View Full Version : The Myth of 'High Resolution' audio
There seems to be some misapprehension concerning 'hi-rez' audio which is surprising considering it is not a new thing, having been on the market well over a decade.
Firstly it is nothing like hi resolution video as some folk still seem to think. We all know that the more pixels you have the more detail you get and we have all experienced the clear difference that a hi-res video signal has to a standard definition one. But this is in no way comparative to audio which works completely differently.
I'm not a technical person but it is not that difficult to understand the basics.
We have bits - 16 bit, 48 bit and 96 bit.
We have sampling frequency - 44.1, 96, 192 KHz
Bits is amplitude and sampling frequency is wavelength. As we know any audio signal is a continuous emission of sound at continuously changing wavelengths and volumes
So the more bits you have the higher the dynamic range, the difference in volume between the quietest and the loudest signals on your recording.
The higher the sampling frequency, the higher the frequency of sound that can be re-produced. This is half the sampling frequency. So sampling at 96KHz we can reproduce sounds up to 48 KHz.
Lets take bits first. 16 bits gives a dynamic range of 110 dB. How much do we need? Well a good vinyl Lp might manage 45 dB of dynamic range. And no-one complained about lack of dynamic range on LP. So 110 is more than enough, well more than enough. So no need for more bits than 16
Now frequency - Even if you have really good hearing you will struggle to hear up to 20kHz, most people will not hear above 16Khz. Do we need to be able to reproduce sound above that level? Well it has ben argued that some few instruments do have overtones that go up higher than that even though we don't hear them we do hear their secondary effect at lower frequencies. That is the theory. In practice no-one can reliably tell the difference so if it is improving sound quality it is at best doing it at a tiny, infinitesimal amount.
So why do my hi rez files sound better than my red book CD copy of the same album?
Because when hi-res (SACD and DVD-A) were launched the record companies went back and painstakingly re-mastered a whole slew of titles from the original tapes, many dating back to the early 'Seventies. Needles to say they did a better job (more time, more money, better resources) than was done at the time, consequently they sound a lot better than the originals.
It isn't always that easy to find a 'hi-rez' and a red book version of the same master consequently a lot of the time people are comparing apples to oranges. The record companies are happy to keep that confusion going.
Now I agree that it is worth having some form of 'hi res' replay in order to hear those better masters when they are not also available in 16/44.1 - but can we please stop thinking/saying that there is 'more information' or 'more detail' or 'higher resolution' available because it just 'aint so.:)
Btw, I made a mistake. The discussion we had, which I referred to on the other thread, took place in the mod room, when I addressed, on the following page, this post of yours: http://theartofsound.net/forum/showthread.php?29321-One-Two-Watch!&p=631918#post631918
Don't worry, only those with the relevant permissions will be able to see it! ;)
Marco.
Btw, I made a mistake. The discussion we had, which I referred to on the other thread, took place in the mod room, when I addressed, on the following page, this post of yours: http://theartofsound.net/forum/showthread.php?29321-One-Two-Watch!&p=631918#post631918
Don't worry, only those with the relevant permissions will be able to see it! ;)
Marco.
This is what you said:
Quite simply, the proof of the pudding is in the listening, as I've got identical albums, both on CD, and high-res versions of such on my hard-drive. There are numerous examples where the latter significantly outperform the former, sonically, with no other variables in the equation. I'll demonstrate that to you when you next visit
Different masters. That's all there is to it.
In your opinion, not mine ;)
Marco.
Opinion doesn't really come into it. How do you know they are the same mastering? Increasing the dynamic range above 110 dB and increasing the frequency response above 22KHz makes no difference to what you hear and that is all 'hi rez' audio does - but a better mastering job can make a hell of a difference in SQ.
Of course opinion comes into it; I intrinsically *know* what I can hear (and what I relate that to), and you disagree - and so no amount of words exchanged on a computer screen is liable to change our respective opinions on the matter.
Therefore, if you wish to debate the subject on purely a technical level, I'm afraid that I'm not your man, and will be too busy enjoying the musical benefits provided by the Pi/DAC combo, partly created, IMHO, as a by-product of the latter's high-resolution capabilities ;)
Marco.
Of course opinion comes into it; I intrinsically *know* what I can hear (and what I relate that to), and you disagree - and so no amount of words exchanged on a computer screen is liable to change our respective opinions on the matter.
Therefore, if you wish to debate the subject on purely a technical level, I'm afraid that I'm not your man, and will be too busy enjoying the musical benefits provided by the Pi/DAC combo, partly created, IMHO, as a by-product of the latter's high-resolution capabilities ;)
Marco.
If they are the same master and sound different then I would really like to hear that. But as I said I strongly suspect you are listening to two different masterings and ascribing the improvement to 'high rez' when it is down to a better master.
16 bit is 96db Macca...
Thanks for that. I did say I wasn't technical ;) Doesn't change the basic point of my post, though. 96dB is still a whole lot of dynamic range.
If they are the same master and sound different then I would really like to hear that.
Precisely; and you will next time you pop up for a sesh :)
Marco.
We will have to do that soon.
Seriously though, you did not answer my question: Do you knew for a fact that they are the same master? You did say that you acquired a drive full of files but what info do you have on their pedigree?
Indeed, it's been too long since the last one. However, we should really have a go at sorting out that system of yours first! :eek: :eyebrows:
Some I know for a fact are the same master (as I had a convo with Dunc about it), others not, and my opinion there is based simply on gut instinct, which as you know is something close to my heart ;)
Marco.
Distortion of low level signals is lower with 24 bit quantisation - have you looked at comparison of the waveforms at -90Db 16bit vs. 24bit? Check this out - http://www.stereophile.com/content/musical-fidelity-m1-dac-measurements
Is distortion at such a low level audible? that's another question, but I suspect that it is.
I tend to agree that higher sample rates aren't that big a deal - 48Khz is definitely better than 44Khz, 88.2/96Khz is probably all you need and 192Khz is a waste of time IMO. Quantisation does make more of a difference to me.
OTOH, storage is cheap, so my Vinyl rips are always 24/96Khz for archiving, but I usually downsample to 24/48 for day to day use. What does that tell me? with common source material, you don't give away much quality with integer down sampling 96Khz -> 48Khz. Bit depth reduction from 24 bits to 16 bits with triangular dither isn't a huge deal either. I think that I could tell the difference in an ABX, but I could live with it.
Distortion of low level signals is lower with 24 bit quantisation - have you looked at comparison of the waveforms at -90Db 16bit vs. 24bit? Check this out - http://www.stereophile.com/content/musical-fidelity-m1-dac-measurements
Is distortion at such a low level audible? that's another question, but I suspect that it is.
.
This is where we get into those marginal grey areas which I accept do exist. That's why I mentioned the possibility that frequencies higher than 22KHz, whilst being inaudible to humans, may have an impact on the reproduction of lower frequencies. As you say the additional bits (dynamic range) are not doing anything useful and any case only relate to dynamics and not the 'quality' of the sound reproduction in any case.
The point of my OP was that hi-rez audio is not in any way like hi-rez television. I think it is important that enthusiasts should be aware of that as it is pretty basic and relevant information.
This is where we get into those marginal grey areas which I accept do exist. That's why I mentioned the possibility that frequencies higher than 22KHz, whilst being inaudible to humans, may have an impact on the reproduction of lower frequencies. As you say the additional bits (dynamic range) are not doing anything useful and any case only relate to dynamics and not the 'quality' of the sound reproduction in any case.
The point of my OP was that hi-rez audio is not in any way like hi-rez television. I think it is important that enthusiasts should be aware of that as it is pretty basic and relevant information.
Wish High Rez audio sounded as good as High Rez televisions looks:)
Think -10db is the human threshold of normal hearing so I would doubt it
Stratmangler
02-04-2015, 15:18
We have bits - 16 bit, 48 bit and 96 bit
Do we?
I'm scratching my bits now.
Marco.
I'm scratching my bits now.
Marco.
You need to get some cream for that:)
Clive197
02-04-2015, 15:29
I think there is a degree of truth and sanity in Macca's post.
A number of things come to mind. When I rip a CD to my NAS I am convinced the streamed music sounds better than the original CD, it shoudnt but it does!
Many so called Hi-Rez downloads are taken from a 16/44.1 files and upscaled. That is NOT Hi-Rez. This has been proved by a number of Hi Fi magazines. HiFi News now test downloads (somehow) and give the results.
SACD has a smoother sound IMO but I also have a number of SACD's which sound no different to the original CD (Blood,Sweat & Tears 3 and Abraxas come to mind) which could easily be down to mastering.
I think the jury is probably still out on this one and opinions will go back and forth for an eternity. Please don't start me on BluRay 24/96 files of which I have a number.
Clive
Hi Clive,
A number of things come to mind. When I rip a CD to my NAS I am convinced the streamed music sounds better than the original CD, it shoudnt but it does!
That's also been my experience, since using the Pi. Don't underestimate the (what can only be negative) influence the transport mechanism in the CD player is having on proceedings...
Many so called Hi-Rez downloads are taken from a 16/44.1 files and upscaled. That is NOT Hi-Rez.
That I buy, which I'd term loosely as 'psuedo hi-res'. So what *is* 'bonafide hi-res', then?
Marco.
You need to get some cream for that:)
Yeah... The bonus is, I found a bit of pizza I'd been saving!
Marco.
Think -10db is the human threshold of normal hearing so I would doubt it
Db level of the threshold of hearing is not the same as relative amplitude (e.g. -xDb).
Distortion is also a funny thing - the human response to it is complex and some types of distortion can be heard at low levels whereas we are tolerant of high levels of even order harmonic distortion (hello valve lovers LOL!). I am prepared to believe that low quantisation resolution and low temporal resolution for low level signals is one of the reasons why 16/44 audio sounds 'odd' to many. Yes, I know about information theory, Nyquist-Shannon's sampling theorem etc but in practice decoding isn't the same as having sufficient bits to accurately encode a band-limited signal, otherwise you would see a nice clean sine at -90db, not that noisy pile of shite. That issue can be dealt with simply by going to 24bit and it should be where possible.
So, from my point of view, 'Hi-Res' PCM starts at 24bit 48Khz, and heads rapidly into diminishing returns at higher sample rates. there are reasons to use high sample rates, but they aren't to do with extending HF response, and these can be realised with upsampling. My CD player upsamples to 24bit/352Khz to allow gentle filters to be used that don't have pre-ringing without risking aliasing in the audio band. Even bats have no use for 192Hkz bandwidth!.
Cheers,
Alex
Clive197
02-04-2015, 15:40
........ So what *is* 'bonafide hi-res', then?
Marco.
Now that's a bloody good question!
Clive
Rothchild
02-04-2015, 15:53
Interesting discussion, sorry to pop in with a couple of points of pedantry.
The measure is dB (deciBel as in 'One tenth of a Bel')
If one is offering for instance -10dB it has to be referenced to something:
0dBFS = Full Scale digital, the largest number that can be represented digitally - Works 'down' because over 0dBFS is clipped/distorted so measurements will generally be -xdBFS
0dB SPL = 20 micropascals (Sound Pressure Level) - Works 'up' 0dB SPL is the quietest registerable sound in the real world, and as you can see relates to a real measurable phenomena ie air pressure
0dBu = .775Volts RMS
0dBV = 1Volts RMS, so 0dBV means X=1V RMS, +X dBV means greater than 1V RMS, and -X dBV means less than 1V RMS. Are both measures of voltage gain based on established engineering standards.
I would suggest that if you calibrated your listening environment such that 0dBFS (in your 16 bit system) = 96dBSPL then at -90dBFS things are going to start getting confused with the quiescent noise of your analogue stages, before you'll really hear any of the type of artifacts shown in the linked graphs.
Finally, for clarity on the first post most DACs run at either 16 or 24bit (not 48 96 which is I think just a typo/confusion with sample rate) however most modern studio software will put those 'fixed point' (ie there are a fixed 24bits) in to a 'floating point' container (normally at 32 or 64bit) this enables files to be manipulated well above 0dBFS without detriment (a 32bit float file has ~1500dB of range) however because converters operate with 'fixed point' numbers all signals must be reduced to 0dBFS or less before conversion to avoid clipping and distortion.
It's easy to tell if your 'hi rez' file is the real deal, simply put it through any number of spectrum analyser tools available (I like voxengo SPAN) and see if there's any signal above 22kHz, if it rolls of steeply at that point odd on it's been 'upsampled' from a 44.1kHz source, if there is signal above that point then chances are it's the real mcoy.
Oldpinkman
02-04-2015, 16:08
In your opinion, not mine ;)
Marco.
It's a simple test really. Take the high-res sample and have it down-sampled to red book using good software, and then compare the hi-res original with the down-sampled copy. Both will be the same master, and so Macca's objection will not be valid. I am still playing with this (well was before I got distracted by France) and the jury is out
I think it is absolutely true to say that there is a great deal of potential variability of quality due to mastering techniques and so it is quite likely that a lot of high-res sounds better due to better mastering than due to the medium itself. I'm not yet sure that means high-res is the same as red book when they're both done properly. And I expect to see later on in this thread others challenging Macca's sums
Sure, I'd be up for carrying out any tests, as long as they involve using your ears, and not graphs or measurement equipment... ;)
Like I said, at the moment, I'm pretty confident I can differentiate between 'bonafide high-res' recordings and those that have simply been mastered better, although I couldn't say for sure what the mechanism responsible for causing that difference is.
Marco.
awkwardbydesign
02-04-2015, 16:23
This is what you said:
Quite simply, the proof of the pudding is in the listening, as I've got identical albums, both on CD, and high-res versions of such on my hard-drive. There are numerous examples where the latter significantly outperform the former, sonically, with no other variables in the equation. I'll demonstrate that to you when you next visit
Different masters. That's all there is to it.
I just watched all 1hr 19min of this. https://www.youtube.com/watch?v=SXbH-yzGNfg From an engineer who cares, and there is a LOT more going on than just the mastering.
Recommended.
pitadavespa
02-04-2015, 16:24
I'm scratching my bits now.
Marco.
I hope you use a lower frequencie than 44.1, 96 and 192 KHz...
--
Martin, that's a good point, I guess you are probably right.
Oldpinkman
02-04-2015, 16:46
Sure, I'd be up for carrying out any tests, as long as they involve using your ears, and not graphs or measurement equipment... ;)
Like I said, at the moment, I'm pretty confident I can differentiate between 'bonafide high-res' recordings and those that have simply been mastered better, although I couldn't say for sure what the mechanism responsible for causing that difference is.
Marco.
The test is a listening one. Just make your own low-res file out of a hi-res one. (Or get someone to do it for you)
As you know I am hard case anti-foo. My mate Owen makes me look positively subjectivist. I was sure I could hear a difference between FLAC and WAV files but assumed I was potty. In fact, turns out there are some good data processing reasons why that might be true. (As I mentioned my own researches got interrupted)
On the subject of no foo - never mind expensive linear supplies, treat yourself to one of these and let me know what you think http://www.ebay.co.uk/itm/151445907328 ;)
Lol... Is this the same one: http://www.ebay.co.uk/itm/10400mah-Portable-Battery-Charger-Power-Bank-for-iPhone-6-5S-4-Samsung-HTC-LG-UK/151635570283?_trksid=p2047675.c100012.m1985&_trkparms=aid%3D444000%26algo%3DSOI.DEFAULT%26ao%3 D1%26asc%3D29952%26meid%3D6d0f8e553a974a18977a7e91 d739806e%26pid%3D100012%26rk%3D5%26rkt%3D10%26sd%3 D151445907328
And would it both fit into the Pi's power socket and be electrically compatible with it? If so, I presume that one would simply leave the battery on a continual charge, thus ensuring that it remains on full power, causing no sonic degradation with the Pi, when the battery starts losing its charge? :)
Marco.
Oldpinkman
02-04-2015, 16:58
looks the same to me, but I think you are comparing an unfinished auction price with a buy-it-now price. Either way - less than a decent bottle of wine. I am a bit wary of "float charge" - since any crud in the power supply (and the SMPS supplied with mine is pretty cruddy) can still reflect on the input to the Pi. I think fully charged you'll get at least 5 hours before you need to worry about a recharge. Put a record on , and plug it in to charge again. ;)
Edit - sorry missed the electrical compatibility bit. Yes - no problem - the lead it comes with is a usb to micro usb lead. The fat end in the battery, the thin end in the Pi. It's a USB power supply. Any 5v 2A USB source will work - use the 2 amp socket on the battery for the Pi - recharge it from the Pi's charger which has a micro usb connector on it, which is what the battery needs! You won't break anything
Cool. So, essentially, the Pi replaces the iPhone that the battery was meant to be used with?
Marco.
skimminstones
02-04-2015, 17:51
think youre all looking far too deep into this, just enjoy whatever music you have :D
Oldpinkman
02-04-2015, 18:12
Cool. So, essentially, the Pi replaces the iPhone that the battery was meant to be used with?
Marco.
Yes sort of. The link you showed was for a battery connected to a Samsung. An iPhone has a different sort of connector on the thin end. So the lead needs to be USB to Micro USB, not USB to iPhone. And strictly you need an ipad compatible battery since an iPhone needs less than 1amp. But that battery using the 2 amp socket and you're rocking and rolling
Perhaps then you would kindly link to the exact item I should be buying, so I can simply click on it and get it? :)
Marco.
Btw, I made a mistake. The discussion we had, which I referred to on the other thread, took place in the mod room, when I addressed, on the following page, this post of yours: http://theartofsound.net/forum/showthread.php?29321-One-Two-Watch!&p=631918#post631918
Don't worry, only those with the relevant permissions will be able to see it! ;)
Marco.
How is this possible!?:scratch:
How is what possible, daftee? :)
Marco.
Oldpinkman
02-04-2015, 19:16
Perhaps then you would kindly link to the exact item I should be buying, so I can simply click on it and get it? :)
Marco.
Will do. When sober
Ha - no worries :D
On a Thursday night, too!
Marco.
Lodgesound
02-04-2015, 20:58
Mastering.......now there is a can of worms.
I have many tools for this in my studio including many types of meters, graphs and analysis tools.
The only ones I actually use when I do it are my ears and a pair of monitors that I am familiar with.
Yes there are technical parameters that must be adhered to however an "instinct" of what sounds good is the most valuable asset when applying such corrections.
High-end analogue magnetic tape can play a very major part in the process also.
StanleyB
02-04-2015, 21:15
think youre all looking far too deep into this, just enjoy whatever music you have :D
I agree with you. Let people like me worry about the equipment to make the bits and samples audible.
I lead a very sad life really. I spend a lot of time trying to engineer ways to make what seems impossible, possible. But I am in no way unique in that respect. AoS members such as Paul Haynes, Nick Gorham, Mark Grant, to name a few, are also doing the same thing in their respective field of expertise. So let us not poo these so called myths. Behind every myth there is at least a grain of truth.
So 16bit is 96dB in terms of dynamic range. Is each one of us capable of resolving 96dB with our current equipment? What's the dynamic range of your amp, your speakers, and your cables? HOLD IT! Let's rewind that a bit and play it again: What's the dynamic range of your cables? I bet that's the first time that most of you have been asked that question, let alone consider it appropriate or relevant. Are you saying that it would be a myth that cables are capable of reducing the dynamic range of an audio signal? This would be worthy of a thread in its ow right. So let's leave it at that as far as this thread is concerned.
Going hires is often cited as the path one has to thread in order to hear more detail. Look, I myself have been designing gear to come up with alternative solutions that do not require hires audio files, and with equally satisfying results. But I am not going to pour scorn on the desire of those who do wish to maintain a library of hires audio files to enjoy. Similarly I would not do the same in questioning is choice of drinks .
I would like to bring to the attention of you guys that when the 16 bit CD format hit the market very few CD players used a 16 bit DAC chip. Most were in fact 14 bit, and at least one used a 12 bit DAC chip. The format was devised in such a way that a 16 bit audio data stream could be decoded and played back at a lower bitrate without most of the consumers being able to notice the difference with the equipment available at the time to the average man in the street. Equally, the same 16 bit data can be decoded with a DAC of a higher bitrate capability. Without bitrate accuracy you would be hard pressed to pick up the difference in signal amplitude in a solo classic guitar presentation.
When digital audio switched from a bitrate war to a sample rate war, bitrate accuracy started to fall out of favour as attention focused on ever increasing sample rates. The arrival of asynch USB brought bitrate accuracy back into mainstream discussions since it was cheap to implement compared to other solutions such as expensive clocks.
But why am I mentioning bitrate when we are talking hires? Well first of all hires means 24 instead of 16 bit ;). The sampling rate takes second place from a technical perspective, but is the number one focus of the hires audio file collector because of marketing trends. The sampling rate tends to just make up the numbers in a lot of recordings. In other recordings it can mean the difference between hearing two singers, or hearing two distinct vocals. But you can achieve an almost identical sounding result through bitrate accuracy ;).
But if you haven't got the hardware that can turn that information into more than just a collection of audio files on your hard drive you are wasting your money. However, I am not saying that you should not waste your money that way. If that's what rocks your boat, then let no man stand between you and your choice of music format. I myself started collecting audio compact discs since 1983. It took me another three years before I actually owned a CD player to play them on. And even more amazing is that I still own those discs and play some of them from time to time. And you know what, Those original discs made in 1983 started to sound better over the years. And do you know why? Because digital audio equipment have improved over the years and those 16bit/44.1kHz audio files have benefited from that. Had I got rid of my CD collection I would never have had the chance to hear the results of the level of progress in digital audio design. I might have been tempted into believing that the reason that I can hear so much more from my system is because I am playing 24/192 instead of a 16bit/44.1kHz CD from 1983.
But can I hear a difference between 16/44.1 and the apparently same file in 24/192? A couple of years back the answer would have been NO. But just as with CD beforehand, equipment capable of doing 24/192 more justice has improved. If you have stood still equipment wise in the area where digital audio have made worthwhile improvements then you are not going to notice much or any difference between lores and hires for the foreseeable future. At the same token, it is easy to assume that if a difference can be heard, the two audio files are not of the same parentage or time of conception. But consideration should be given to the fact that equipment capable of hires reproduction has also improved. And they are now so good on occasions that they are beginning to make a small audible difference possible if the source material is content rich.
Yes there are technical parameters that must be adhered to however an "instinct" of what sounds good is the most valuable asset when applying such corrections.
Hear, hear! There's no substitute for the benefits of 'experienced ears' :clap:
What always gets me is why more folk don't have the gumption or confidence to trust their instincts, when it comes to assessing various aspects of audio.....
Marco.
Good post Stan. Either my equ or me aint up to it as I really cant tell the difference, or much of a difference. Sacd's sound the same as red books; possibly worse actually.
When I was at Gary's we listened to a track in hi and redbook and even with his hi quality gear I could only hear a fraction of difference so I guess its me...but I am happy enough with redbook or ripped flac. Vinyl is still no 1 though.
Oldpinkman
03-04-2015, 07:21
Ha - no worries :D
On a Thursday night, too!
Marco.
Sue is an overworked teacher. Yesterday was the last day of term. F*** me it's a long way back to find your link after stans post. I'm having a breather and looking for my sauna. Back soon
Oldpinkman
03-04-2015, 07:24
Lol... Is this the same one: http://www.ebay.co.uk/itm/10400mah-Portable-Battery-Charger-Power-Bank-for-iPhone-6-5S-4-Samsung-HTC-LG-UK/151635570283?_trksid=p2047675.c100012.m1985&_trkparms=aid%3D444000%26algo%3DSOI.DEFAULT%26ao%3 D1%26asc%3D29952%26meid%3D6d0f8e553a974a18977a7e91 d739806e%26pid%3D100012%26rk%3D5%26rkt%3D10%26sd%3 D151445907328
And would it both fit into the Pi's power socket and be electrically compatible with it? If so, I presume that one would simply leave the battery on a continual charge, thus ensuring that it remains on full power, causing no sonic degradation with the Pi, when the battery starts losing its charge? :)
Marco.
This one is fine but has 30 minutes left at time of writing. It comes with the lead you need. In the notes it says if you want to use it with an iPhone you'll need to supply the lead. I'll try and find you a buy it now next
Stratmangler
03-04-2015, 07:27
Sacd's sound the same as red books; possibly worse actually
What are you playing SACD discs on?
StanleyB
03-04-2015, 07:31
F*** me it's a long way back to find your link after stans post.
I can try to delete it if it's in the way of this Pi thread.
Oldpinkman
03-04-2015, 07:43
http://www.ebay.co.uk/itm/External-Backup-Battery-Portable-Charger-12000mAh-for-mobile-phone-iphone-ipad-/181284725215?pt=LH_DefaultDomain_3&hash=item2a356961df
That one will do. Use the 2.1amp socket with the Pi :)
Oldpinkman
03-04-2015, 07:44
I can try to delete it if it's in the way of this Pi thread.
No its fine Stan. It was the hangover talking ;)
What are you playing SACD discs on?
Used to be a denon 2900. I dont have many sacds but I wasn't impressed. Don't even have a sacd player now
This one is fine but has 30 minutes left at time of writing. It comes with the lead you need. In the notes it says if you want to use it with an iPhone you'll need to supply the lead. I'll try and find you a buy it now next
Cheers, dude. I don't have an iPhone, so that's not an issue. I just need a battery that I can use with the Pi, and which comes supplied with all the bits I need, solely for that purpose :)
Marco.
http://www.ebay.co.uk/itm/External-Backup-Battery-Portable-Charger-12000mAh-for-mobile-phone-iphone-ipad-/181284725215?pt=LH_DefaultDomain_3&hash=item2a356961df
That one will do. Use the 2.1amp socket with the Pi :)
Cheers! Soz, replied to your earlier post before I saw this one.....
Marco.
Audio Advent
03-04-2015, 14:46
Back to the OP - I think it does miss some major points of hi-res simply because, as Macca says himself, he's not a technical person.
The praises in the recording world for higher resolutions are exactly about the technical aspects such as the necessary filters used in the initial digitisation processes - by pushing the sample rate so high, you can take pretty much all effects of those filters way out of the audio band. Need at least 88.2kHz to get there and it gets better/easier the higher you go. Then there is impulse response of the digitisation process which improves immensly as you go up the sample rates and impulse response a.k.a "leading edge" is a very important part of making something life-like and "real" sounding. This then also translates into making it a lot easier to get a signal with fewer problems in the D/A process too - e.g. gentler filters which again won't cause pre-ringing etc of imperfect filters into the audible range. Then Stan brings up some good techincal points too which I don't know anything about.
So, if you're not techinical then I say don't bother making any technical judgement of what's good or not, just use your ears and leave it at that and leave an open mind as to future possibilities of high-res. Little information is a dangerous thing is what they say - it leads to incorrect conclusions due to a lack of nuance of understanding. Thankfully there's not much in danger here..
From a consumer's point of view, a lot of it is marketing and you'll find some stuff is only say 16/48 but because it is too much for CD, it will be upsampled and dithered to at least 24/96 as that is a more common standard. But it's not a 24/96 recording! So when the consumer compares and can't hear a difference between 16/44.1 and 16/48 they then believe that they're listening to 24/96 and so dismiss all high-res. Also, much of the high-res we have from the last decade or more simply has been recorded on early high-res digital equipment which probably didn't/hasn't addressed the technical advantages of high-res because people weren't thinking that way back then.
awkwardbydesign
03-04-2015, 15:07
The praises in the recording world for higher resolutions are exactly about the technical aspects such as the necessary filters used in the initial digitisation processes - by pushing the sample rate so high, you can take pretty much all effects of those filters way out of the audio band. Need at least 88.2kHz to get there and it gets better/easier the higher you go. Then there is impulse response of the digitisation process which improves immensly as you go up the sample rates and impulse response a.k.a "leading edge" is a very important part of making something life-like and "real" sounding. This then also translates into making it a lot easier to get a signal with fewer problems in the D/A process too - e.g. gentler filters which again won't cause pre-ringing etc of imperfect filters into the audible range.
Also, as Andrew Sheps says here https://www.youtube.com/watch?v=SXbH-yzGNfg reverberation tails get truncated, and they are one of the clues that make music sound "real". I listen to a lot of music with an "atmospheric" sound (Arvo Part, William Byrd, Coil, Arild Andersen, etc) and the way notes die away is a large part of it. I suspect our differences in listening will strongly influence whether or not we are affected by such things.
Stratmangler
03-04-2015, 15:09
Used to be a denon 2900. I dont have many sacds but I wasn't impressed. Don't even have a sacd player now
I had a dig around for comments about the Denon 2900 and its SACD performance, and the comments are a right old mixed bag.
I have a theory that your problems might have been related directly to gain - DSD masters don't seem to have been afflicted with overly hot mastering, and I think it is probably directly related to DSD being a bits on demand format.
If you have a lot of high energy sound to represent then there just aren't sufficient bits to describe it all, and the sound field starts to collapse.
If the gain isn't high enough to drive your amplifiation then the sound will come across as a bit soft and lacklustre,
Audio Advent
03-04-2015, 16:04
Also, as Andrew Sheps says here https://www.youtube.com/watch?v=SXbH-yzGNfg reverberation tails get truncated, and they are one of the clues that make music sound "real". I listen to a lot of music with an "atmospheric" sound (Arvo Part, William Byrd, Coil, Arild Andersen, etc) and the way notes die away is a large part of it. I suspect our differences in listening will strongly influence whether or not we are affected by such things.
Just started watching it....
Interesting point near the beginning of how people at the beginning of recording technology talked of how recording would "destroy music" because music is all about being in the room with the musicians.
Similar things are said about all sorts still today, whether it's about analogue sound or record sleeves or digital resolution or solid state or or or or ... maybe it's nothing but most people's resistance to change which then becomes emotional which then directly effects our enjoyment/mentality and therefore becomes confirmation bias of the philsophical kind.
awkwardbydesign
03-04-2015, 16:39
There is a kernel of truth in it, though, as music has been devalued into musak, it's just a background noise for people's lives far too often.
awkwardbydesign
03-04-2015, 16:46
I have worked at several places where my co-workers couldn't function without pop radio playing. They got jittery. It seems it is used to stop them being alone with their thoughts, which they find scary. I would prefer silence, as my thoughts don't scare me. They may scare others, of course!
And when they went home they watched the TV, hardly any listened to music.
Audio Advent
03-04-2015, 17:01
There is a kernel of truth in it, though, as music has been devalued into musak, it's just a background noise for people's lives far too often.
I'd say those people have been enabled by technology to do exactly what they've always wanted to do but were restricted by the old tech.
That "value" was never there naturally, it was enforced. But who cares if those of us who do value music are perfectly able to carry on valuing it and do so in so many and more convenient ways?
Distortion of low level signals is lower with 24 bit quantisation - have you looked at comparison of the waveforms at -90Db 16bit vs. 24bit? Check this out - http://www.stereophile.com/content/musical-fidelity-m1-dac-measurements
Is distortion at such a low level audible? that's another question, but I suspect that it is.
I tend to agree that higher sample rates aren't that big a deal - 48Khz is definitely better than 44Khz, 88.2/96Khz is probably all you need and 192Khz is a waste of time IMO. Quantisation does make more of a difference to me.
OTOH, storage is cheap, so my Vinyl rips are always 24/96Khz for archiving, but I usually downsample to 24/48 for day to day use. What does that tell me? with common source material, you don't give away much quality with integer down sampling 96Khz -> 48Khz. Bit depth reduction from 24 bits to 16 bits with triangular dither isn't a huge deal either. I think that I could tell the difference in an ABX, but I could live with it.
Obviously if you record a signal on a system with 96dB dynamic range at -90dB it will be a bit horrible, compared with the same signal on a system with 144dB of dynamic range, but tis is either academic or somebody trying to con the gullible. Only a halfwit would record at -90dB. The vast majority of domestic hifi would be incapable of playing back a -90dB signal with the volume control set normally.
The typical signal to noise ratio of almost all domestic hifi equipment is less than 16-bit, and some of it quite a lot less.
I found it impossible to find a 16/44 file and a 24/96 file from the same master. I have heard a file downsampled from 24/96 to 16/44 then re-upsampled to 24/96. This has the effect that the original and processed files are the same size and treated the same way in the DAC, but that any information in the octave from 22 to 44 kHz have been removed, and any signal which was lower than -96dB in the original has been removed.
I have only done it the once, on a top class system, but not my own, and they sounded the same to me.
This could be expectation bias, though, because I know of no plausible reason why they could sound any different once the key issues are properly handled.
awkwardbydesign
03-04-2015, 18:43
I'd say those people have been enabled by technology to do exactly what they've always wanted to do but were restricted by the old tech.
That "value" was never there naturally, it was enforced. But who cares if those of us who do value music are perfectly able to carry on valuing it and do so in so many and more convenient ways?
I care, in part because I was forced to endure the distorted racket without which they couldn't function. And I actually care about others, too.
And the "value" was that music was something either you made for yourself, or you listened to someone else making it. It wasn't just a convenient way of avoiding thought!
Play with bit depth in JRiver. You only start to be able to hear a difference at 13 bits and less. 24 bits? Think about what voltage the LSB represents.
Obviously if you record a signal on a system with 96dB dynamic range at -90dB it will be a bit horrible, compared with the same signal on a system with 144dB of dynamic range, but tis is either academic or somebody trying to con the gullible. Only a halfwit would record at -90dB. The vast majority of domestic hifi would be incapable of playing back a -90dB signal with the volume control set normally.
The typical signal to noise ratio of almost all domestic hifi equipment is less than 16-bit, and some of it quite a lot less.
I found it impossible to find a 16/44 file and a 24/96 file from the same master. I have heard a file downsampled from 24/96 to 16/44 then re-upsampled to 24/96. This has the effect that the original and processed files are the same size and treated the same way in the DAC, but that any information in the octave from 22 to 44 kHz have been removed, and any signal which was lower than -96dB in the original has been removed.
I have only done it the once, on a top class system, but not my own, and they sounded the same to me.
This could be expectation bias, though, because I know of no plausible reason why they could sound any different once the key issues are properly handled.
Frank,
Yes, I agree - nobody would sensibly record at that level, so it is largely academic. Quantisation noise is an issue at the bottom end of the modulation range - at 90dbf THD noise reaches about 4%, and is still 0.2% at -60dbfs. Is this audible? I don't know. I think I can hear a marginal loss of detail when down-sampling my own recordings from 24bit/96khz to 16bit/48khz with dither . I can't say I hear any difference at 24/48, so that is what I typically use within my own library. I certainly couldn't hear a -90dbfs signal on my system - it would be lost in system and/or background noise.
In your experiment, how did you down-convert and upsample again? by addition of dither at the right level, you can pretty much eliminate quantization noise and achieve 18bit effective resolution which I would expect to be 'good enough' to all intents and purposes. Maybe 24 bits sounds slightly better because of the internal architecture of the DAC and differences in how it processes different word lengths, filter coefficients etc. it may be optimized for 24bit operation over 16bits in some non-obvious way.
Regards,
Alex
Audio Advent
04-04-2015, 00:47
Play with bit depth in JRiver. You only start to be able to hear a difference at 13 bits and less. 24 bits? Think about what voltage the LSB represents.
Overall, I really don't think you can be so logical about what should or shouldn't be audible as firstly it might be audible for completely different secondary reasons (like the practical workings of digital equipment and files) and secondly the improvements may be subtle to the point of effecting long-term listenability rather than snap judgements.
13 bit might be where you can hear an obvious difference instantly but so often distortion presents itself as a feeling in the listener, perhaps of fatigue or general long-term dissatisfaction with the music. A cheap CD player for me I will want to turn down after a while and fatigue will set in, whilst with vinyl I want to turn it up and can listen to for long periods no problem.
Voltage of LSB: probably 20 bit is the practical limit for most good hifi but 24-bit architecture in processor electronics I think is more standard and so easier and cheaper to implement. If you're going to set a standard then 24 bit would be it rather than 18 or 20. So much of digital processing of a decade or so ago had been via 24 bit or 32 bit architecture DSP, I come to that conclusion. 18bit, 20bit ADC chips and DAC chips were the norm at points in history but always followed by adding zeros to the data to then be processed at 24 or 32.
Audio Advent
04-04-2015, 01:01
I care, in part because I was forced to endure the distorted racket without which they couldn't function. And I actually care about others, too.
And the "value" was that music was something either you made for yourself, or you listened to someone else making it. It wasn't just a convenient way of avoiding thought!
Sorry for the misunderstanding, I was only commenting on that particular comment I quoted, not any other of your comments.
Still, now you mention it, I think it is perfectly fine for people to value music as a generally uplifting background noise - it's a different way of valuing it to maybe you and I but it is still giving it value. I don't think people do listen to background music to avoid thought either - that's just your way of trying to understand a behaviour you don't yourself relate to. People listen to background music because it entertains them and actually engages their brain on a musical level they enjoy, instead of being forced to only think about the possibily very boring and brain-dead job that they are compelled to endure day after day after day to pay the mortgage they've found themselves trapped in after following society's expected life path of school, job, family, mortgage, retirement, death. (:lol: this is a jolly post!) Allowing them that IS caring. Turning it off because it annoys you is something else.
Light Dependant Resistor
04-04-2015, 02:52
Frank,
Yes, I agree - nobody would sensibly record at that level, so it is largely academic. Quantisation noise is an issue at the bottom end of the modulation range - at 90dbf THD noise reaches about 4%, and is still 0.2% at -60dbfs. Is this audible? I don't know. I think I can hear a marginal loss of detail when down-sampling my own recordings from 24bit/96khz to 16bit/48khz with dither . I can't say I hear any difference at 24/48, so that is what I typically use within my own library. I certainly couldn't hear a -90dbfs signal on my system - it would be lost in system and/or background noise.
In your experiment, how did you down-convert and upsample again? by addition of dither at the right level, you can pretty much eliminate quantization noise and achieve 18bit effective resolution which I would expect to be 'good enough' to all intents and purposes. Maybe 24 bits sounds slightly better because of the internal architecture of the DAC and differences in how it processes different word lengths, filter coefficients etc. it may be optimized for 24bit operation over 16bits in some non-obvious way.
Regards,
Alex
No its not academic, a studio reel to reel recorder without companding is capable of just 55db( 8.8 bits ) and add 15db if you want to introduce THD by overloading level.
With companding the same recorder be it using Dolby or Type 1 DBX you can get about 85 db. 13 bits,
If we change the recorder we lose the ability of analog to overload gracefully as a digital recorder similarly has possibly worse issues that you cannot overload it
risking sharp cutoff noise hence there is without help from companding,less dynamic range as level has to be reduced.
With companding though which preserves dynamic range a 16 bit recorder is capable of close to 20 bit capability.
My post suggests we start understanding and discussing what is involved with preserving dynamic range be it from super dooper recorders
or methods such as companding, as it is at the heart of the art of sound.( there i said it )
http://theartofsound.net/forum/showthread.php?37814-More-Bits-Dynamic-range
Overall, I really don't think you can be so logical about what should or shouldn't be audible as firstly it might be audible for completely different secondary reasons (like the practical workings of digital equipment and files) and secondly the improvements may be subtle to the point of effecting long-term listenability rather than snap judgements.
13 bit might be where you can hear an obvious difference instantly but so often distortion presents itself as a feeling in the listener, perhaps of fatigue or general long-term dissatisfaction with the music. A cheap CD player for me I will want to turn down after a while and fatigue will set in, whilst with vinyl I want to turn it up and can listen to for long periods no problem.
Voltage of LSB: probably 20 bit is the practical limit for most good hifi but 24-bit architecture in processor electronics I think is more standard and so easier and cheaper to implement. If you're going to set a standard then 24 bit would be it rather than 18 or 20. So much of digital processing of a decade or so ago had been via 24 bit or 32 bit architecture DSP, I come to that conclusion. 18bit, 20bit ADC chips and DAC chips were the norm at points in history but always followed by adding zeros to the data to then be processed at 24 or 32.
Try it. I had a Philips 14 bit CDP that sounded really very nice long term. A16 bit Technics that was hard to listen to for any period of time.
You can also upsample lower sampling rates to much higher but that will definitely give you nothing.
Rothchild
04-04-2015, 07:43
Does anyone know what the response of a high quality tape deck would be at -90dBFS (assuming that -18dBFS = 0dBVU which is a pretty well recognised standard in studios)? - I'd wager that it would generate some equally horrific graphs.
It's true that at the upper limit analogue recording machines are more forgiving (pleasant even) but the main reason for moving to 24bit in a recording context is to ensure that there's enough headroom to never have to get close to full scale (thus avoiding the horrors that await with digital clipping)
Hi Marc,
How are you getting on with that task, relating to my use of the Pi/IQ-Audio DAC, you were going to do for me? :)
Marco.
Rothchild
04-04-2015, 08:39
Hi Marc,
How are you getting on with that task, relating to my use of the Pi/IQ-Audio DAC, you were going to do for me? :)
Marco.
That thread progressed so quickly that I understood you had already resolved the issue, making my writing up of it somewhat moot.
Lol... I thought you were going to do some sort of tweaked 'gain structure' thing, which would optimise my use of the Pi/DAC, or have I gotten that wrong? :)
Marco.
Audio Advent
04-04-2015, 14:56
Try it. I had a Philips 14 bit CDP that sounded really very nice long term. A16 bit Technics that was hard to listen to for any period of time.
You can also upsample lower sampling rates to much higher but that will definitely give you nothing.
The 14 bit Philips CDP wasn't 14 bit though - it uses two 14 bit chips yes, but they work together in a clever way to process and deliver a 16 bit signal with full 16 bit dynamic range.
So they are both 16 bit machines but one CDP sounded better than the other. Distortions which are hard to listen to come from all over, get them in pre-amps, power-amps too, so could be the just the output stage design in the Technics you didn't like (or wasn't working properly), the power supply could be getting a load of noise on it effecting both the signal directly or setting up resonances or overload in the opamps etc etc etc A whole world of possibilities.
To isolate only the bit rate as the variable in a listening test takes a pain in the bum to set up properly, ensuring the same A/D conversion, mastering etc etc. Really needs to be overseen from the studio side. I bet there are some self-recorded files out there to compare though on studio equipment forums.
I had a Philips CD101 too which with my set up at the time sounded a bit harsh over the long term and a Meridian version of it sounded much better, smoother. Then a recent listen to a Marantz CD54 sounded nice and smooth too - all use the same 14 bit TDA1540 chips.
Audio Advent
04-04-2015, 15:01
It's true that at the upper limit analogue recording machines are more forgiving (pleasant even) but the main reason for moving to 24bit in a recording context is to ensure that there's enough headroom to never have to get close to full scale (thus avoiding the horrors that await with digital clipping)
Yeah, good point!
When people talk of 16bit etc , that is the MAXIMUM. I think people forget that. For quiet passages on dynamic music like classical, the dynamic resolution is then down to say 13 bits leaving all the headroom for the loud passages.
So that 13 bit where it is easily discernable that the sound quality is effected comes into play and probably quite often crops up on some classical and for nuances hidden in a mix.
Having 24 bit allows the quiet parts to still have a good dynamic resolution and therefore allows you to "see" into the mix further whilst also giving large headroom so that clipping is never close.
To AA - it was either a CD100 or 101, borrowed while I had the Technics. Via Martin Logans I preferred the much older 4x interpolating Philips.
I agree with you I think the output stage of the Technics was a poor sounding effort.
StanleyB
04-04-2015, 16:40
To AA - it was either a CD100 or 101, borrowed while I had the Technics. Via Martin Logans I preferred the much older 4x interpolating Philips.
Which of the Philips players were 4X interpolating? I can only remember that Marantz brought out a couple of 4X oversampling players.
Both 14 bit machines were 4x oversampling Stan. Don't zeros get inserted for the non-sampled "samples" and then get passed through an interpolator to make up some pseudo sample values?
StanleyB
04-04-2015, 17:12
The chipsets were different as far as I can remember. I probably still got the training and service manuals in my shed somewhere. The only reason I remember this stuff is because I had to endure regular visits to Philips in Eindhoven for training courses in their players. And one of the points that the lecturers were keen to point out were the differences between the way Philips and Marantz approached the D to A processing.
Can't help Stan but Google might.
Re-read my quick explanation and modded it a tinsy bit to make slightly better sense.
May I suggest that audible dynamic range at any one time is only 30db or so, but on a sliding scale. Thirteen bits depth is around that of FM hiss. I'm probably wrong, but do any commercial recordings have music down at minus 60 or lower? The midrange on vinyl is only around minus 30 or 40 tops and around 70db at hf on a mint pressing I remember.
Those early '14 bit' players did struggle to reach 16 bits in a linear fashion, judging by the tests done at the time and distortion at lower levels could be huge. I found the early Philips, Marantz and Meridian MCD could give me a headache after a while, although the next batch - Mission DAD7000, B&O CDX (a hugely pleasant surprise) and my first CD player, the Meridian MCD-Pro, were rather better, but shallow in depth perspective I remember, as were the Sony 101's I heard and demonstrated. The tech matured very quickly though, didn't it and by the late 80's, the best players were and still remain really good indeed :)
Firebottle
04-04-2015, 18:01
I've got a Philips CD-610 that is 4 times oversampling.
First one I had got nicked, nothing special in the sound. Second one I bought as a replacement sounds SO much better, much cleaner and more engaging, how is that?
:) Alan
Audio Advent
04-04-2015, 18:09
Both 14 bit machines were 4x oversampling Stan. Don't zeros get inserted for the non-sampled "samples" and then get passed through an interpolator to make up some pseudo sample values?
From the first Philips models - is how they managed to get a 16-bit equivalence via oversampling and noise shaping with the 14bit dac chip. Here's a summary on this page:
http://www.dutchaudioclassics.nl/the_evolution_of_dac_the_digital_filter/
Audio Advent
04-04-2015, 18:13
May I suggest that audible dynamic range at any one time is only 30db or so, but on a sliding scale. Thirteen bits depth is around that of FM hiss. I'm probably wrong, but do any commercial recordings have music down at minus 60 or lower? The midrange on vinyl is only around minus 30 or 40 tops and around 70db at hf on a mint pressing I remember.
Those early '14 bit' players did struggle to reach 16 bits in a linear fashion, judging by the tests done at the time and distortion at lower levels could be huge. I found the early Philips, Marantz and Meridian MCD could give me a headache after a while, although the next batch - Mission DAD7000, B&O CDX (a hugely pleasant surprise) and my first CD player, the Meridian MCD-Pro, were rather better, but shallow in depth perspective I remember, as were the Sony 101's I heard and demonstrated. The tech matured very quickly though, didn't it and by the late 80's, the best players were and still remain really good indeed :)
Do you mean psychologically/physiologically we can only discern a 30dB dynamic range at any one time?
They did get pretty good pretty quick, quite an achievement. I still favour 20bit DACs+ for CD play back though just to allow for noiseshaping dither trying to squeeze 19bits+ equivalence out of a 16 bit CD.
Frank,
Yes, I agree - nobody would sensibly record at that level, so it is largely academic. Quantisation noise is an issue at the bottom end of the modulation range - at 90dbf THD noise reaches about 4%, and is still 0.2% at -60dbfs. Is this audible? I don't know. I think I can hear a marginal loss of detail when down-sampling my own recordings from 24bit/96khz to 16bit/48khz with dither . I can't say I hear any difference at 24/48, so that is what I typically use within my own library. I certainly couldn't hear a -90dbfs signal on my system - it would be lost in system and/or background noise.
In your experiment, how did you down-convert and upsample again? by addition of dither at the right level, you can pretty much eliminate quantization noise and achieve 18bit effective resolution which I would expect to be 'good enough' to all intents and purposes. Maybe 24 bits sounds slightly better because of the internal architecture of the DAC and differences in how it processes different word lengths, filter coefficients etc. it may be optimized for 24bit operation over 16bits in some non-obvious way.
Regards,
Alex
I listened in to somebody else's test. The person who did the downsampling and upsampling was a BBC sound engineer.
I -have- heard a difference on a different test where 2 different software algorithms were used to downsample from 24/96 to 16/44. The two 16/44 files sounded different to each other.
At Scalford a year or two ago "Pluto" took a 24/96 file of mine and downsampled it to 8 bit. It was noticeably hissy, but when he switched noise sampling on, which on a 96kHz file moves the noise to the inaudible high frequencies, the hiss disappeared. It sounded just like the original. I had to check out the computer to make sure it really was noise-shaped 8-bit not the original file. Very interesting.
Mind you one group of listeners were completely convinced it was a trick and would absolutely not believe that what had been impressing them was 8-bit...
Light Dependant Resistor
06-04-2015, 00:43
The important point is not what you presently listen back on, rather it is that recordings are continued being made
and preserved where full dynamic range was captured.
The listener when and if he or she then becomes aware that you can with the right choice of equipment
appreciate what was contained in that recording with dynamics, will and should greatly appreciate the earlier care
taken with recordings.
As example a generation sadly has been brought up on compressed music, thankfully there is a captive audience
largely unsatisfied, and at the same time many skilled recording engineers and mastering engineers similarly
who totally reject these formats, and who patiently put in the hours properly preserving sound.
" If there is a continuum from one recording to the next, it is the ability to hear the dynamic range of the music, as well as the clarity and focus of the players"
http://www.tapeop.com/interviews/91/jan-erik-kongshaug-bonus/
There are also many audiophiles doing the same thing understanding companding and recording,where
dynamic range that is preserved in the recording, but greater than the source equipment can provide, is
then with the right equipment choice readily achievable.
Once grasped audiophiles are eager to embrace the next round of technology dedicated to these aims.(1)
(1)ftp://ftp.dbxpro.com/pub/pdfs/WhitePapers/Type%20IV.pdf
Hi Clive,
That's also been my experience, since using the Pi. Don't underestimate the (what can only be negative) influence the transport mechanism in the CD player is having on proceedings...
That I buy, which I'd term loosely as 'psuedo hi-res'. So what *is* 'bonafide hi-res', then?
Marco.
But surely the 10p CD mechanism in most computers would have an even larger (negative) effect when ripping?
The mech may introduce mechanical vibration which may have an effect on the overall replay system. But ripping a file is just reading it and storing it elsewhere. The contents of the file will be unaffected no matter how crap the mech, in the same way your Word document would not rip with added spelling mistakes.
Rothchild
06-04-2015, 07:58
The mech may introduce mechanical vibration which may have an effect on the overall replay system. But ripping a file is just reading it and storing it elsewhere. The contents of the file will be unaffected no matter how crap the mech, in the same way your Word document would not rip with added spelling mistakes.
It's actually more complicated than that, ripping a cd is not quite just copying a file (like copying a word document) the data on the cd is not in a neat 'container' (like a .doc document) and there isn't a overall checksum (or something like an MD5 Hash) to verify the data against, so ripping from a more stable platform is likely to yield better results.
There's a really good explanation here: https://thomas.apestaart.org/thomas/trac/wiki/DAD/Rip and I'd throughoughly recommend his excelent morituri ripper: https://thomas.apestaart.org/morituri/trac it's reassuringly slow! ;-)
Light Dependant Resistor
06-04-2015, 08:01
The mech may introduce mechanical vibration which may have an effect on the overall replay system. But ripping a file is just reading it and storing it elsewhere. The contents of the file will be unaffected no matter how crap the mech, in the same way your Word document would not rip with added spelling mistakes.
This end of the world we prefer Libre Office. What will happen with a poor transport is dropouts, followed by
not being able to read the disc at all. You are placing too much faith in error correction. A transport
mechanism sufficient (if it exists at all) for retrieving bit for bit information is always going to
provide higher resolution rather than perpetually error correcting.
Stratmangler
06-04-2015, 08:04
It's actually more complicated than that, ripping a cd is not quite just copying a file (like copying a word document) the data on the cd is not in a neat 'container' (like a .doc document) and there isn't a overall checksum (or something like an MD5 Hash) to verify the data against, so ripping from a more stable platform is likely to yield better results.
There's a really good explanation here: https://thomas.apestaart.org/thomas/trac/wiki/DAD/Rip and I'd throughoughly recommend his excelent morituri ripper: https://thomas.apestaart.org/morituri/trac it's reassuringly slow! ;-)
So it comes full circle back to EAC ;)
I use dbPoweramp.
It has a nice user interface, and it gives identical results to those derived using EAC.
Rothchild
06-04-2015, 08:11
So it comes full circle back to EAC ;)
Pretty much yes.
You could do a home made version (or for disks not in the EAC database) by ripping the disk multiple times and comparing MD5s of each of the rips, if you have the same data the hashes will be the same, the more you have that are the same the safer it is to assume that you have a good rip (or that you're at least consistently reading any errors in the same way).
Stratmangler
06-04-2015, 08:23
I've just trusted EAC to report errors as and when they occur - if a CD rips with no reported errors then there are no errors.
The on the fly data comparison does its job very well.
It's the same with dbPoweramp.
I wrote this year n years ago as a post on PFM. It may help to understand the Error Recovery scheme for Red Book CD....
C1 errors exist or are present on all disks to a certain degree so we have to live with them and correct them… they are small random bit errors. C2 errors are nastier and we don’t want any of them… they are larger burst errors.
All audio CD’s use Cross Interleaved Read-Solomon code (CIRC). It’s the fundamental encoding and error correction scheme used on CD. It consist of three levels, considered good enough for audio data but not good enough for computer data. Audio data that cannot be corrected via C2 correction can be interpolated or guessed. You can’t guess with computer data!
The three error correction levels are C1, C2 and interleaving of the data on the audio CD. IE: the CD is formatted in such away blocks of data are mixed up in a predefined manner to allow for reading of the disk if scratches / imperfections are present. Interleaving cannot correct for errors by itself it only allows you to recover sufficient data blocks for C1 and C2 to work.
C1 and C2 Error Correction Codes (ECC) are applied to the audio data during the mastering process of the CD IE: when interleaving the Audio Data prior to it being placed on the CD. They are used at different points of the de-interleaving process or playback of the Audio data, C1 first then C2.
Now our Audio CD mechanism manufacture may only wish to implement the basic C1 (he has to implement CIRC otherwise he couldn’t read the disk!) and ignore C2 or he can spend a little more money and implement it fully.
However, there is also the physical quality of the mech. to consider. A cheap mech. with poor servos and supply regulation will produce an ‘eye pattern’ of the disk that is not clean (the optics need to track the CD as accurately as possible), this means any error correction scheme has to work harder to correct errors and as the resulting digital stream has to be created from a sinusoidal output from the laser jitter becomes a potential issue. Maybe it’s why you can notice differences in transports…
With a PC based mech. when we perform DAE we are reading the data from the CD in the same way as when we play it except the data this time is directed over whatever computer interface the mech. has (IDE, SCSI etc) and not directed to a DAC. The de-interleaving and error correction schemes are still applicable.
The quality of the mech. is again important and again it really needs to support C2 error correction and have the feature enabled in firmware. The difference now is we have the ability via s/w on the computer to buffer the read data and also to re-read parts of the CD and perform offset reads to recover the audio track. Software can do this because it’s not constrained by playing the CD in real time. However, not all PC based mechs. are equal but the latest DVD multifunction drives seem to do a very good job. EAC is well know and seems to do an incredible job of recovering read errors given a compatible drive.
In theory, playing back the Audio CD from a dedicated player into a DAC compared to playing the same CD extracted to HDD via the same DAC will sound exactly the same…as long as the copy to HDD to bit correct and the dedicated player is capable of recovering in real time the same exact bit data.
BTW the basic CIRC error recovery scheme means the error rate is just one byte in 1,000,000,000 (1 GB) which is pretty good and certainly good enough for Red Book CDs ;)
This end of the world we prefer Libre Office. What will happen with a poor transport is dropouts, followed by
not being able to read the disc at all. You are placing too much faith in error correction. A transport
mechanism sufficient (if it exists at all) for retrieving bit for bit information is always going to
provide higher resolution rather than perpetually error correcting.
I hought we were talking about a cheap mech, not one that is faulty. Corrected errors are corrected they are not going to affect the sound, only uncorrected errors wil do that. That is more likely to be damage to the disc beyond the ability to interpolate. I can see how a cheap mech can add noise to replay but not to a rip.
Marc I read your link don't see anything there to back up what you are saying although I am not saying you are wrong and indeed the data on a disc is not the same as a word file, I can appreciate that.
IME the biggest difference in the quality of CD replay is the power supply in the CD player. Not the mech. Lots of high end players use cheap computer type mechs.
IME the biggest difference in the quality of CD replay is the power supply in the CD player. Not the mech. Lots of high end players use cheap computer type mechs.
Soz, have to disagree with that, as experience strongly contradicts it.
Sure, lots of 'high-end' players today use cheap computer-type mechs, but that's simply a cost-cutting exercise (yes even at those prices) and because there are so few genuinely high-quality CD-only mechs available now. However, look at players from the likes of Esoteric, which use TEAC VRDS-NEO mechanisms, built like brick shithouses for good reason. Check this out:
http://www.esoteric.jp/products/esoteric/p05/indexe.html
Click on the 'Features' button to see an image of the superbly engineered, all metal transport mechanism, and you'll appreciate what it takes to design a proper one!!
One of the things that makes the Sony I use so good at its job (it features the use of something similar to the Esoteric), and other top-notch 'cost no object' vintage CDPs, is the quality of its (diecast metal) transport mechanism, in terms of how well engineered it is, and the fact that it's dedicated CD-only, not a plastic DVD-ROM piece of crap.
Yes, high-quality (large over-specced) power supplies also undoubtedly make a huge difference to performance, but that has no influence whatsoever on the ability of a CDP to not only accurately read the information on discs, but crucially, transfer that information intact (with minimal noise and distortion), and thus with maximum integrity, to the DAC.
It is that last bit which separates the truly great CDPs (modern and vintage) from the merely mediocre, and why up until now, with the RPi/IQ-Audio DAC, I've felt no need to go down the FBA route, as a result of inferior sounding CD replay.
Marco.
Meridian for one say they do not use fancy mechs because they contribute nothing. I suppose they could just be making that up for marketing purposes, but if a CDP is already at the 3 or 4 K level why bother to skimp if they genuinally felt it affected SQ that much?
I take your point about your Sony but there are plenty of variables that will affect the sound of a CD player, I don't think you can necessarily isolate the influence of the mech from the other factors purely by listening.
Trust me, mate, if you research the subject properly, you'll find that there are very good reasons for why transport mechanisms (such as the TEAC VRDS-NEO) contribute massively to accurate CD replay. Essentially, it's about controlling resonance.
Meridian are saying what they're saying simply because it's cheaper and easier to produce their designs using readily available parts, not because something better engineered wouldn't be fundamentally better.
Also, if you're talking about today's genuinely high-end CDPs, such as the Esoteric, then £3-4k will probably just about get you the box that they come in!! ;)
Marco.
I've heard one of those big TEACs they are okay but not anything special.
I think there is more of a pride of ownership thing with the over=engineering more than anything. Like car manufacturers putting lead weights in the doors so they feel 'heavy'.
tubehunter
06-04-2015, 11:08
Yes I remember a Rotel CD player I had, big piece of steel fitted underneath.
I've heard one of those big TEACs they are okay but not anything special.
I think there is more of a pride of ownership thing with the over=engineering more than anything.
For some, that undoubtedly comes into it. However, as I said, my Sony uses a transport mechanism similar to that featured in the Esoteric, and that's a major reason why it sounds so good! :)
One of the main reasons why I targeted the Sony X-777ES (and the R1 transport before it, which matched my DAS-R1 DAC) was because of the sheer quality of its transport mechanism, confirmed by Mark Bartlett (from Audiocom) as being a significant reason why it sounded the way it did.
Thus we both considered it worthy of modifying with the best of modern components, in order to create a truly 'state-of-the-art' CDP, by marrying the best of old and new technology:
http://www.thevintageknob.org/sony-CDP-R1.html
As for weight, the R1 or DAS-R1 doesn't weigh 20kg because of any 'bricks' stuck inside it!
Marco.
A decent pc dvd drive with c2 error detection, used in conjunction with EAC or DbPoweramp CD ripper and Internet checksum comparison holds all of the aces.
If I rip a CD securely and the perfect rip checksum has a confidence level or say 10 or more than I can be very confident that the digital audio extraction process has worked accurately and the bits in my WAV or FLAC file are 100% correct. At that point a file is just a file. A CD drive is reliant on correctly reading the disk first time or successfully correcting C1 or burst errors in a single pass.
The quality of the drive is surely a factor here, but not so much in the ripping case, although some stand a better chance of reading a marginal disk than others. My old plextor was great, Sony was not, hitachi wss good, and lite - on have been good also.
Some will maintain that the post rip sound quality is influenced by the drive used, but I have never bought this and there is no plausible mechanism by which two bit perfect rips can sound different.
StanleyB
06-04-2015, 11:53
Some will maintain that the post rip sound quality is influenced by the drive used, but I have never bought this ..
The worse that the drive performs, the more error correction is needed to recover the errors. A high level of error correction ultimately affects the true accuracy of the audio file. A poor eye pattern from the laser on a poor drive will not be giving you the same level of accuracy as a good eye pattern on a good drive.
Yup... 'Source first' and all that! :)
Aside from that, I'd maintain that if you took two otherwise identical CD players, one fitted with a 'cost no object' uber-engineered CD-only (all-metal) transport mech, and the other, a cheap plastic DVD-ROM mech, the former would sound significantly better to anyone with discerning ears.
Incidentally, the all-metal transport thing is significant, in order to reduce resonance (a very valid phenomenon that is often ignored in these discussions), which "perfect rip checksum" aside, is fundamental in ensuring the integrity of the music signal, and with it, the highest quality of CD replay, whether discs are being ripped and replayed via another digital interface, or simply played directly through a CDP.
Therefore, transport mech quality *does* matter, and is one of the main reasons why when FBA is done well, it sounds so good, as you're removing the problematic and less than ideal physical interface of a laser mechanism reading a disc, which some players address better than others, simply because of the quality of their respective transport mechanisms.
Marco.
Yep, but AlexM is not talking about CD Players, but CD/DVD/BR drives used to rip files. At the end of a EAC rip you can compare the checksum of the ripped track with a online database of others who have ripped the same track. If the checksum matches then I think there is a very close to 100% chance that the data on your computer is identical with the data sent to the CD mastering plant in the first place.
Now how you turn that data into an analogue signal is as open to problems as ever, but in terms of source first, this has the potential to be actually perfect.
Sure, Nick, but how are those CD/DVD/BR drives, used to carry and process the source music signal, designed and constructed?
I maintain that if they're cheap, flimsy plastic things, not designed specifically for CD-only use, they'll introduce a 'sonic signature' detrimental to the quality of the end results obtained, which wouldn't happen if a higher quality, better engineered drive was used, designed solely for CD (not DVD) use.
IMO, it's about more than simply accurate DATA processing. Other things come into it, such as RESONANCE CONTROL (with anything that's spinning discs and carrying embedded musical information).
As an aside, that's one of the reasons why I've never enjoyed any of the CD-based systems put together at Owston, which featured a cheap DVD player as a source. In that respect, it's not just about DACs. Transport quality matters.
Marco.
Yep, but AlexM is not talking about CD Players, but CD/DVD/BR drives used to rip files. At the end of a EAC rip you can compare the checksum of the ripped track with a online database of others who have ripped the same track. If the checksum matches then I think there is a very close to 100% chance that the data on your computer is identical with the data sent to the CD mastering plant in the first place.
Now how you turn that data into an analogue signal is as open to problems as ever, but in terms of source first, this has the potential to be actually perfect.
I have found by using a good quality external drive with external psu i get more reliable results than with a cheaper drive in the pc. usually quicker as well ...It regularly matches checksum without having to do much either. as you say getting it out to your speakers is the tricky bit. thats why i would be interested in a pi if it was set up right and plugnplay
Sure, Nick, but how are those CD/DVD/BR drives, used to carry and process the source music signal, designed and constructed?
I maintain that if they're cheap, flimsy plastic things, not designed specifically for CD-only use, they'll introduce a 'sonic signature' detrimental to the quality of the end results obtained, which wouldn't happen if a higher quality, better engineered drive was used, designed solely for CD (not DVD) use.
IMO, it's about more than simply accurate DATA processing. Other things come into it, such as RESONANCE CONTROL (with anything that's spinning discs and carrying embedded musical information).
As an aside, that's one of the reasons why I've never enjoyed any of the digital systems put together at Owston, which featured a cheap DVD player as their source. In that respect, it's not just about DACs. Transport quality matters.
Marco.
Yes, but Marco, if the checksum of the file on the computer matches the checksum of everyone one else who has ripped the same CD track, the files are identical, exactly the same, no question about that. There is no, nill, none chance that the transport could affect the sound of the files.
Marco,
There's always a danger of "cross-pollinating" the idea of potential losses from the analogue way of doing things to the digital?
Things like resonance control can of course affect the analogue signal output in a CD player, but at the initial reading stage of a CD, as Nick says, if the checksums match its exactly the same digital file that's been put on the CD.
This is one of the reasons (imo) why file based audio is potentially better than cd players because you ELIMINATE the risk of things like resonance.
No need to paint the CD edges green either!
If the sums are right the ripped file is perfect, and its not hard to achieve, not requiring an uber CD mechanism like a Teac or whatever.
Just a friendly fyi fella. :)
Yes, but Marco, if the checksum of the file on the computer matches the checksum of everyone one else who has ripped the same CD track, the files are identical, exactly the same, no question about that. There is no, nill, none chance that the transport could affect the sound of the files.
I completely understand that, but I still contend that if the CD/DVD/BR drives used to rip the files in the first place, were superbly engineered bits of kit, rather than mass-produced plastic pieces of shit, the end results would be even better still.
That's something that someone should try sometime by, say, using something of the ilk of my Sony CDP, as the 'ripping drive', rather than whatever else is used normally, and compare the results.
At the end of the day, I'm afraid that I don't automatically buy the notion that just because the maths says something is so, that it unquestionably *is*, so in that respect, we'll have to agree to disagree :)
Marco.
Marco, they wouldn't. Error correction is error correction one drive will recover the data as well as another to the same accuracy, the construction has nought to do with it. If different drives didn't recover from error in the same manner and to the same accuracy then the various book standards would fundamentally not work.
Where I think the confusion is coming from is a red book CD player has to play in real time and correct bit errors using C1 and C2 error correction on the fly, here I would agree with you the construction could and probably does play a large part in the sound quality. But when you can buffer and re-read the the same data more than once IE you are not constrained by having to read the data in real time the software and drive firmware are of far more importance.
TBH as I touched on before the standard CIRC is more than enough if for Audio, the error rate is just a single byte in 1GB, it really is not an issue.
Ok, I accept all of that, but I'll believe it when I hear it! ['Fraid that's just the way I am] ;)
Marco.
But a better drive may stand a higher chance of getting a clean rip. That is at least my experience - some of my disks are in shit condition and 5 hey might rip perfectly in my lite - on dvd writer, but end up having to rescan lots of frame using a laptop drive. Once they are on the hd, if the checksums match then they are the same file though.
With traditional CD playback, the situation is different and the quality of the drive can indeed influence the sound quality by having a lower read error rate, use less interpolation and error masking in less than ideal conditions with imperfect disks. There are also interactions between the drive and the digital and analogue circuitry to consider. If a servo has to work overtime adjusting the position and focus of the laser pickup to track the disk, that could modulate the power supply rails and introduce noise. A mechanism with superior mechanical qualities and a better control algorithm may well be more effective than a cheapo one. It only gets a single pass over the disk and must read the data as well as possible first time, every time. A CD player is actually reading an analogue signal sourced from an optical laser detector, going through a discriminator that decides whether it is a pit or a land, and therfore a 1 or 0, into a FIFO buffer and then read out from the buffer and decoded and error corrected as frames into another buffer and then sent to a DAC. All of this is a stream of bits read, transformed and processed as the disc rotates at a speed determined closed loop by the bits streaming off the disk. There are lots of opportunities for data to be corrupted or misread (and either corrected or not), and for jitter to be introduced.
The situation is very different in a CD ripping situation as the drive can reread the data as many times as it wants, and actually check that it is correctly read it by calculating a checksum and comparing it. Once the data has been read and stored during the ripping process and verified, there is no possibility of it changing and the impact of the drive mechanism can't change these stored bits. The bits are static binary states with no timing information to upset the determination of their value, be influenced jitter or anything else. This really is a case where bits are bits - the checksum is the guarantor of the data stored in the file, and identical checksums means identical files. Identical files means identical sound through the same dac. We also accept as an article of faith almost that a computer can read, write and manipulate the binary content of these files without introducing corruption. This can happen but the error rate is generally vanishingly low, and it is always possible to verify the integrity of the data at any point. Assuming that we use a programme that reads these files, checks for errors, and passes the data to the dac in a verifiability bit perfect way, then the sound must be identical too.
Sorry if this is teaching anyone to such eggs and I don't mean to patronise anyone, but it just how the technology works... Once you have a good rip with a matched checksum the source is irrelevant.
Excellent post, Alex, and nicely explained! I understand where you're coming from :)
Good to know that better drives are still handy for something, though.... ;)
Marco.
StanleyB
06-04-2015, 20:26
So the general consensus is that two files recorded on different drives that have the same checksum will sound the same if played back on the same equipment.
Now, why does an audio track played back through a CDP and fed to a DAC sound more engaging than the same file ripped from the same CD, but played back through a PC to the same DAC?
awkwardbydesign
06-04-2015, 22:44
I think there is a very close to 100% chance
I love this phrase! I'm not disagreeing with any of the post (I don't know enough), I just like this bit.
It reminds me of "almost a virgin". :D
And BTW, I'm kinda with Marco here. I am simply not comfortable enough with the mathematical and "virtual" world of computer algorithms to trust them Not emotionally, anyway.
There would appear to be a fundamental difference with reading a disc in real time (CD player) and ripping where the computer can take it's time and check. Hard to get to grips with when you're old and senile. Maybe that's just me, though.
Stan,
Not sure about that, and i dont know your products, but one explanation might be that I would guess that the PC is using a USB input and the CD player optical or coax SPDIF?. I have noticed that my own CD player/DAC inputs sound different between the CD drive and an accurately ripped FLAC of the same CD playing through the USB input. I also find that the coax spdif, toslink and USB inputs all sound different - my preference is in the same order.. coax spdif sounds most vivid and snappy, opti dal sounds a little smoother... probably too smooth, and USB sounds boring, mushy and lacking in detail. I have no real idea why this is but I suspect your observations are driven by the interface more than the bits themselves. In the case of my DAC, jitter measured by the Dunn J-test by Paul Miller is lowest for the coax spdif, and highest for USB, so that might be it in my case at least.
I am simply not comfortable enough with the mathematical and "virtual" world of computer algorithms to trust them...
Ditto, although I understand and accept the explanation given. However, when it comes to audio, I'll always be an 'ears first, maths or measurements second', kinda guy! It's just the way my brain is wired :exactly:
Marco.
I love this phrase! I'm not disagreeing with any of the post (I don't know enough), I just like this bit.
It reminds me of "almost a virgin". :D
And BTW, I'm kinda with Marco here. I am simply not comfortable enough with the mathematical and "virtual" world of computer algorithms to trust them Not emotionally, anyway.
There would appear to be a fundamental difference with reading a disc in real time (CD player) and ripping where the computer can take it's time and check. Hard to get to grips with when you're old and senile. Maybe that's just me, though.
Its the mathematician in me showing. It is certain that more than one sequence of source data will produce the same checksum, the different bit lengths of the original music file and the size of the checksum guarantees that. However if the checksum is calculated correctly it would need much more than a few bits in error to produce the same checksum. The point being about the checksum is not that it ensures that two files are totally different, its purpose is to show that two files are very slightly different, and thats what we want here, we can tell if they are different by a lot, one will be music, the other almost certainly noise, but we won't hear a couple of bits being different, the checksum will show us that.
Ditto, although I understand and accept the explanation given. However, when it comes to audio, I'll always be an 'ears first, maths or measurements second', kinda guy! It's just the way my brain is wired :exactly:
Marco.
Understood, but in this case, if two files from the same CD are ripped with different drives, that both give the same checksum as the one in the cloud, are played back from the same PC, with everything in the replay chain being the same, then I would bet my mortgage on them sounding the same. In fact I would define this as being a method to test if we can tell with ears if the same thing played twice (or more) sounded the same.
Now, why does an audio track played back through a CDP and fed to a DAC sound more engaging than the same file ripped from the same CD, but played back through a PC to the same DAC?
One could only imagine that the process of playing the disk on a CDP is producing a different data stream than ripping the CD to disk. Its all about the different error correction available in both cases.
Understood, but in this case, if two files from the same CD are ripped with different drives, that both give the same checksum as the one in the cloud, are played back from the same PC, with everything in the replay chain being the same, then I would bet my mortgage on them sounding the same. In fact I would define this as being a method to test if we can tell with ears if the same thing played twice (or more) sounded the same.
Yup, but you see the thing with me, not being a mathematician, is that I'd only accept it as certain once I'd conducted the listening test and heard the results, not automatically beforehand, simply because 'logic' dictates it! ;)
As they say, 'Vive la différence' :)
Marco.
I'm not convinced by this whole 'error correction' argument.
A CD player either corrects the error so that you get a continuous wave form or it does not. If it doesn't you hear it as a drop out or a spitch type sound. Otherwise it is just a continuous wave form. I dont see how it is possible that a corrected read error will translate to the sound being 'less engaging' Digital just doesn't work that way AFAIK.
Light Dependant Resistor
07-04-2015, 08:01
I'm not convinced by this whole 'error correction' argument.
A CD player either corrects the error so that you get a continuous wave form or it does not. If it doesn't you hear it as a drop out or a spitch type sound. Otherwise it is just a continuous wave form. I dont see how it is possible that a corrected read error will translate to the sound being 'less engaging' Digital just doesn't work that way AFAIK.
In the process of correcting ( it would be lovely if LP similarly corrected ) the player is tasked with
interpolation ( making new information ) and memory. To firstly read the disc there are wobble generators
as wobble can be defined and therefore later compensated. the upside you get to hear the music free of
apparent anomalies, the downside with a poor transport is jitter, a term that describes time increment distortion appearing
as a result of excess error correction. Given CD players are very good at error correction, but a poor
transport has potential to create sound with jitter that distances what was recorded, and subsequently
played back with excess error correction as losing some resolution.
StanleyB
07-04-2015, 08:22
I'm not convinced by this whole 'error correction' argument.
A CD player either corrects the error so that you get a continuous wave form or it does not. If it doesn't you hear it as a drop out or a spitch type sound. Otherwise it is just a continuous wave form. I dont see how it is possible that a corrected read error will translate to the sound being 'less engaging' Digital just doesn't work that way AFAIK.
I don't think that you understand the principle behind the CD format, and the way that the data is extracted and errors corrected.
The format was devised in such a way that it would be able to cope with quite large data losses, but without that being audible or noticeable as an obvious error. The fingerprint test and drilled disc test are examples of this on the Philips Audio CD Test discs.
When there is a loss of audio data a lookup table is used to determine what the missing data might have been. It's best that you do a Google on this since it is quite a bit of involved reading.
Given CD players are very good at error correction, but a poor
transport has potential to create sound with jitter that distances what was recorded, and subsequently
played back with excess error correction as losing some resolution.
Yup, and I can clearly hear the results of that loss of resolution.
As an aside, I think one of the dangers in these types of discussions is applying a 'broad-brush' mentality and treating mathematical data and musical information as one and the same. IMO, they're not, with the latter being rather more 'delicate', and thus its integrity (in an audio sense) much harder to preserve.
Therefore, just because a numerical readout on a piece of electronic equipment says, in that sense, that 'everything is perfect', doesn't automatically mean that it is.
Marco.
I'm intrigued by what people are describing as a CD player having to do it's job in 'real time' or 'on the fly' as opposed to a computer that can take it's time to correct error. What is the difference in time? Why can a CD player not take it's time? Would we notice the time difference?
Light Dependant Resistor
07-04-2015, 08:55
Yup, and I can clearly hear the results of that loss of resolution.
As an aside, I think one of the dangers in these types of discussions is applying a 'broad-brush' mentality and treating mathematical data and musical information as one and the same. IMO, they're not, with the latter being rather more 'delicate', and thus its integrity (in an audio sense) much harder to preserve.
Therefore, just because a numerical readout on a piece of electronic equipment says, in that sense, that 'everything is perfect', doesn't automatically mean that it is.
Marco.
The critical parameter is reducing jitter, if that's done it does get close to perfect.
My article here written in 1997 http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
discusses removing SPDIF when using outboard DAC's and replacing SPDIF which is multiplex digital transmission
with Bitclock data and LRCK separately transferred and clocked
I have also raised on many threads the use of real time and recorded companding which does similar
audio improvement in the analog domain.and digital domain, by improving dynamic range which translates to
more resolution the same as having 24 bit vs 16 bit.
http://theartofsound.net/forum/showthread.php?37814-More-Bits-Dynamic-range
I'm intrigued by what people are describing as a CD player having to do it's job in 'real time' or 'on the fly' as opposed to a computer that can take it's time to correct error. What is the difference in time? Why can a CD player not take it's time? Would we notice the time difference?
Hi Andrew, I believe that the difference referred to here is that in the case of the computer, it is ripping the CD to create a file - this does not have to be done in 'real time' so it can take its time to do as much error correction as it or you want. EAC is a good case in point, sometimes the software takes quite a time to accurately rip the mucic from the CD, and at the end of the process it reports the checksum so you can see how accurate the rip has been. In the case of just playing a CD on a player, this is done in close to 'real time' i.e. you are listening to the music and would obviously notice any pauses....
Morning Graeme. Yes, I appreciate one is ripping the other playing. What I am interested in, is, how long are the delays whilst ripping. Are they intermittent? Are they long enough to be a problem for a CD player if the player used a similar process?
The critical parameter is reducing jitter, if that's done it does get close to perfect.
Close to perfect, but not perfect? That's the problem.
I'm a firm believer that our ears are far more sensitive, for judging matters audio related, than any man-made scientific apparatus, and so that is why when it comes to simply assessing how something sounds, there are no better 'tools' for the job than the God-given organs strapped to the sides of one's head! ;)
Marco.
Morning Graeme. Yes, I appreciate one is ripping the other playing. What I am interested in, is, how long are the delays whilst ripping. Are they intermittent? Are they long enough to be a problem for a CD player if the player used a similar process?
Andrew, yes they can be very noticable. I use EAC pretty much daily, to archive a load of CDs and some tracks simply zip through, whilst others take quite a while. Longer than actual playing time! You would certainly hear the process if you were listening at the same time.
Thanks Graeme. I use Foobar and have never really paid much attention to how long it takes to rip an individual track. I just knew that to rip a whole album it was a fraction of the time needed to play it!
Morning Graeme. Yes, I appreciate one is ripping the other playing. What I am interested in, is, how long are the delays whilst ripping. Are they intermittent? Are they long enough to be a problem for a CD player if the player used a similar process?
Hi Andrew, I think the standard 'architecture' (for want of a better term) for a CD player is ostensibly 'Synchronous', i.e. real time playback, however I see no reason why one couldn't develop an 'Asynchronous' CD player, ripper/buffer/DAC all on the same device, I'm sure someone is already doing it...given the low cost of hardware these days...bit of a niche market though :)
I'm a firm believer that our ears are far more sensitive, for judging matters audio related, than any man-made scientific apparatus, and so that is why when it comes to simply assessing how something sounds, there are no better 'tools' for the job than the God-given organs strapped to the sides of one's head! ;)
Marco.
I've been into file streaming for about 6 years now, and have steadily wound my neck back in from an original IT 'all bits are the same' position...;)
Once your file has been ripped, and reliably transferred to your playback device for transcoding we start to leave the IT domain, and enter a less defined world of physics and electronics...the analogue signal (and the generation thereof) is a very delicate beast IMHO
I can reliably reproduce a bit perfect signal to my MDAC by either COAX, optical, USB, and 'isolated' (via iFi power) USB, but in this instance the last option is much better in SQ terms.
There are many variables here, the architecture of the DAC itself, the delivery method, its preferred sampling rate, inbuilt isolation etc. etc.
I've also experimented with SoX upsampling on the fly to 24/96, which in the case of one DAC (an early Beresford) seemed to improve the sound, is this because the DAC was built for 24/96, and upsampling prior to the DAC removes its workload? :scratch:
Too many questions, but one thing to consider, even subtle changes in DAC firmware can have SQ effects, there's a release of MDAC firmware where the DAC chip array is loaded in reverse sequence (V05 phase shift), and there's a noticeable difference in SQ...and not even JW himself can explain that one...so we are still a little in the dark as to what is really going on...so ears it is then, for the foreseeable future! ;)
16 bit is already high resolution, considering that reel-to-reel has a dynamic range of less than 60db, and vinyl below 70db, 96db is more than enough...
24 bit and beyond is a great thing for production & mastering though, giving a lot more headroom to work with without introducing clipping.
Close to perfect, but not perfect? That's the problem.
I'm a firm believer that our ears are far more sensitive, for judging matters audio related, than any man-made scientific apparatus, and so that is why when it comes to simply assessing how something sounds, there are no better 'tools' for the job than the God-given organs strapped to the sides of one's head! ;)
Marco.
This is true, but then the grey matter doesn't half play some tricks on us sometimes.
StanleyB
07-04-2015, 10:38
Hi Andrew, I think the standard 'architecture' (for want of a better term) for a CD player is ostensibly 'Synchronous', i.e. real time playback, however I see no reason why one couldn't develop an 'Asynchronous' CD player, ripper/buffer/DAC all on the same device, I'm sure someone is already doing it...given the low cost of hardware these days...bit of a niche market though :)
They exist. Quite a few portable CD/Radio players are made along that line these days. One of the reason for that is to prevent skipping etc. So the data is read from the disc at high speed, and stored in a buffer for playback at the correct playback speed.
They exist. Quite a few portable CD/Radio players are made along that line these days. One of the reason for that is to prevent skipping etc. So the data is read from the disc at high speed, and stored in a buffer for playback at the correct playback speed.
...not quite so niche then Stan ;)
Must be a bu66er with 'copy protect' disks!
This is true, but then the grey matter doesn't half play some tricks on us sometimes.
Of course, and I don't doubt or deny that, but one shouldn't automatically assume that's the case in EVERY INSTANCE where what one hears contradicts 'currently accepted wisdom'. *That* is the difference! It's called being a free-thinker, rather than being controlled by scientific doctrine or brainwashed by dogma.
In audio, the answer for me is always to apply healthy scepticism to any phenomenon considered as 'heretical', and if necessary, test for it objectively. I will never just 'automatically believe' anything: every opinion that I form in life (not just in hi-fi) must first satisfy my own judgement criteria.
However, once done, if the results of any objective tests fly in the face of what I can actually hear, and I'm convinced beyond any reasonable doubt that what I'm hearing is right, then in the final analysis, I'll always trust my ears/senses and gut instincts, over supposed 'proof' to the contrary, produced by any man-made measurement apparatus or scientific assessment processes :)
To do otherwise, would be to behave more like a robot than a free-thinking and sentient human being.
Contrary to what some dogmatic 'measurists'/reductionists would like us to believe, not every aspect of audio is a 'done deal'. There is still much more to be learned and 'unexplained phenomena' properly explored!
Marco.
I don't think that you understand the principle behind the CD format, and the way that the data is extracted and errors corrected.
The format was devised in such a way that it would be able to cope with quite large data losses, but without that being audible or noticeable as an obvious error. The fingerprint test and drilled disc test are examples of this on the Philips Audio CD Test discs.
When there is a loss of audio data a lookup table is used to determine what the missing data might have been. It's best that you do a Google on this since it is quite a bit of involved reading.
Not sure I understand your post Stan. You tell me I don't understand then proceed to agree with what I said...
audio improvement in the analog domain.and digital domain, by improving dynamic range which translates to
more resolution the same as having 24 bit vs 16 bit.
http://theartofsound.net/forum/showthread.php?37814-More-Bits-Dynamic-range
More bits does not mean higher resolution it just means more dynamic range.
I think there is something of a language issue here.
Something like MP3 where musical inforamtion has been removed to make a smaller file could be described as low resolution There are varying degrees of this. Red book CD therefore could be described as full resolution since no information has been removed except for frequencies above 22Khz where those actually existed on the recording to begin with which is far from every recording.
Anything with a sampling rate higher than red book CD is not higher resolution - the only difference is that it *may* contain musical information beyond 22Khz i.e outside the range of human hearing and quite possibly beyond the ability of your speakers to reproduce.
Whether this has *any* impact on what you actually hear is still the source of much debate.
This is not 'higher resolution', at least not in any sensible use of the term.
I've kept out of this thread mostly, but wanted to pop my head in for this.
Macca, your assertions to an extent seem to me to hinge on the definition of what "resolution" in audio actually means?
In reality, higher sample rate means just that, more samples of the original analogue waveform when being converted to a digital representation.
One can argue therefore that a digital file with higher samplrate (such as 96k, 192k etc) has less "granularity" when looked at under a scope, so by definition has greater resolution.
A fair point? :)
Conversely, as has been discussed here there's more to it than that, and mastering quality can play a bigger part in the subjective sound quality when it hits your lug'oles.
Not always though...it's never that simple.
I've some 320k MP3s that sound wonderful, and 24/192 and upwards material that sounds flat and lifeless.
The answer is...it depends.
The only exception I've found where I can hear superior quality consistently is playing back DSD files.
More relaxing, open with no loss of detail or connection to the music.
I've a mix of DSD material recorded at the various sample rates and Interestingly (tying in with the argument that hires can sound better) the double rate DSD sound better. I appreciate good digital and good analogue, and this stuff sounds closer to the experience of listening to vinyl.
Speaking of scopes...an informative video here for the measurebators amongst you, just to further muddy the waters. ;)
http://xiph.org/video/vid2.shtml
...just to further muddy the waters. ;)
http://xiph.org/video/vid2.shtml
...Muddy Waters! Tremendous idea, time to get back to listening...;)
Rothchild
07-04-2015, 13:56
In reality, higher sample rate means just that, more samples of the original analogue waveform when being converted to a digital representation.
One can argue therefore that a digital file with higher samplrate (such as 96k, 192k etc) has less "granularity" when looked at under a scope, so by definition has greater resolution.
A fair point? :)
Speaking of scopes...an informative video here for the measurebators amongst you, just to further muddy the waters. ;)
http://xiph.org/video/vid2.shtml
Measurebator, golden ears, flat earther or however you want to demean your fellow music fans, Monty's demo shows that 'granularity' is a misnomer, because there are no steps in the analogue reconstruction of a digital wave. I think in terms of the semantics between Macca and Gazjam it's probably more appropriate to talk about a 'wider' or 'broader' resolution (because more bits means more dynamic range and more samplerate means more frequencies) rather than to think of it in terms of 'finer' resolution.
Anyway, this is about where this argument goes circular I think, because we know we'll not resolve the differing balance we all place between ears, brains, science and logic, but it's been fun (again!) ;-D
Marc, ask around, I've been here a while and I'm never one to demean anyone. :)
Going back to the discussion..I think dynamic range in hifi playback is important and to me one of the more important things in getting closer to the experience of live music.
A bigger factor I'd wager in playback than samplerate/bit depth?
Perhaps this is why some people perceive an improvement with hires material, greater dynamic range?
Just a thought.
And yup, unfortunately our ears are analogue so we all hear differently!
Though conduction in my mastoid ear bone means I can perceive way past 20k....cos I'm special...
Digital playback sounding closer to vinyl? Lord, I hope not :lol: :respect:
The 'fine detail' we hear is only around 20db down on the average level a lot of the time I believe (play a signal hard left or right on vinyl and listen to the other 'unspoken' channel as proof of around 25 to 35db difference), but I do agree that for whatever reason (badly supplied or configured analogue output stages?), earlier cheaper CD players sounded bleached out and weak on lower level transfers on CD. I don't think this happens now...
My experiments with playback gear and speakers especially, have shown this pair of old lug-'oles that we don't really need to worry about 'Digital' as such. It's how the (domestic) playback system, especially the speakers, most of which have lossy and phasey passive crossovers, which are the weakest part still, copes with the extra HF energy and possibilities of extended bass in some recordings - in my opinion.
Rothchild
07-04-2015, 15:01
Earholes are analogue but the actual listening device (the brain) is better thought of as digital.
"This extraordinary signal path [the human ear] encompasses four distinct states of information: acoustic, mechanical (solid), mechanical (liquid), and electric, more specifically electro‑chemical. The very nature of the information also changes: from analogue, it becomes digital. Really! Whereas mechanical information propagation is analogous to the original sound wave, the nervous signal is even more decorrelated from the sound wave than AES‑EBU audio signals can be. Put simply, the ear includes a built‑in analogue‑to‑digital converter. Figure 2, right, summarises these changes of state and nature in the sound information."
Really good in depth article here: https://www.soundonsound.com/sos/mar11/articles/how-the-ear-works.htm
It also mentions that the maximum dynamic range of the ear is around 140dB and we know a 24bit signal can encode 144ishdB of data, so just about enough to capture everything from the quietest thing you can hear to the loudest thing you can stand without damaging yourself (and assuming you had a system both sensitive and large enough to deliver both).
I agree with you that mastering is likely a great contributor, I'm less sure about dynamic range overall as outside classical, jazz, folk and other naturalistic styles most contemporary music (even from the 'golden age') rarely makes use of much more than about 25dB of range (thanks to the use of tape and compressors) and that's the well made stuff, lots of 'loudness wars' stuff has tiny range (often in the sub 10 if not 5dB range).
Marc, ask around, I've been a while and I'm never one to demean anyone. :)
Going back to the discussion..I think dynamic range in hifi playback is important and to me one of the more important things in getting closer to the experience of live music.
A bigger factor I'd wager in playback than samplerate/bit depth?
Perhaps this is why some people perceive an improvement with hires material, greater dynamic range?
Just a thought.
And yup, unfortunately our ears are analogue so we all hear differently!
Though conduction in my mastoid ear bone means I can perceive way past 20k....cos I'm special...
+1 I've some low bitrate stuff with great range which sounds fab and some higher def stuff which just doesn't sound right.
A track I've played at some bake offs to prove the point is Kraftwerk Aerodynamic.
Dynamic range at the point of recording is always more important.
I've kept out of this thread mostly, but wanted to pop my head in for this.
Macca, your assertions to an extent seem to me to hinge on the definition of what "resolution" in audio actually means?
In reality, higher sample rate means just that, more samples of the original analogue waveform when being converted to a digital representation.
One can argue therefore that a digital file with higher samplrate (such as 96k, 192k etc) has less "granularity" when looked at under a scope, so by definition has greater resolution.
A fair point? :)
This really is the crux of it and the main reason for my OP.
There is not less granularity, as you put it, as I said before it is not at all analagous to hi def telly. All you get with the higher sampling rate is higher frequencies - which may or may not improve the SQ (I suspect if they do the improvment is infinitessimal).
Anyway regardless that is the point I was making originally since I see it repeated far too often and wheras on other forums there is always someone who will jump in and correct the error that doesn't really happen on AoS (since we are all so pleasant to each other ;) ) but as a major UK audio foruim we really should not be disseminating mis-information like that.
I am sure that way back we all saw the 'stair step' waveform graph that used to be used to show the superiority of analogue to digital and is now used to show the superiority of 'hi rez' to red book.
All lies and marketing I'm afraid.
StanleyB
07-04-2015, 16:05
Anyway regardless that is the point I was making originally since I see it repeated far too often and wheras on other forums there is always someone who will jump in and correct the error that doesn't really happen on AoS (since we are all so pleasant to each other ;) ) but as a major UK audio foruim we really should not be disseminating mis-information like that.
There are enough points covered that I would strongly disagree with based on my own practical research, but why bother? I would have to go and take pictures of real life waveforms on my scope, crop them, put them on my server, write the text to debunk some of the nonsense, and then what? Best be polite and enjoy the banter I say :). It's far more interesting to let people decide if they can hear the difference, instead of telling them why they should be hearing a difference.
This really is the crux of it and the main reason for my OP.
There is not less granularity, as you put it, as I said before it is not at all analagous to hi def telly. All you get with the higher sampling rate is higher frequencies - which may or may not improve the SQ (I suspect if they do the improvment is infinitessimal).
Anyway regardless that is the point I was making originally since I see it repeated far too often and wheras on other forums there is always someone who will jump in and correct the error that doesn't really happen on AoS (since we are all so pleasant to each other ;) ) but as a major UK audio foruim we really should not be disseminating mis-information like that.
I am sure that way back we all saw the 'stair step' waveform graph that used to be used to show the superiority of analogue to digital and is now used to show the superiority of 'hi rez' to red book.
All lies and marketing I'm afraid.
so you agree with me then?;)
sq225917
07-04-2015, 16:33
The worse that the drive performs, the more error correction is needed to recover the errors. A high level of error correction ultimately affects the true accuracy of the audio file. A poor eye pattern from the laser on a poor drive will not be giving you the same level of accuracy as a good eye pattern on a good drive.
Stanley it's called 'error correction' for a reason. They don't call it error part correction, because what it gives you is a 100% accurate recreation of the original data. If it can't do that then you get skips. There's no half way house.
You're conflating jitter with error correction, a poor eye pattern on a poor drive gives you exactly the same data as a good drive does, it just might not give it to you with equal periodicity. No worries, the dad chip will fix that for you. A drive that was unable to read the data due to gross error would give you exactly the same drop outs.
There are only two types of cd errors, recoverable and none recoverable and the first type are inaudible.
I am not convinced myself - I have heard some very good hires material and attributed it to the carrier format. Down sampling with dither leaves me ambivalent - sometimes I think I can hear a difference, sometimes not. Would I be able to reliably distinguish in a double blind test? I'm not sure.
My Vinyl rips are all done at 24/96 because my ADC supports it and I have the disk space to store unprocessed master recordings. All subsequent processing I do is at 24/96 to minimise the effects of mathematical operations such as normalisation, channel balance etc, but the final output is usually 24/48 or 16/48 if the record isn't in good enough condition that I feel like treating myself to an extra 8 bits worth of noise :). Both sound pretty damn good once I've finished :).
Interestingly old material seems to be mastered with dynamics and less gross use of compression, so it may be more justifiable to use 24 bits than most modern stuff which seems to have a dynamic range of about 10 db. Lol - Even 8 bits would be enough for that! (Only kidding, or am I?)
It also mentions that the maximum dynamic range of the ear is around 140dB
And even that is not as it seems, yes, the difference between the quietest sound the ear can detect and the threshold of pain is maybe 140dB. But it can't hear them both at the same time, the mechanism of the ear includes something similar to a a sliding sensitivity control. if we are listening to a 120dB sound (say a Jet engine at 100 m) we cant hear a 10dB sound (Light leaf rustling)
Rothchild
07-04-2015, 17:39
And even that is not as it seems, yes, the difference between the quietest sound the ear can detect and the threshold of pain is maybe 140dB. But it can't hear them both at the same time, the mechanism of the ear includes something similar to a a sliding sensitivity control. if we are listening to a 120dB sound (say a Jet engine at 100 m) we cant hear a 10dB sound (Light leaf rustling)
Unless you're a true audiophille....:lol:
As I understand it this is in part the theory behind lossy codecs like mp3.
A jet engine is pretty close to white noise, so it's full spectrum, if you had a 140dBSPL 15kHz tone and a quieter 3kHz tone you could probably hear both over a wider volume range (especially as we're more sensitive to 3kHz than we are 15kHz)?
This is the best overall explanation of sampling theory that I've found if anyone is interested..
http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf
Unless you're a true audiophille....:lol:
As I understand it this is in part the theory behind lossy codecs like mp3.
A jet engine is pretty close to white noise, so it's full spectrum, if you had a 140dBSPL 15kHz tone and a quieter 3kHz tone you could probably hear both over a wider volume range (especially as we're more sensitive to 3kHz than we are 15kHz)?
Maybe, but I think people do have a tendency of using a number like 120dB with no real understanding of just what a range that covers.
I think a 140 dB 15Khz tone would probably be an issue for health and safety.
I've been into file streaming for about 6 years now, and have steadily wound my neck back in from an original IT 'all bits are the same' position...;)
Once your file has been ripped, and reliably transferred to your playback device for transcoding we start to leave the IT domain, and enter a less defined world of physics and electronics...the analogue signal (and the generation thereof) is a very delicate beast IMHO
I can reliably reproduce a bit perfect signal to my MDAC by either COAX, optical, USB, and 'isolated' (via iFi power) USB, but in this instance the last option is much better in SQ terms.
There are many variables here, the architecture of the DAC itself, the delivery method, its preferred sampling rate, inbuilt isolation etc. etc.
I've also experimented with SoX upsampling on the fly to 24/96, which in the case of one DAC (an early Beresford) seemed to improve the sound, is this because the DAC was built for 24/96, and upsampling prior to the DAC removes its workload? :scratch:
Too many questions, but one thing to consider, even subtle changes in DAC firmware can have SQ effects, there's a release of MDAC firmware where the DAC chip array is loaded in reverse sequence (V05 phase shift), and there's a noticeable difference in SQ...and not even JW himself can explain that one...so we are still a little in the dark as to what is really going on...so ears it is then, for the foreseeable future! ;)
Interesting stuff, Will. I suspect that we occupy a similar position in this matter.
Marco.
StanleyB
07-04-2015, 18:26
Stanley it's called 'error correction' for a reason. They don't call it error part correction, because what it gives you is a 100% accurate recreation of the original data. If it can't do that then you get skips. There's no half way house.
The CIRC truth table doesn't accurately recreate the original data in every instance. The fingerprint and drilled hole test on the Philips test CD is an example of how data can be recreated and substituted for actual data even though no real information from the missing data exists. I am not in the mood to do a write up on the process, but I had to teach that stuff to repair technicians when I worked for Granada. And before I could do that I spent quite some time being trained myself by Philips engineers from 1983 to 1986. So I know what I am talking about.
https://xiph.org/video/vid1.shtml
https://xiph.org/video/vid2.shtml
These pages don't work in IE, but do in Chrome for me.
Stan, I think you're talking about 'Interpolation,' which is audible on occasion and straightforward Error Correction, which is inaudible.
Rothchild
07-04-2015, 18:48
I think a 140 dB 15Khz tone would probably be an issue for health and safety.
I'm sure it would, if you focussed it in to a beam you could probably burn concrete with it! ;-)
I think the 2 frequencies at once thing is a bit of a red herring. The point I'm aiming at is that; given an input system (microphones etc) that could physically capture it and an output system (amps, speakers) that could generate 0 - 140dBSPL sound, then a 24bit process could encode and replay every 'dynamic' from (something like) a butterfly by your ear or a jet plane at 100m.
Given that in day-to-day life and music works in a range of about 60dB 24bit is a big enough 'resolution' for all audibility.
Day to day life and music is honestly more like a 30db range, but on a sliding scale much of the time I understand. Come out of a LOUD monitoring session (even a long car journey) and it'll take hours to be able to hear quieter things again properly if at all (been there and done it). Hearing the ceremonial cannons go off at our local fort and being within a hundred feet or so is a deafening shock and makes you wince almost.
This really is the crux of it and the main reason for my OP.
There is not less granularity, as you put it, as I said before it is not at all analagous to hi def telly. All you get with the higher sampling rate is higher frequencies - which may or may not improve the SQ (I suspect if they do the improvment is infinitessimal).
Anyway regardless that is the point I was making originally since I see it repeated far too often and wheras on other forums there is always someone who will jump in and correct the error that doesn't really happen on AoS (since we are all so pleasant to each other ;) ) but as a major UK audio foruim we really should not be disseminating mis-information like that.
I am sure that way back we all saw the 'stair step' waveform graph that used to be used to show the superiority of analogue to digital and is now used to show the superiority of 'hi rez' to red book.
All lies and marketing I'm afraid.
Did you see the video I linked to earlier Macca?
http://xiph.org/video/vid2.shtml
No marketing,lies or disseminating mis-information there. :)
*EDIT*
See Daves posted a second one from that site - good information.
Stan, I think you're talking about 'Interpolation,' which is audible on occasion and straightforward Error Correction, which is inaudible.
Stan is correct though, its error correction if the error can be corrected ;) CIRC cannot always correct the error and if it's not possible to interpolate the missing data a Red Book CD player must mute the output...
Here is an interesting snippet on high resolution audio from Bob Stuart of Meridian. The long and short of it is that you already have it with red book audio!. Ok, not in all of the characteristics of an 'ideal' digital system, but in dynamic range anyway.. The full paper can be read here: https://www.meridian-audio.com/meridian-uploads/ara/coding2.pdf. This is certainly an eye-opener for me.. I didn't fully appreciate what could be achieved in a PCM system by use of noise-shaping and dither. When done right, 16 bits can deliver over 110Db of dynamic range.
It is well worth your time to read if you are interested in how digital coding and decoding systems work.
Coding High Quality Digital Audio by J. ROBERT STUART, Meridian audio
Even among audio engineers, there has been considerable misunderstanding about digital audio, about
the sampling theory, and about how PCM works at the functional level. Some of these
misunderstandings persist even today. Top of the list of erroneous assertions are:
i. PCM cannot resolve detail smaller than the LSB (least-significant bit).
ii. PCM cannot resolve time more accurately than the sampling period.
Let’s take (i) first. What is suggested is that because (for example) a 16 bit system defines 64K steps,
that the smallest signal that can be ‘seen’ is 1/64K or about –96dB. Signals dropping off because they are smaller than the smallest step or Least Significant Bit (LSB) is a process we call truncation.
Now you can arrange for a PCM channel to truncate data below the LSB – but no engineer worth his salt has worked like that for over ten years. One of the great discoveries in PCM was that, by adding a small random noise (that we call dither) the truncation effect can disappear. Even more important was the realisation that there is a right sort of random noise to add, and that when the right dither is used [27], the resolution of the digital system becomes infinite. What results from a sensible digitisation or digital operation then is not signal plus a highly-correlated truncation distortion, but the signal and a benign low level hiss. In practical terms, the resolution is limited by our ability to resolve sounds in noise. Just to reinforce this, we have no problem measuring (and hearing) signals of –110dB in a well-designed 16-bit channel.
Regarding temporal accuracy, (ii), if the signal is processed incorrectly (i.e. truncated) it is true that the time resolution is limited to the sampling period divided by the number of digital levels2. However, when the correct dither is used the time resolution also becomes effectively infinite. So, we have established the core point, that wherever audio is digitised (like in an analogue–digital converter) or re-digitised (as in a filter or other DSP process) there is a right way and a wrong way to do it. Neglect of the quantisation effects will lead to highly-audible distortion (as we will see later).
However – and this is perhaps the most fundamental point of all – if the quantisation is performed using the right dither, then the only consequence of the digitisation is effectively the addition of a white, uncorrelated, benign, random noise floor. The level of the noise depends on the number of the bits in the
channel – and that is that!
Staying with practical experience here, I remember Linn commenting in the late 80's that many discs were being pressed with errors outside the Red Book standard, thinking they'd get away with it because so many players would track iffy CD's like top Shure's track vinyl. I remember a CD we bought that got mangled in one of B&O's wafer thin aluminium drawers and the only players that would play it with barely a murmur were machines with the little Philips CDM-9 mechanism in them, the Arcam Alpha 5 and 6 being the best of them all. All the others would hang up or jump over the bad part I remember. The Philips transports that came after (the CDM-12 and so on) refused to track this disc without jumping forward and I'm assuming that the mechanisms' reading ability was held back slightly. This leads me to assume that although 'error correction' is readily allowed, interpolation by 'making up' info to fill gaps is now all but frowned on, so that if you get a severe error, the transport won't play it, either hanging or skipping.
I hope I have it right...
cvision123
07-04-2015, 20:26
I'm on vacation today so I have some time to rant about hi-res music.
IMHO, the music industry is a very special industry where music lovers (especially audiophiles) being taken advantage of is a norm. The marketing people/dealers keep presenting us the "cures" for "symptoms" but never discuss the root/basic foundation of the issue. Thus, they can take us go round and round without the need to really invest in R&D.
Hi-Res music has been there for a long time but... in studio. The idea is to re-sale the material at a higher cost but how??? "Hi-res sounds much better" is the answer :-).
But why??? Can a same master at different sample rate sound much different? Everything has to do with your DAC. The maths says that with higher sample rate, you have less constraint on the filter to perfectly reconstruct the original signal (please forget the b**sh*t staircase figure that the marketing/nasty half-educated "engineers" have been using the demonstrate the differences). One of the most common argument is the quoting of Nyquist theorem. The theorem is totally correct assuming that you have a perfect filter. This is not the case for many DACs out there.
So, IMHO, a lot of people are pushing for hi-res as they really sound good (or better) but you're paying for:
- incompetent DAC designers who didn't manage to design good filters. In their opinions, it's not their faults so you have to buy their "newer version" where they can do better with less constraints to achieve a better result.
- companies that try to sell you the same music all over again with new "packaging". This is a common strategy: reissue on bluray-spec CD, ...
As long as people don't know the basic foundation, they can state anything they like and take our money. They don't need to be competent in the work, just need to provide a so so solution and then improve and... They are well protected by the crowd of dishonest people called themselves "reviewers with golden ears".
But I don't really care. I love listening to music so I just keep listening... :-)
Light Dependant Resistor
07-04-2015, 21:12
More bits does not mean higher resolution it just means more dynamic range..
(1) "resolution - When an analog signal is digitized, it is represented by a finite number of discrete voltage levels. The resolution is the number of discrete levels that are used to represent the signal. To more accurately replicate the analog signal, the resolution must be increased. Resolution is usually defined in bits. Using converters with higher resolutions will reduce the quantization error"
(1) http://www.maximintegrated.com/en/de...rsion.cfm#ENOB
Light Dependant Resistor
07-04-2015, 21:19
Marc, ask around, I've been here a while and I'm never one to demean anyone. :)
Going back to the discussion..I think dynamic range in hifi playback is important and to me one of the more important things in getting closer to the experience of live music.
A bigger factor I'd wager in playback than samplerate/bit depth?
Perhaps this is why some people perceive an improvement with hires material, greater dynamic range?
Just a thought.
And yup, unfortunately our ears are analogue so we all hear differently!
Though conduction in my mastoid ear bone means I can perceive way past 20k....cos I'm special...
Yes dynamic range ability ftp://ftp.dbxpro.com/pub/pdfs/WhitePapers/Type%20IV.pdf
and jitter free recordings and playback
Light Dependant Resistor
07-04-2015, 21:25
Close to perfect, but not perfect? That's the problem.
I'm a firm believer that our ears are far more sensitive, for judging matters audio related, than any man-made scientific apparatus, and so that is why when it comes to simply assessing how something sounds, there are no better 'tools' for the job than the God-given organs strapped to the sides of one's head! ;)
Marco.
Perfection I would define as having the musicians in the room, which could get awkward with the likes of Black Sabbath or AC/ DC
hence the wisdom of high fidelity equipment being slightly less than perfect.
Light Dependant Resistor
07-04-2015, 22:53
For perfection other than Ozzy being nearly there with you, the high fidelity audiophile buying
public has to start understanding industry compromises, and ideally start knowing how to use a soldering iron
and the ins and outs of electronics, as most manufacturers are very slow to take up good ideas
other than a dedicated few.
Lets start with 98% of available equipment and just 2% then available as balanced.
http://en.wikipedia.org/wiki/Balanced_audio
99% of recordings are done on balanced equipment, inferring if you want to hear what
was recorded you need similar resolution capability, but there is more to it than just
using balanced equipment.
Abandon SPDIF which invites jitter - time increment distortion. Nothing available
other than AES/EBU and separate word clock on some equipment- which is an improvement.
Ideally all digital transfer should extract and process balanced Data, balanced Bitclock
and balanced LRCK ( Left Right Clock ) onward to a DAC with similar balanced
capability.
Abandon low quality MP3, and replace it with Flac or Ogg if file space is a consideration
Start using the best op amps available in equipment which are presently the LM4562 dual and LME49710 single
Start using companding, a DBX 150x built for companding tape, happens to work
very well with CD when its op amps are changed for the above types adding approx 4 bits
to playback, and is better still with a midway recording device. inputs and outputs
in the 150x can be balanced or unbalanced, if using inputs as Unbal always make the
plug a mono phone jack, and outputs a stereo phone jack - refer to the 150x manual
The audiophile community need to lobby DBX or Dolby to build a
dedicated real time CD compander to be available, rather than using remaining
second hand 1980's 150x or 180a or 155's and having to change their op amps
The reason for companding equipment is that CD player manufacturers are unlikely to
be able to provide higher resolution players. The last dare I say it, bit of resolution
is the job of a compander. My preference would be DBX as they have always
concentrated on full frequency resolution.
http://www.stereophile.com/news/11303/
Once you are there, Ozzy may indeed start appearing in your room
with all of the consequences that brings.
This guy knows his stuff.
This is the best overall explanation of sampling theory that I've found if anyone is interested..
http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf
Yes this is worth a read but for those who cant be bothered here are the salient points:
While this article offers a general explanation of sampling, the author's motivation is to help dispel the wide spread misconceptions regarding sampling of audio at a rate of 192KHz. This misconception, propagated by industry salesmen, is built on false premises, contrary to the fundamental theories that made digital communication and processing possible.
The notion that more is better may appeal to one's common sense. Presented with analogies such as more pixels for better video, or faster clock to speed computers, one may be misled to believe that faster sampling will yield better resolution and detail. The analogies are wrong.
The great value offered by Nyquist's theorem is the realization that we have ALL the
information with 100% of the detail, and no distortions, without the burden of "extra fast" sampling.
There are reports of better sound with higher sampling rates. No doubt, the folks that like the "sound of a 192KHz" converter hear something. Clearly it has nothing to do with more bandwidth: the instruments make next to no 96KHz sound, the microphones don't respond to it, the speakers don't produce it, and the ear can not hear it. Moreover, we hear some reports about "some of that special quality captured by that 192KHz is retained when down sampling to 44.1KHz. Such reports neglect the fact that a 44.1KHz sampled material can not contain above 22.05KHz of audio.
“If you hear it, there is something there” is an artistic statement. If you like it and want to use it, go ahead. But whatever you hear is not due to energy above audio. All is contained within the "lower band". It could be certain type of distortions that sound good to you. Can it be that someone made a real good 192KHz device, and even after down sampling it has fewer distortions? Not likely
Does Dan mention dynamic range at all in that article?
awkwardbydesign
08-04-2015, 09:09
(1) "resolution - When an analog signal is digitized, it is represented by a finite number of discrete voltage levels. The resolution is the number of discrete levels that are used to represent the signal. To more accurately replicate the analog signal, the resolution must be increased. Resolution is usually defined in bits. Using converters with higher resolutions will reduce the quantization error"
(1) http://www.maximintegrated.com/en/de...rsion.cfm#ENOB
Andrew Scheps in his address explained it nicely for me. If you take the sampling rate as the vertical lines on a graph. and the bit rate as the horizontals, the intersections of those lines will be closer together. So the chances of hitting the right the right voltage when sampling are increased as the bit rate increases. And then the wave you reproduce will be closer to the original wave. Attack and decay will be closer too. A higher sample frequency will push the filter up beyond audibility, meaning either a gentler filter can be used, or a sharp one will produce less (or no) pre-ringing in the audio band. Or both. Having no audio content up there doesn't matter, it's the reduction in filter artifacts that makes hi-res sound better. Makes sense to me in my general ignorance.
awkwardbydesign
08-04-2015, 09:16
99% of recordings are done on balanced equipment, inferring if you want to hear what
was recorded you need similar resolution capability, but there is more to it than just
using balanced equipment.
Isn't the use of balaced equipment in studios more to do with noise reduction, especially with multiple pieces of equipment? Rather than higher resolution? Of course less noise will increase the perceived resolution (in the same way replay equipment with a lower noise floor will; which is why I like my TVC and LDR pre), but that's slightly different.
Light Dependant Resistor
08-04-2015, 11:18
Isn't the use of balaced equipment in studios more to do with noise reduction, especially with multiple pieces of equipment? Rather than higher resolution? Of course less noise will increase the perceived resolution (in the same way replay equipment with a lower noise floor will; which is why I like my TVC and LDR pre), but that's slightly different.
Yes noise reduction, and higher resolution both,
explained here: http://www.perreaux.com/blog/index.cfm/2012/2/27/Balanced-vs-Unbalanced-Audio
Audio Advent
08-04-2015, 12:30
Perfection I would define as having the musicians in the room, which could get awkward with the likes of Black Sabbath or AC/ DC
hence the wisdom of high fidelity equipment being slightly less than perfect.
Or John Cage's Symphony for 12 Radios (with the original 60's america broadcasts) or a Kate Bush's "Big Sky" for the jet fly past or or or - live music is pretty limited in scope let's be honest!
Or Cage's 4'33"
No, hang on...think I'm missing the point. :D
Audio Advent
08-04-2015, 12:45
Yes this is worth a read but for those who cant be bothered here are the salient points:
While this article offers a general explanation of sampling, the author's motivation is to help dispel the wide spread misconceptions regarding sampling of audio at a rate of 192KHz. This misconception, propagated by industry salesmen, is built on false premises, contrary to the fundamental theories that made digital communication and processing possible.
The notion that more is better may appeal to one's common sense. Presented with analogies such as more pixels for better video, or faster clock to speed computers, one may be misled to believe that faster sampling will yield better resolution and detail. The analogies are wrong.
The great value offered by Nyquist's theorem is the realization that we have ALL the
information with 100% of the detail, and no distortions, without the burden of "extra fast" sampling.
There are reports of better sound with higher sampling rates. No doubt, the folks that like the "sound of a 192KHz" converter hear something. Clearly it has nothing to do with more bandwidth: the instruments make next to no 96KHz sound, the microphones don't respond to it, the speakers don't produce it, and the ear can not hear it. Moreover, we hear some reports about "some of that special quality captured by that 192KHz is retained when down sampling to 44.1KHz. Such reports neglect the fact that a 44.1KHz sampled material can not contain above 22.05KHz of audio.
“If you hear it, there is something there” is an artistic statement. If you like it and want to use it, go ahead. But whatever you hear is not due to energy above audio. All is contained within the "lower band". It could be certain type of distortions that sound good to you. Can it be that someone made a real good 192KHz device, and even after down sampling it has fewer distortions? Not likely
Only problem with Dan Lavry's "propagated by industry salesmen" part above is that he has created a bit of a cult following with his views on high-res audio and sells some very good but over-priced compared to the competition 24/96 limited DACs and ADC hardware. His top of the line pieces are £5000 for just 2-channels.
Many DACs and ADCs are better at one resolution than others depending on their architecture. By selling this idea of anything above 24/96 being completely pointless and perhaps even detrimental, he is at the same time marketing his niche products which focus so narrowly on producing top-notch 24/96.
That's the problem with dismissing commercial products which raise the spec bar - a lie of the new spec being better is indistinguishable from real passion and belief in your higher-spec sonic breakthrough product or even your deliberately lower spec product.
Audio Advent
08-04-2015, 12:49
Or Cage's 4'33"
No, hang on...think I'm missing the point. :D
No no, you're correct! Especially on isolating headphones, you capture and can hear the intended event. Live you get a different shuffling of audiences and coughs etc each and every time and those 26dB sounds drown out Cage's music. Actually, thinking about it, you really need a special system to properly reproduce Cage's 4'33" - the noise floor on most systems is just far too high. I mean, I don't consider someone an audiophile or serious about their music unless their system can properly reproduce -140dB. And no, turning it off doesn't count.
Rothchild
08-04-2015, 12:51
Or Cage's 4'33"
No, hang on...think I'm missing the point. :D
You'd definitely need 24 192 to resolve the inky blackness adequately.....
Audio Advent
08-04-2015, 13:13
I'm on vacation today so I have some time to rant about hi-res music.
IMHO, the music industry is a very special industry where music lovers (especially audiophiles) being taken advantage of is a norm. The marketing people/dealers keep presenting us the "cures" for "symptoms" but never discuss the root/basic foundation of the issue. Thus, they can take us go round and round without the need to really invest in R&D.
Hi-Res music has been there for a long time but... in studio. The idea is to re-sale the material at a higher cost but how??? "Hi-res sounds much better" is the answer :-).
But why??? Can a same master at different sample rate sound much different? Everything has to do with your DAC. The maths says that with higher sample rate, you have less constraint on the filter to perfectly reconstruct the original signal (please forget the b**sh*t staircase figure that the marketing/nasty half-educated "engineers" have been using the demonstrate the differences). One of the most common argument is the quoting of Nyquist theorem. The theorem is totally correct assuming that you have a perfect filter. This is not the case for many DACs out there.
So, IMHO, a lot of people are pushing for hi-res as they really sound good (or better) but you're paying for:
- incompetent DAC designers who didn't manage to design good filters. In their opinions, it's not their faults so you have to buy their "newer version" where they can do better with less constraints to achieve a better result.
- companies that try to sell you the same music all over again with new "packaging". This is a common strategy: reissue on bluray-spec CD, ...
As long as people don't know the basic foundation, they can state anything they like and take our money. They don't need to be competent in the work, just need to provide a so so solution and then improve and... They are well protected by the crowd of dishonest people called themselves "reviewers with golden ears".
But I don't really care. I love listening to music so I just keep listening... :-)
What people pay for is their own choice! People don't need all sorts of things, especially designer gear, but people spend so much money on brand names whilst the item e.g. a handbag, cost the same as any other handbag to make.
So, why single out the audio industry for trying to extend "designer lifestyle" to digital downloads? We're in a transitory digital peroid - at some point the high-res files will become standard and therefore there will be no price difference to what people pay now. The same can't be said for designer handbags - they will forever be there to rip people off.
All that has really happened is that the original studio master formats can be distributed directly to the listener so there is now no need for some down-grading of the original master to shoe-horn it onto an obsolete format like CD. This original studio master format idea currently has some cachet at the moment because people are still stuck in CD resolutions or too used to crap MP3s, but that will soon loose it's specialist tag as it becomes more the norm. Also I would like to download files directly from the artists themselves so I wouldn't expect them to know how to use noise shapped dither to expand the dynamic range and all that jazz - they can just record in high-res and that dynamic range is automatically there for them in the format. How they then use their analogue and recording gear is up to them.
Sam I think you are overlooking two things;
1) The mass market is more than happy with MP3 and a good portion of it does not want to pay for music at all.
2) 'Hi Res' is currently being used by the music industry to charge a premium price. Therefore they will not be looking to reduce that price anytime soon.
Sam I think you are overlooking two things;
1) The mass market is more than happy with MP3 and a good portion of it does not want to pay for music at all.
2) 'Hi Res' is currently being used by the music industry to charge a premium price. Therefore they will not be looking to reduce that price anytime soon.
I would agree. My daughter and all her friends, of which she has many, all play mp3 through their phones, often blue toothed, or through a dock, if not using earbuds. She is not especially young either, being married with 2 kids and has a near professional job. Younger people mostly dont even want loads of cds lying around and dont want complex or expensive systems and files. They are happy with lower quality mp3 etc, and why not. I guess if I had grew up listening to it I would be too.. She laughed when she saw the size of a 12" vinyl.....being born when vinyl was on its way out and cds just beginning; that year I think.
Rothchild
08-04-2015, 16:21
I She laughed when she saw the size of a 12" vinyl.....being born when vinyl was on its way out and cds just beginning; that year I think.
Heh, OT but my mum was telling me that the lad next door to her (must be 12 or 13 years old) found a box of vinyl at the bus stop and phoned his mum to tell her that he'd found a box of the biggest CDs he'd ever seen! :lol:
There's an interesting (also OT) riff in what you say though Grant, do we think that the 'de-materialisation' of our music sources is necessarily linked to our satisfaction with lower grade sources (and replay devices) or is it just that SQ is the easiest compromise to make against the huge convenience and immediacy of portable music? Stereotypically most of what 'the kids' are listening today was jumbled up in Fruity Loops from a pirated sample disk anyway, does it really warrant a great deal of effort to reproduce?
Heh, OT but my mum was telling me that the lad next door to her (must be 12 or 13 years old) found a box of vinyl at the bus stop and phoned his mum to tell her that he'd found a box of the biggest CDs he'd ever seen! :lol:
There's an interesting (also OT) riff in what you say though Grant, do we think that the 'de-materialisation' of our music sources is necessarily linked to our satisfaction with lower grade sources (and replay devices) or is it just that SQ is the easiest compromise to make against the huge convenience and immediacy of portable music? Stereotypically most of what 'the kids' are listening today was jumbled up in Fruity Loops from a pirated sample disk anyway, does it really warrant a great deal of effort to reproduce?
In many ways, very much the point. The stuff most young un listen to is easily reproduced in mp3 format and sounds good, if you like that sort of stuff. I think the main point is that they like the convenience and the minimalistic way in which its done...IE, its not really important to them, unlike getting their hair done which they are happy to drop 50/100 quid a go for....and nails, and face packs. and new cloths for every do they go to :lol:
Audio Advent
08-04-2015, 17:27
Heh, OT but my mum was telling me that the lad next door to her (must be 12 or 13 years old) found a box of vinyl at the bus stop and phoned his mum to tell her that he'd found a box of the biggest CDs he'd ever seen! :lol:
There's an interesting (also OT) riff in what you say though Grant, do we think that the 'de-materialisation' of our music sources is necessarily linked to our satisfaction with lower grade sources (and replay devices) or is it just that SQ is the easiest compromise to make against the huge convenience and immediacy of portable music? Stereotypically most of what 'the kids' are listening today was jumbled up in Fruity Loops from a pirated sample disk anyway, does it really warrant a great deal of effort to reproduce?
Fruity Loops - blast from the past. It's called "FL Studio" now don't you know! Yes "Studio" is in the name so it must be pretty much a professional product :lol:
Audio Advent
08-04-2015, 18:16
Sam I think you are overlooking two things;
1) The mass market is more than happy with MP3 and a good portion of it does not want to pay for music at all.
2) 'Hi Res' is currently being used by the music industry to charge a premium price. Therefore they will not be looking to reduce that price anytime soon.
Maybe, maybe not - I see that as more a snapshot of the status today and not a sign of the future. I'm more optimistic and see all the reasons MP3s became popular in the first place changing to make it redundant: storage capacity, download bandwidth etc. Basically the masses will use whatever is readily available and convenient - that is why poor quality MP3 is popular today. As tech progresses, so convenience and availability will equally be applicable to high-res and MP3 seen as old fashioned and backward.
The music industry charged a premium for CD originally - and now look at it, can't give secondhand ones away.
Actually CD prices have gone up in the past year o so. I know 'cause I still buy them. My prediction a couple of years back was that by now they would be paying people to take them away. That just hasn't happened.
With regards to MP3 if I look back at my teenage years we had cassette tape. And most of them were copies made of your mate's records on their parent's Ferguson music centre and played back on your parent's Ferguson music centre. Not exactly hi-fi but it did for us back then and I suppose nothig much has changed except the format. To move up to hi-fi you have to get serious about replay and that will only ever be a small percentage of music lovers.
The inflation-adjusted price of a CD bought in 1984 at 10.99 (was that the going rate?) would be £33.17. No wonder the music industry has fallen on hard times!
The Black Adder
09-04-2015, 09:31
I couldn't believe how much I've spent on music in the past... now, every cd is worthless. I have a very cynical attitude when it comes to the music industry as in it's main monster publishers.
Independent music rules.
walpurgis
09-04-2015, 09:47
I'm quite happy buying CDs and very happy with the outstanding sound I get from them. The value of my CD collection does not matter to me, as they were bought to listen to, not as an investment.
I have enough good sounding players not to have to worry about units failing. Not that I've ever had one pack up.
I won't bow to fashion as far as equipment goes and I don't use a PC in any way for music (yet).
I predict, that to an extent, CD will go the way of vinyl and once enough people have binned their collections and cheapo CD players, values will pick up. In fact prices for good players and DACs are on the rise.
I couldn't believe how much I've spent on music in the past... now, every cd is worthless. I have a very cynical attitude when it comes to the music industry as in it's main monster publishers.
Independent music rules.
its true that cds on the second hand market do not command much of a price but lets face it most vinyl doesnt either. The main reason you buy music is to listen and at average price of about 8 quid a cd is the equiv of 2 pints or a bottle of good wine. I know a few folk who got rid of their cds and regretted it a few years later just the way folk regret dumping their vinyl in the dark ages:eyebrows:
I love having a large physical music collection, on vinyl and CD, for all the reasons I've mentioned many times here before. I will *never* be without it, simply because I've now embraced FBA :)
Like I said before, the most passionate of music lovers have audio systems that contain more than one source of accessing music! ;)
Marco [who accesses his favourite choons daily, via FBA, vinyl, CD, tape and radio].
Audio Advent
10-04-2015, 23:36
I couldn't believe how much I've spent on music in the past... now, every cd is worthless. I have a very cynical attitude when it comes to the music industry as in it's main monster publishers.
Independent music rules.
I'm not sure what you mean about every CD being worthless or at least what the problem is with a CD loosing it's financial value?
If I go to the cinema and watch a film just once I have absolutely nothing to take away with me, nothing to watch again and again and yet I've come away having thought it was good money spent (assuming I liked the film) and it influences my life.
If I buy a book and enjoy it, I rarely read it more than once (that's how I am with books) but again I'd consider it money well spent. Books quickly loose their financial value too.
If I buy a CD I can listen to it again and again and again forever and a day (if I take care of it). That is surely money better spent than either of the other two, yet you seem miffed that it's "worthless"?
It's also a licence to listen and play that music as you wish privately, legally speaking.. If you get rid of that CD then you have no legal right to keep any digital copies of it and effectively can't listen to that music again when and where you want without forking out for another licence with download licences being much more restrictive (streaming services get around this but you have to either suffer advertising or pay for a the service/licence too). So it has value there too.
If you mean you only now view any purchase of anything as a financial "investment" ... then I think humanity has lost yet another soul and you can't be saved.
Ali Tait
11-04-2015, 06:56
It's LOSE man, LOSE!
LOOSE is the opposite of tight!
Ah, feel so much better now. :D
It's LOSE man, LOSE!
LOOSE is the opposite of tight!
Is that what you told her last night? :D :eyebrows:
Marco.
sq225917
11-04-2015, 08:00
No need to be so uptight, you need to losen up.
Ali Tait
11-04-2015, 08:06
:lol:
No need to be so uptight, you need to losen up.
Bring out the KY jelly again?
Marco.
Audio Advent
14-04-2015, 17:35
Massage oil
Audio Advent
14-04-2015, 17:40
It's LOSE man, LOSE!
LOOSE is the opposite of tight!
Ah, feel so much better now. :D
Exactly! I was being creative with language.. makes perfect sense if you open your mind. Loose is what I meant, honest. Perhaps "loosen" would have been better in hindsight..
Massage oil
sort of; ...in a way :lol:
Audio Advent
14-04-2015, 17:51
sort of; ...in a way :lol:
I mean massage oil will help loosen him up. The KY? Well... bringing it out might have the opposite effect!
The worse that the drive performs, the more error correction is needed to recover the errors. A high level of error correction ultimately affects the true accuracy of the audio file. A poor eye pattern from the laser on a poor drive will not be giving you the same level of accuracy as a good eye pattern on a good drive.
It is a very interesting thread and discussion, a I agree with above, but CD players have also a buffer, large enough to correct the error before it gets to the conversion, am I right?
Light Dependant Resistor
17-04-2015, 08:48
Hi
You are referring to a parity bit, http://www.digitalprosound.com/Features/2000/Sept/RecCD4.htm
error correction, buffering , memory , all add what is called jitter- time increment skew distortion
http://enjoythemusic.com/magazine/manufacture/1104/index.html
and my article that proposed to assist curing one of the causes of jitter
http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
Cheers / Chris
Hi
You are referring to a parity bit, http://www.digitalprosound.com/Features/2000/Sept/RecCD4.htm
error correction, buffering , memory , all add what is called jitter- time increment skew distortion
http://enjoythemusic.com/magazine/manufacture/1104/index.html
and my article that proposed to assist curing one of the causes of jitter
http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
Cheers / Chris
I agree, that jitter as it is a difference in time (fluctuation) between particular frames received, but nowadays most of the hardware is able to deal with that (it's not late 80s no more) on levels that are mostly inaudible. So, we can say that optical or buffering may cause some jitter, but if we are able to hear it I think that's a different story. Also now, most of the DAC chips are locking data stream on preamble, meaning that for real data the jitter is even less.
Light Dependant Resistor
17-04-2015, 10:00
I agree, that jitter as it is a difference in time (fluctuation) between particular frames received, but nowadays most of the hardware is able to deal with that (it's not late 80s no more) on levels that are mostly inaudible. So, we can say that optical or buffering may cause some jitter, but if we are able to hear it I think that's a different story. Also now, most of the DAC chips are locking data stream on preamble, meaning that for real data the jitter is even less.
When you have heard a ultra analog 20400A unconstrained by having SPDIF feeding it, you will quickly then know the sound signature that jitter causes
Assuming all is well because you connect a cable from a CD player to an outboard DAC ignores you are locking on to a multiplexed form of data with
inherant jitter, No matter how you clock SPDIF it always has jitter.
When you have heard a ultra analog 20400A unconstrained by having SPDIF feeding it, you will quickly then know the sound signature that jitter causes
Assuming all is well because you connect a cable from a CD player to an outboard DAC ignores you are locking on to a multiplexed form of data with
inherant jitter, No matter how you clock SPDIF it always has jitter.
Well, let's take as an example fairly low priced CD player Marantz CD5004 measured by Stereophile here: http://www.stereophile.com/content/marantz-cd5004-cd-player-marantz-cd5004-cd-player-measurements
Talking about jitter is less than -125dB according to their measurements and to cite them up: "I haven't given a numeric figure for the player's jitter level, as it was below the Miller Analyzer's resolution limit."
Audio Advent
17-04-2015, 12:49
It is a very interesting thread and discussion, a I agree with above, but CD players have also a buffer, large enough to correct the error before it gets to the conversion, am I right?
That would surely require the cd to be read multiple times by the CD mechanism? I'm sure some do do that though... but I'd guess the minority. I would have thought a buffer can only help correct some errors but not those experienced by the single-pass read process in the first place, an estimate of what might be there would still have to be made.
That would surely require the cd to be read multiple times by the CD mechanism? I'm sure some do do that though... but I'd guess the minority. I would have thought a buffer can only help correct some errors but not those experienced by the single-pass read process in the first place, an estimate of what might be there would still have to be made.
Interesting, I've heard somewhere that allowing to read the CD player the CD twice may increase the sound quality…
Audio Advent
17-04-2015, 12:51
When you have heard a ultra analog 20400A unconstrained by having SPDIF feeding it, you will quickly then know the sound signature that jitter causes
Assuming all is well because you connect a cable from a CD player to an outboard DAC ignores you are locking on to a multiplexed form of data with
inherant jitter, No matter how you clock SPDIF it always has jitter.
That's a DAC chip from the 90s isn't it, used in high-end players of the time? Is it a favourite of yours, where did you hear it without s/pdif and how did that incarnation sound?
Audio Advent
17-04-2015, 13:01
Interesting, I've heard somewhere that allowing to read the CD player the CD twice may increase the sound quality…
When you're able to do whilst still playing the CD in real time, that I guess you get much closer to ripping the CD and playing it as a file (just that that file is only a partial file and never saved).
Just did a quick Google search. Some Meridian CD players of the past decade do this (actually 12 years ago at least), e.g. the G08 and 808. Here is a passage from their .pdf datasheet: https://www.meridian-audio.com/meridian-uploads/data/808_ds_scr.pdf
Meridian’s 808 is the latest in a series of optical disc players that use a specially-selected ROM drive for reading. The ROM drive allows multiple passes to be made, ensuring that the correct data are recovered from the disc and improving Compact Disc’s error-correction a hundredfold. It also allows complete buffering of the recovered data.
To ensure the lowest possible jitter, 808 incorporates three buffers, two of which are used as FIFOs. By the time the data is passed to the DACs or the digital output, the jitter is incredibly low – in fact 808 has the lowest jitter we have ever measured on a CD player: around 90 picoseconds, with the jitter spectrum held below 0.1Hz.
The last paragraph implies that jitter is a complete seperate problem to error correction (although I'd guess it's more complex than that and one effects the other in someway..).
WOStantonCS100
17-04-2015, 21:56
The Myth of 'High Resolution' (digital, I'm assuming) audio??? Not in my house, with my gear. The differences are all too apparent. Anywho... :whistle:
Light Dependant Resistor
17-04-2015, 23:40
Well, let's take as an example fairly low priced CD player Marantz CD5004 measured by Stereophile here: http://www.stereophile.com/content/marantz-cd5004-cd-player-marantz-cd5004-cd-player-measurements
Talking about jitter is less than -125dB according to their measurements and to cite them up: "I haven't given a numeric figure for the player's jitter level, as it was below the Miller Analyzer's resolution limit."
But the Marantz CD5004 is a player not an outboard DAC ! . It would be a different picture with jitter if internal to the player SPDIF was used to connect to
the oversampling chip. The 5004 uses the proper method of Data LRCK and Bitclock as separate data streams, As my article suggests similarly this should be extended to
connect ouboard DACs as 3 digital links, and to dispense with SPDIF - where the market is dictated by audiophile demands.
http://www.enjoythemusic.com/magazine/viewpoint/0401/deficienciesofspdif.htm
SPDIF can certainly be still offered for those who care less about quality.
Here is another viewpoint reaching the same conclusion that SPDIF is flawed.
www.researchgate.net/ profile/ Malcolm_Hawksford/ publication/ 267420574_Is_The_AESEBU__SPDIF_Digital_Audio_Inter face_Flawed_/ links/ 54636ca10cf2c0c6aec4b9e1.pdf
But this is (maybe unintentionally) conflating jitter and error correction as if once a signal has the evil jitter, its doomed. Any digital transmission medium will suffer from jitter. All you have to do is engineer the receiver to remove it. Its perfectly possible to transmit error free 24 bit audio via s/pdif such that it produces a bit perfect copy of the original. Ethernet, USB, spdiff, i2s single ended or via LVDS signalling are all perfectly capable of doing whats needed as long as the implementation is not broken.
The problem is always the implementation, just picking a alternative solution doesn't guarantee any better results but itself.
Light Dependant Resistor
19-04-2015, 07:49
Unfortunately its not that easy.
www.stereophile.com/content/jitter-digital-interface-page-1
Unfortunately its not that easy.
www.stereophile.com/content/jitter-digital-interface-page-1
Page not found
If you mean this article from 1993 http://www.stereophile.com/reference/1093jitter/index.html then I am not sure what your point is?
I didn't say reducing jitter wasn't important, but that there is no reason why it can't be fixed, and if thats done then it wont matter what transmission medium is used.
Its how engineering works. You find a problem, then you find a fix, then you move onto the next problem. Jitter is not some special secret that DAC makers have been ignoring.
Light Dependant Resistor
19-04-2015, 10:21
Page not found
apologies here it is: http://www.stereophile.com/reference/1093jitter/index.html
and Robert Harley's article http://www.stereophile.com/content/jitter-game-page-4
where he states:
"Because a CD player has no S/PDIF interface between the transport and processor, one would expect it to have low jitter at the DAC"
Leading to obvious meaning if you can avoid SPDIF, then digital integrity is far better. Accepting SPDIF
as a consumer add on is fine same as you accept your washing machine to do your clothes, but it should not be used
or taken seriously, if audiophile demands are to be met.
TBH, as that article mentions, the bigger Jitter problem for us is not s/pdiff, its jitter at the original A to D stage of the recording. There is absolutely nothing we can do now to correct that now.
apologies here it is: http://www.stereophile.com/reference/1093jitter/index.html
and Robert Harley's article http://www.stereophile.com/content/jitter-game-page-4
where he states:
"Because a CD player has no S/PDIF interface between the transport and processor, one would expect it to have low jitter at the DAC"
Leading to obvious meaning if you can avoid SPDIF, then digital integrity is far better. Accepting SPDIF
as a consumer add on is fine same as you accept your washing machine to do your clothes, but it should not be used
or taken seriously, if audiophile demands are to be met.
Another interface between A to D conversion is I2C, I think it's quite similar.
Light Dependant Resistor
19-04-2015, 11:27
TBH, as that article mentions, the bigger Jitter problem for us is not s/pdiff, its jitter at the original A to D stage of the recording. There is absolutely nothing we can do now to correct that now.
Very true, but there are a few good ways of appreciating labels who go the extra yard with quality like ECM
and not abandoning all hope of retrieving what is there as best we can. Methods including using single box CD players
and improving dynamic range which is resolution otherwise untapped and unheard I have covered ways of doing that already.
Very true, but there are a few good ways of appreciating labels who go the extra yard with quality like ECM
and not abandoning all hope of retrieving what is there as best we can. Methods including using single box CD players
and improving dynamic range which is resolution otherwise untapped and unheard I have covered ways of doing that already.
Ditto
Another interface between A to D conversion is I2C, I think it's quite similar.
i2c is a control interface, you would not use that for data at audio rates.
http://en.wikipedia.org/wiki/I%C2%B2C
Dac chips often have a a i2c interface, but its for things like setting up the chip and muting and volume control. Most audio DAC chips will use i2s or some variant of it, normally 4 single ended lines, bit clk, lr clk, data, and a mclk which is normally a multiple of the bit clock. The raspberry Pi is a good example of i2s, the processor has a built in i2s transmitter that is sent over the interface port and can be used by add on boards like Marco has discovered. The Pi in that case is a perfect example of what I am talking about, the music source is often remote and data is sent over ethernet or wifi, jitter and transmission errors are irrelevant to the final result as the pi can error correct, ignore jitter on the external interface and produce a high quality feed to the DAC. With a bit of work a s/pdiff receiver and a cpld you could do just the same, interface s/pdiff, convert to parallel in the cpld, then buffer in the pi, driving the i2s to the dac in the process removing any jitter on the s/pdiff interface.
Would agree than Spdif RCA isn't the ideal connection for digital audio, but Spdif BNC is another matter?..
In my own personal experience, a properly done 75ohm impedance matched BNC at both Dac and transport (transformer coupled BNC) is the best digital hookup I've heard in my system.
Looking a bit up market from my setup, the best digital guys in the game (probably) dCS, offer BNC connections as well as USB in their very high end DACs.
The Absolute Sound reviewed their Debussy Dac and BNC was the preferred digital connection saying the USB got close, but not quite. This was on an $11K Dac, so you'd reckon their USB implementation would be top notch?
http://www.theabsolutesound.com/articles/dcs-debussy-dac-tas-209/
This illustrates I think that you can't sensibly generalise about this stuff, which is what I feel some folk are maybe doing?
Sure, USB has the advantage of being able to pass higher sample rates and bit depths, but as this thread proves with talk of mastering quality etc, it's a moot point whether thats actually an advantage?
I can hear subjective improvements of a 24/96 track over 16/44 OF THE SAME RECORDING AND MASTERING, and really that all I need to know.
The recent Led Zep remasters are available in 16/44 and 24/96 and you can hear the difference.
Higher than 24/96...is good for recording engineers making what we listen to, but for home listening I wouldn't bother personally.
Spdif bad, USB good...Like all things audio it ain't ever quite that simple.
As Nick said, it's an engineering problem and it's all in the implementation.
Or as Banarama put it...
"it ain't what you do it's the way that you do it" :)
awkwardbydesign
19-04-2015, 13:27
apologies here it is: http://www.stereophile.com/reference/1093jitter/index.html
I just read it. I had a headache by the end of page 3! I should stick to sawing and hammering. :(
amigarobbo
09-05-2015, 10:23
I just watched all 1hr 19min of this. https://www.youtube.com/watch?v=SXbH-yzGNfg From an engineer who cares, and there is a LOT more going on than just the mastering.
Recommended.
Hang on, isn't he the man responsible for Metallica's Death Magnetic, apparently one of the worst sounding albums eva?
Would agree than Spdif RCA isn't the ideal connection for digital audio, but Spdif BNC is another matter?..
In my own personal experience, a properly done 75ohm impedance matched BNC at both Dac and transport (transformer coupled BNC) is the best digital hookup I've heard in my system.
Looking a bit up market from my setup, the best digital guys in the game (probably) dCS, offer BNC connections as well as USB in their very high end DACs.
The Absolute Sound reviewed their Debussy Dac and BNC was the preferred digital connection saying the USB got close, but not quite. This was on an $11K Dac, so you'd reckon their USB implementation would be top notch?
http://www.theabsolutesound.com/articles/dcs-debussy-dac-tas-209/
This illustrates I think that you can't sensibly generalise about this stuff, which is what I feel some folk are maybe doing?
Sure, USB has the advantage of being able to pass higher sample rates and bit depths, but as this thread proves with talk of mastering quality etc, it's a moot point whether thats actually an advantage?
I can hear subjective improvements of a 24/96 track over 16/44 OF THE SAME RECORDING AND MASTERING, and really that all I need to know.
The recent Led Zep remasters are available in 16/44 and 24/96 and you can hear the difference.
Higher than 24/96...is good for recording engineers making what we listen to, but for home listening I wouldn't bother personally.
Spdif bad, USB good...Like all things audio it ain't ever quite that simple.
As Nick said, it's an engineering problem and it's all in the implementation.
Or as Banarama put it...
"it ain't what you do it's the way that you do it" :)
Yes I use BNC too, my DAC only has 2x BNC inputs and I connect one of those to my BNC Halide Bridge for the USB to SPDIF. From a technical perspective BNC is the way to go, there is no RCA connection possible that maintains 75R impedance due to plug and socket geometry.
In any case my current USB to SPDIF has no glare or digital nasties & changed my long-standing view that CD transports sounded better. Looking at the measurements of the Halide Bridge it's easy to see why it sounds so good.
Stratmangler
09-05-2015, 11:03
Hang on, isn't he the man responsible for Metallica's Death Magnetic, apparently one of the worst sounding albums eva?
And your point is?
It doesn't matter how good the recording and production is because the final production mastering has final say as to how good something sounds.
And if the brief coming through from the client (in this case the record company) demands that it's mastered so that everything sound louder than everything else then that's the way it's done.
Metallica are just as culpable for the shit mastering as the record company - they've been around for long enough to know that the only way to get a job done properly is to take control of it themselves.
They didn't, and left it to the record company.
Engineers in the recording industry try to produce the best work they can, but if the client doesn't give a flying f**k about the quality then they're in a quandry - do I make the very best job of this that I can and possibly not get paid or do I do what the client demands and put food on the table?
I know which one I'd pick, and it's not the suffering for my art route ;)
The Halide Bridge is a great piece of kit, looked into one before I got my current USB Dac.
Bit skint at the time so went for the Vlink 192, but the Halide Bridge was better.
Hang on, isn't he the man responsible for Metallica's Death Magnetic, apparently one of the worst sounding albums eva?
An album that goes up to 11. :rolleyes:
amigarobbo
09-05-2015, 11:33
And your point is?
It doesn't matter how good the recording and production is because the final production mastering has final say as to how good something sounds.
And if the brief coming through from the client (in this case the record company) demands that it's mastered so that everything sound louder than everything else then that's the way it's done.
Metallica are just as culpable for the shit mastering as the record company - they've been around for long enough to know that the only way to get a job done properly is to take control of it themselves.
They didn't, and left it to the record company.
At about 1hr 5 mins he does mention that he may have played them a/b test samples and the group themselves preferred the 'loud' version, I'm guessing that he's signed some NDA or something, given how vague he is about it.
Anyway,
Have we all seen this?
http://xiph.org/~xiphmont/demo/neil-young.html
Audio Advent
09-05-2015, 13:47
And your point is?
It doesn't matter how good the recording and production is because the final production mastering has final say as to how good something sounds.
And if the brief coming through from the client (in this case the record company) demands that it's mastered so that everything sound louder than everything else then that's the way it's done.
Metallica are just as culpable for the shit mastering as the record company - they've been around for long enough to know that the only way to get a job done properly is to take control of it themselves.
They didn't, and left it to the record company.
Engineers in the recording industry try to produce the best work they can, but if the client doesn't give a flying f**k about the quality then they're in a quandry - do I make the very best job of this that I can and possibly not get paid or do I do what the client demands and put food on the table?
I know which one I'd pick, and it's not the suffering for my art route ;)
Not even that. It's a matter of taste! Your idea of quality is just that, yours.
There are other videos I stumbled across also interviewing the same guy and he said he has absolutely no regrets over the album.
Why? Because Metallica thought it sounded superb! That was the sound they wanted, compressed, punchy and loud and they loved it and the producer mentioned also liked the sound given the material. He too was happy with the result and in no way did anyone think it had been f**ked up.
It's kind of like saying that Edvard Munch's painting "The Scream" is unrealistic.. Let's be honest, it doesn't look like a real person at all and the colours are just stupidly wrong! He ruined it - god knows who the producer was on that painting.
http://upload.wikimedia.org/wikipedia/commons/thumb/f/f4/The_Scream.jpg/220px-The_Scream.jpg
"It doesn't resemble live sound at all! "
Audio Advent
09-05-2015, 13:48
haha - I'm not sure if the line's a comment on the painting or if the guy's screaming about the Metallica album :lol:
Audio Advent
09-05-2015, 13:54
Anyway,
Have we all seen this?
http://xiph.org/~xiphmont/demo/neil-young.html
Gah..! Who hasn't? That article by some opinioned person is wheeled out all the time even when everything in it is argued against very well by others with opposing views. Not de-bunked 'cos it's not necessarily completely wrong, just that it's an opinion being backed by selected evidence rather than an opinon formed AFTER the production of evidence.
Unfortunately 95% of audiophile internet arguments in any direction are formed first by belief and then "facts" selected to back up those views - I can back this up too once I've found the right evidence to present to you.
Not even that. It's a matter of taste! Your idea of quality is just that, yours.
There are other videos I stumbled across also interviewing the same guy and he said he has absolutely no regrets over the album.
Why? Because Metallica thought it sounded superb! That was the sound they wanted, compressed, punchy and loud and they loved it and the producer mentioned also liked the sound given the material. He too was happy with the result and in no way did anyone think it had been f**ked up.
It's kind of like saying that Edvard Munch's painting "The Scream" is unrealistic.. Let's be honest, it doesn't look like a real person at all and the colours are just stupidly wrong! He ruined it - god knows who the producer was on that painting.
http://upload.wikimedia.org/wikipedia/commons/thumb/f/f4/The_Scream.jpg/220px-The_Scream.jpg
"It doesn't resemble live sound at all! "
This is a very good point. There is a difference between trying to capture a live sound as accurately as possible and a studio album. With a studio album you are getting what the artist intended, whether you like it or not.
Audio Advent
09-05-2015, 14:30
whether you like it or not.
or even the fans!
Rothchild
09-05-2015, 17:37
Gah..! Who hasn't? That article by some opinioned person is wheeled out all the time even when everything in it is argued against very well by others with opposing views. Not de-bunked 'cos it's not necessarily completely wrong, just that it's an opinion being backed by selected evidence rather than an opinon formed AFTER the production of evidence.
Unfortunately 95% of audiophile internet arguments in any direction are formed first by belief and then "facts" selected to back up those views - I can back this up too once I've found the right evidence to present to you.
I'm puzzled, could you point out the bits that are just opinion and/or direct me to the valid and recognised opposing arguments? I think painting Chris Montgomery as an ill-informed internet opinion poster is a tad off the mark, given that he created the OGG Vorbis codec and maintained FLAC (and isn't selling anything) I think his 'opinion' is probably one worth paying attention to?
Audio Advent
09-05-2015, 23:20
I'm puzzled, could you point out the bits that are just opinion and/or direct me to the valid and recognised opposing arguments? I think painting Chris Montgomery as an ill-informed internet opinion poster is a tad off the mark, given that he created the OGG Vorbis codec and maintained FLAC (and isn't selling anything) I think his 'opinion' is probably one worth paying attention to?
I haven't painted him as that - that's your reading of what I posted. You read what I posted and interpretted it in your own way, imparting perhaps your expectations and previous experience in doing so. I can see why you might have interpretted it in that way trying to put myself in another's shoes, another coming from a particular position, but I can assure you it wasn't meant in the way you think.
Similarly someone like Chris Montgomery will interpret info from about the place and form an opinion of which he is similarly convinced is true based on his own expectations and experience. He may or may not be wrong but his opinion is no more relevent just because, as a programmer, he's turned his hand to some open source audio codecs! I'd guess he turned his hand to that from a love of music and a love or interest in mathematical algorithms.
Audio codecs are about programming algorithms and open source codecs by their nature of not trying to sell anything and therefore no money for research, will take lossy compression techniques from elsewhere and re-purpose them. In other words there is no evidence that the OGG codec re-examined psycho-acoustics for themselves via scientific experimentation (it has been done already for MP3 and so why would you?) nor did any new scientific research into audio at all. It is knowledge taken from elsewhere and his skill is presumably in the programming side - lots of programmers also come in from non-audio based worlds and create great codecs but it doesn't mean they necessarily have any specialist knowledge of high-res audio and whether it sounds better or why it might.
In my opinion - from experience of the odd coder I've met and the general gist of the mindset of a coder to become just that - programmers are often more of the logical disposition and extrapolate from rules and memes they've learned to be true. It comes with the territory of the building blocks of each language. Therefore, again only my opinion, you are more likely to be complete sceptical of anything which diverges from that accepted logic and teachings.
I've read many good arguments opposing some of what is said in his article by people who have researched audio for themselves as part of their job and people who work with high-res everyday on an experiencial level. No I won't spend hours trying to pull up threads on forums I've read over the years - better if you find things yourself I guess (quite often when people say "show me ..." they are not actually going to read the links anyway, more interested in just arguing their curent position).
The position of only trusting people who have nothing to sell is kind of baffling to me too. That's almost like saying that the bloke down the pub is more knowledgeable than someone doing research on the matter for a company, simply because he has nothing to sell... Also that's ignoring that people also sell themselves and sell their opinion and sell their egos. I mean, it's one thing to have an opinion but why would you write a whole set of webpages for everyone to reference if you didn't have a reputation to sell on some level? That you think his opinion is worth paying attention to shows that he has something to sell, otherwise I'm sure you would never have heard of him and he wouldn't have a wikipedia page with his face on it..
In other words, he may or may not be correct but there's no reason I can see to take his word and opinion over many others with differing views and being a programmer certainly doesn't add to that.
Audio Advent
09-05-2015, 23:23
I think my personal problem with it is his argument style - he indeed brings up a lot of audio facts but then concludes an opinion that this means x y and z in terms of what is important to the human being listening to audio.
I mean, impulse response isn't mentioned once for example. Likely because it didn't come up in his study of previous, non-open source codecs. Neither is the effect of non-perfect filters on the audio which can be pushed far far into inaudibility when using high resolutions.
Besides, his take is all about playback. There's a lot more going on in the A/D conversion side which higher resolutions can improve (and why would you not then listen to those native resolutions on playback to remove an unnecesary step of sample rate conversion?).
EDIT: Hey look at his own video here http://www.xiph.org/video/vid1.shtml at the 11 minute mark! He even goes over the fact that the low pass filters for 96 and 192KHz are easier to build and gives that very fact as a reason for the use of high-sample rate audio..
This is also the very SAME reason that some in the professional audio industry use as an explaination of the supposed superior sound of high resolutions.
And yet when making his own arguements AGAINST anything over 16/48 he conveniently ignores it all in order to make his point - I guess it brings too much doubt to his logical conclusion that you can't hear the difference, opens it up to a subjective take on the sound of the filters.. Just goes to show that web pages like that do have ulterior motives even if that is just to come across as authoritative to get people agreeing with you (and why would you write such a public page if the intention wasn't to get people to agree with you, agree with what you believe to be correct?).
Posted that video 10 pages ago!
If you can't hear the improvement of higher rates than 16/44 then your deaf and probably a liar...with a crap system.
Too much? :D
I'll get me coat..
Gaz assuming we are 100% confident that it is the same master, would you describe the difference as blatently obvious, or very subtle. Or somewhere in between?
To these ears Martin I'd say somewhere in between, it's not blatantly obvious but it's that thing where you miss that "something" when it's not there.
Shows itself to me as a less strident more relaxed sound yet conversely with a more open mid and better detail at the top end.
A more relaxed natural presentation...again quite subtle but noticeable when it's not there.
Just my £0.02 worth, my ears my system etc. :)
Posted that video 10 pages ago!
If you can't hear the improvement of higher rates than 16/44 then your deaf and probably a liar...with a crap system.
Too much? :D
I'll get me coat..
well Im deaf and have a crap system!!!!...
...no, im lying:lol:
Audio Advent
10-05-2015, 11:43
Reading reviews of studio kit over the years with the reviewers with no ulteria motive in saying that 24/96 sounded better than 24/48 on the same peice of equipment even if you could cynically say they were given a bribe to push the product, when people make record their own stuff as part of the review the higher resolutions give them a better sense of "air" and clarity and naturalness - same words you get going higher and more so with people who are enamoured with their own DSD recordings.
When you're trusting someone else's recordings and mastering processes etc etc, it's very hard to know what you're actually listening to to make easy judgement.
Of course they will be comparing the A/D process directly too at the same time and engineers in the recording industry (selling their products) talk of it making the difference more in the A/D stage not the D/A stage, yet it's the D/A stage where the high-res tech and files are sold to audiophiles regardless of the music's recording resolution.
Audio Advent
10-05-2015, 11:49
Posted that video 10 pages ago!
If you can't hear the improvement of higher rates than 16/44 then your deaf and probably a liar...with a crap system.
Too much? :D
I'll get me coat..
Ah! Different video ... caught you out.
Powered by vBulletin® Version 4.2.3 Copyright © 2024 vBulletin Solutions, Inc. All rights reserved.