved away from the Pi for the moment. The SBCs I have are ethernet wired but the GUI is on a phone via wifi. Some little glitch with the wifi can be a minor hassle from time to time. Then there's the hassle of getting a new router and having set up the static addresses again along with a new IP address for the music library - yes my library is local but I want to access it from my chair so wifi is unavoidable. There's also updates to the likes of Volumio - either don't do them regularly and experience pain when you do or take them regularly with less pain until one doesn't work. The UI I have with JPLAY Femto is so very good that is is currently my preference, it's Bubbleupnp on my phone, this is the most stable so far but even so I couldn't give it anyone non-IT savvy. I can fix all the minor foibles but I spend my day in front a computer already...
I could run JPLAY Femto totally off grid but I'd then need go to the monitor and keyboard to get it to play music. It's finding the lesser of all evils - so far it's Bubbleupnp and JLAPY Femto. BTW my wifi signal is strong but there's also quite a lot of congestion so I end up playing with wifi channels every so often as the auto selection isn't as good as I'd like.
CDPs are easier but then huge CD collections are a hassle as is finding stuff. Oh it's just everyday life and I'm moaning about it....1st world problems!
Surely sampling is by definition being selective, and so what a waveform is doing between sampling points is disregarded, which of course raises the question of whether or not the sampling rate is high enough to present a facsimile of the original waveform which is indistinguishable from the original.
The period between the samples is neglected, and what occurs in the waveform in this neglect, cannot be even detected, let alone reproduced.
Stratmangler
27-03-2019, 10:10
Stability or lack of is one of the reasons I moved away from the Pi for the moment. The SBCs I have are ethernet wired but the GUI is on a phone via wifi. Some little glitch with the wifi can be a minor hassle from time to time. Then there's the hassle of getting a new router and having set up the static addresses again along with a new IP address for the music library - yes my library is local but I want to access it from my chair so wifi is unavoidable. There's also updates to the likes of Volumio - either don't do them regularly and experience pain when you do or take them regularly with less pain until one doesn't work. The UI I have with JPLAY Femto is so very good that is is currently my preference, it's Bubbleupnp on my phone, this is the most stable so far but even so I couldn't give it anyone non-IT savvy. I can fix all the minor foibles but I spend my day in front a computer already...
I could run JPLAY Femto totally off grid but I'd then need go to the monitor and keyboard to get it to play music. It's finding the lesser of all evils - so far it's Bubbleupnp and JLAPY Femto. BTW my wifi signal is strong but there's also quite a lot of congestion so I end up playing with wifi channels every so often as the auto selection isn't as good as I'd like.
CDPs are easier but then huge CD collections are a hassle as is finding stuff. Oh it's just everyday life and I'm moaning about it....1st world problems!
Have you tried piCorePlayer?
It just works, and reliably so.
The issue with a number of server/client setups is that the interface goes to an offsite server, which then looks back down the connection for rendering devices on which to play the files.
Talk about long winded, and it's a dependent on that 2 way communication.
If the 2 way connection doesn't work properly the whole shebang goes tits up.
LMS on piCorePlayer deals with local files locally.
If you want to stream from a 3rd party content provider then you need a decent internet connection, but you really shouldn't need a decent internet connection to play local material.
quite liked pcp when I used it, but my memory is poor and when i went back to it i kept getting lost :doh:
Have you tried piCorePlayer?
It just works, and reliably so.
The issue with a number of server/client setups is that the interface goes to an offsite server, which then looks back down the connection for rendering devices on which to play the files.
Talk about long winded, and it's a dependent on that 2 way communication.
If the 2 way connection doesn't work properly the whole shebang goes tits up.
LMS on piCorePlayer deals with local files locally.
If you want to stream from a 3rd party content provider then you need a decent internet connection, but you really shouldn't need a decent internet connection to play local material.
I haven't tried piCorePlayer though have thought about it in the past. It's not the player stability that's the issue really, it's the networking if wifi is involved to a hand held GUI. Wifi was fine when there were only about 3 device in the house using it, now we have more like 15 to 20 connected, probably I just need a better router. But....that again is more hassle.
Stratmangler
27-03-2019, 10:35
I haven't tried piCorePlayer though have thought about it in the past. It's not the player stability that's the issue really, it's the networking if wifi is involved. Wifi was fine when there were only about 3 device in the house using it, now we have more like 15 to 20 connected, probably I just need a better router. But....that again is more hassle.
Better routers cost bigger money.
ISPs don't send out routers capable of dealing with lots of clients simultaneously. They always send the least expensive device they can get away with.
Most of the IT companies I do infrastructure installation for use Draytek routers, and don't even bother unboxing the ISP provided kit.
Nearly all the network capable stuff here is hard wired, barring the Squeezebox Touch in my son's room, the Chromecast, the laptops, the tablets, and the smartphones, which amounts to 9 devices.
The Chromecast gets used in preference to the Virgin V6 box because the Chromecast blocks all the advertising. The Virgin box breaks up long Youtube videos with adverts.
always fancied my own router, but dont know enough about them to confidently play about so stick with the crap you get :lol: TBH, i dont get much trouble with dropouts etc. Laptop to RPI is all wired anyway. Ive got an ethernet switch as ive a few things on cables. Just my tablet and phone mostly that are wireless.
Agree about the ISP routers. Hardwiring is hard over 4 floors, I have some powerline plugs to use when absolutely necessary - they work very well but I don't want the pollution.
Is a digital life preferable to an analogue one?
ha.. im multicultural.
just me in house now so I can do what I want
...and prance around the place, dressed as you wish:eyebrows:
Marco.
...and prance around the place, dressed as you wish:eyebrows:
Marco.
indeed, and in any dress i choose :ner:
Stratmangler
27-03-2019, 12:42
I have a friend who lives in a 4 storey house.
He has a wireless access point in the bottom storey, and another 2 floors up.
Add in the ISP's router, and he has superb WiFi coverage.
My friend lives in an old stone house, where the lowest level is below street level, but backing out onto a canal.
Mostly our wifi is fine, I use a similar placement to you friend with a 4 storey house but without a repeater - coverage even on the top floor is fine. Indeed my music room is just one wall and floor away, strength is good; it's the competing signals or router which can't cope with so many connections. It only gets its knickers in a twist every couple of weeks or so and it take seconds to sort out usually, but it's not fit for non-IT iterate people.
I use the powerlines if I need loads of bandwidth at the extremes of the range.
Basically for convieneince. The correct way is a lollypop graph, the stair step graphs that we've been fed with for decades are technically incorrect and confusing, there are no gaps in a sampled, correctly bandwidth limited signal. This is still a great video from Xiph.org FF to 6mins, explains it all....and at 20:55 https://www.xiph.org/video/vid2.shtml you'll need chrome or firfox to view.
What happens Inbetween the samples?
Surely sampling is by definition being selective, and so what a waveform is doing between sampling points is disregarded, which of course raises the question of whether or not the sampling rate is high enough to present a facsimile of the original waveform which is indistinguishable from the original.
The period between the samples is neglected, and what occurs in the waveform in this neglect, cannot be even detected, let alone reproduced.
Excellent post and questions I have pondered over for sometime.
What happens Inbetween the samples?
Seems like it's an educated guess which will mostly be correct....then we're told that nothing weird can happen between samples....this is where my original question came in. Are we relying of 1940's theory here is what I'm wondering; at what point does the theory break down?
The output of a DAC is not a staircase (or collection of lollipops) but the raw data surely is. If it's incontrovertible that the linking of two samples can only ever follow one curve then fine but at my superficial level it's hard to believe.
What happens Inbetween the samples?
At a rate of 44100 samples per second, how big would you expect any 'gap' to be?
As an experiment you might try cutting a hundredth of a second slice out of a guitar note wave form (using an audio editor)
Listen back to the result - do you hear the jump?
At a rate of 44100 samples per second, how big would you expect any 'gap' to be?
As an experiment you might try cutting a hundredth of a second slice out of a guitar note wave form (using an audio editor)
Listen back to the result - do you hear the jump?
I worry that's the logic that's been used. If you don't hear the jump there can't be any sonic impact whatsoever.
I worry that's the logic that's been used. If you don't hear the jump there can't be any sonic impact whatsoever.
I think most of the confusion stems from our complete inability to grasp just how ridiculously fast the sample rate is - and therefore come up with all kinds of straw-clutching problems that don't actually exist.
I think most of the confusion stems from our complete inability to grasp just how ridiculously fast the sample rate is - and therefore come up with all kinds of straw-clutching problems that actually don't hold up.
I expect that is true is reality but it seems 44,100hz isn't overkill, it's just enough but here I'm conflating frequency extension with the gaps between samples.
Is though "there are so many samples, of course it's good enough", a good scientific proof? Can we double the gap or make it 10 times bigger without issue or are we on the edge?
Trying to find a simple explanation of what each 16 bit sample means is surprisingly difficult. Is each sample for the whole frequency spectrum across both channels? Is there one sample per each frequency? This is really basic stuff. There must be a dummies guide somewhere.
I expect that is true is reality but it seems 44,100hz isn't overkill, it's just enough but here I'm conflating frequency extension with the gaps between samples.
I think it has been mentioned a number of times, but worth repeating. The sampling process is continuous - so where would these 'gaps' that are continually being alluded to come from?
It's not like the computer gets really tired between samples and needs a little sit down to recover .. :)
There still seems to be some misunderstanding, there is nothing between the samples. The signal is bandwidth limited. The xiph.org video explains this. Now if you believe that there are signals above 20khz that you can hear or affect the signal below 20khz and you want to capture them, then you need to raise the Nyquest frequency to encompass those signals, let’s say up to 24Khz...then you need a Nyquest sampling frequency of 48khz and the analogue signal to be bandwidth limited to 24Khz.
I think it has been mentioned a number of times, but worth repeating. The sampling process is continuous - so where would these 'gaps' that are continually being alluded to come from?
It's not like the computer gets really tired between samples and needs a little sit down to recover .. :)
But does continuous mean every single millisecond is accounted for? If so then even the lollipop diagrams are not correct. Indeed it's closer to analogue in terms of sampling but with restricted digital levels....though less restricted than analogue!
one of the issues with 24/96 or higher is the room it takes up. It may or may not improve things, depending, but its subjective. I usually upsample but i fear there is little poit and have reduced this. I do quite like the dsd effect of upsampling that way tho. Not sure why, nor do i especially care tbh. hearing results is results to me
Firebottle
27-03-2019, 14:33
Trying to find a simple explanation of what each 16 bit sample means is surprisingly difficult. Is each sample for the whole frequency spectrum across both channels? Is there one sample per each frequency? This is really basic stuff. There must be a dummies guide somewhere.
A 16 bit sample is taken of each channel, 44100 times per second.
That's two 16 bit samples taken every 22.6 microseconds.
Imagine any music waveform you like (band limited) and the instantaneous level is taken every 22.6 microsecond, on a zero to 65536 scale. This gives the 16 bit sample.
Trying to find a simple explanation of what each 16 bit sample means is surprisingly difficult. Is each sample for the whole frequency spectrum across both channels? Is there one sample per each frequency? This is really basic stuff. There must be a dummies guide somewhere.
You need to sample each channel L+R individually for stereo. At the sample point the signal is converted into a number, for 16 bit that’s anywhere from +/- 32,767 Now this throws up an issue with ADC and that’s quantisation error where the sampled signal doesn’t fit exactly into the granularity of 16bits. This is where dither comes into play to add a randomisation to the sample point via the use of shaped noise to reduce this quantisation error. The video explains this as well.
Maybe also worth pointing out that this all refers to the ADC not the playback. As enthusiasts and music lovers we can do sod all about what happened in the ADC process, it’s out of our control. All we can obsess over or worry about is the playback side.
Thanks guys, that clears up where I was hearing conflicting statements.
At a rate of 44100 samples per second, how big would you expect any 'gap' to be?
As an experiment you might try cutting a hundredth of a second slice out of a guitar note wave form (using an audio editor)
Listen back to the result - do you hear the jump?
But the sampling rate may not be enough so when sampled at a higher rate you may be able to hear the difference
Have a look at hearing sensitivity and response to timing with lower 16 bit sampling rate. Higher sampling rates may be closer to the timing issues our hearing may percieve?
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