[James Tanner]

James, do you have any idea at what point, as we descend away from 0dBFS in 16bit and 24bit systems we would hit 3% THD+N.
Scotty

Hi Scotty

No sorry I do not.

james

[Mag]

I knoze whats I'ze hearing!

If it's not the bits it must be the sampling rate, 48k sample rate captures more instrument harmonics than 44.1.

[Russell Dawkins]

The finger is not appreciated.
Good for you for thinking you are able to hear extra harmonics with 48 over 44k.

48kHz sampling rate allows just more than one (1) semitone higher limit than 44.1 kHz. There are 12 semitones in one octave. Our hearing range is just shy of 9 octaves as adults; 10 as a child. How many extra harmonics do you think that allows??
48k sampling exists not because it's intended to be better than 44.1k, but to make it incompatible with 44.1k - to separate the pros from the consumers. 44.1k was the pro standard; 48k was the standard chosen when consumers were first able to record digitally - with DAT recorders and minidiscs and the like, so they could not make CDs from their recordings without going to a professional who could sample rate convert to 44.1 for CD release.
Now anybody and his dog can do this conversion - and plebes can argue over the audibility of harmonics with 44k sampling compared to 48k.
Here's my finger! 

On the other hand, 24 bit word lengths are useful for headroom in recording and the benefit of >176.4k sampling is audible.

[James Tanner]

Heres the link for further discussion - very informative 


http://www.audioaficionado.org/redirect-to/?redirect=http%3A%2F%2Fwww.head-fi.org%2Ft%2F415361%2F24bit-vs-16bit-the-myth-exploded

james

[James Tanner]

Hi Folks,

The above link is quite long and involved so I have distilled it down to some interesting info - Black text is G and Red is questions from others.

Gegorio

After more than 2 years, I thought I'd pop back here briefly to see how it was going and to thank those of you for the kind emails you sent. I started this thread to provide information to help you, the consumers, to avoid the huge amount of snake oil present in the audio and HiFi industry. I'm glad I was of help to many of you. I'm not so glad that the industry continues using deception as a marketing tool. If 24bit is a waste of time (and space) for playback and 96kHz sampling rate is already higher than the optimum rate for Analogue to Digital Converters, the latest ADCs boast 32bit and 384kHz! We are getting further and further away from the optimal design, therefore resulting in less accurate conversion, while the marketing is trying to convince you of exactly the opposite. This is a real shame, instead of developing better quality filters, clocking mechanisms and analogue components in 96kHz converters to give us better quality audio, the industry has decided it's cheaper and easier to develop poorer quality converters (at 192kHz and 384kHz) and take us for fools by marketing them as better. It's deceptive and unethical and makes me pretty mad as a consumer myself.
 
The paper I linked to previously (http://www.lavryengineering.com/documents/Sampling_Theory.pdf) is as true today as it was the day it was written. Computing power has obviously dramatically increased since 2004 but there is a finite speed limit to how quickly a capacitor can be charged and amps can settle, no amount of computing power is going to change the laws of physics. I was pleased to see at least one other manufacturer come out recently and tell the truth about sampling rates. My estimation of them as a company has been greatly enhanced: Benchmark official statement 96kHz vs.192kHz
 
I might pop back here occasionally (time permitting), to answer questions where I'm able and to help those others here like me trying to combat the marketing fraud being perpetrated. Accurate information is the best weapon in my opinion. If the highly offensive personal attacks from shills and those entrenched in their ignorance starts up again though I'll just call it a day again.
 
Regards,
 
Gegorio

One last point: 32bit integer should in theory provide 192dB dynamic range, have a look at the S/N ratio output of any so called 32bit DAC, what do the specs say; 110dB, 120dB maybe even 130dB? Hang on though, even at 130dB what's happened to the other 62dB? That represents over 10bits of data, just gone, what's happened to it? It's just noise! Even the very finest DACs out there cannot re-produce more than 22bits worth of dynamic range and this is not going to change until someone can come along and change the laws of physics. Literally, we have just about reached the absolute limit of the laws of physics and incidentally, exceeded what a human being can safely listen to by a factor of roughly 1000 (at 22bits)! Virtually no recording ever released exceeds a dynamic range of about 60-70dB.

There's a DAC somewhere which supports up to 384kHz sampling rate. Slap a bigger number on it and audiophiles will flip out, even if there's no recordings that would actually use those extra bits.

It's worse than that I'm afraid, it not just a case of recordings being able to use those bits or extra bandwidth theoretically provided by 384kHz sampling rates. I only discussed sampling rates a little much earlier in this thread but we never really considered the whole thing together, sampling rates and bit depth. Electronic engineering and signal processing these days quite often comes up against the laws of physics and there is an axiom; the larger the bandwidth the lower the accuracy. In professional Analog to Digital Converters (ADCs) it's not uncommon to oversample by 256 or even 512 times, IE. A sample frequency of over 22mHz giving an audio bandwidth of over 11mHz but at this large bandwidth we are only capable of 6 or so bits of accuracy. It is possible to sample at over a gHz but the accuracy would obviously be proportionately lower. This isn't a problem which is going to go away. It's not a question of throwing more computing power at the problem, it's a limitation of the laws of physics. A great deal of ADC design is about trade-offs!

So, what is the trade off or optimal bandwidth to accuracy ratio for music (as opposed to other signal processing, say in the telecom industry)? If we say that 24bits is the accuracy which we want (for recording) then the best evidence available at this point in time suggests a sampling rate of roughly 60-70kHz. Unfortunately, the industry has decided not to implement say a 65kHz sampling frequency so the best trade off would be 88.2kHz. 88.2kHz gives plenty of space for smooth and error free filtering (required by the Nyquist Theorem) and only a marginal and relatively inconsequential loss of accuracy. But this is not true of 192kHz and even less true of 384kHz. There is a price to pay for larger bandwidths and the larger the bandwidth the larger the price. The price is paid in non-linearity, in other words 192kHz isn't just about the futility of trying to record and reproduce irrelevant audio frequencies, it actually reduces the quality of those frequencies which you can hear! Yes, you are understanding correctly, 24/192 is actually poorer quality audio than say 24/96! For evidence of this please read the white paper published by Dan Lavry and confirmed last year by Benchmark (links on the previous page). The Lavry paper is quite technical but the Benchmark confirmation is written in a way everyone can understand. Lavry's paper was published in 2004, so it's not as if it's a recent discovery. That hasn't stopped the DAC manufacturers and some in the music industry using marketing to convince you that 24/192 is better than 24/94 and then charging you a premium for it! So, lower quality now seems to cost you a higher price, please don't get sucked in!!!!!

I want to make it clear, I am not saying that you won't hear a difference between a 24/96 and a 24/192 recording, it's possible you might, as there is more noise and distortion present in a 24/192 recording than there is in a 24/96 recording. It's obvious that many reviewers seem to prefer 192 sample rates, I can't say if this is because they are following the advertising revenue, believe 192 should sound better and therefore hear an improvement or whether they honestly prefer the sound of more noise/distortion. The bottom line though hasn't changed from when I started this thread. 16bit exceeds the resolution of playback systems (and your ears) and 70kHz sampling rate exceeds the bandwidth required to eliminate all artefacts. Unfortunately 16/70 format doesn't exist, neither does 16/88.2 or 16/96, so the best trade off that the industry allows us is 24/88.2 or 24/96. This represents the highest quality audio format which is currently available.

Gegorio


Interesting to see this thread is still alive and well 4 years after I started it! Some of the questions and responses are also interesting, as is the fact that some posters can't seem to get their head's around the concepts in my OP. Generally this appears to stem from the fact that they have an entrenched concept which they are unwilling or unable to question, namely that more resolution = more detail. One can't entirely blame them for this entrenched position as it has been created and reinforced by the marketing of all those companies who sell higher bit depth files or equipment. The very term "Hi Rez" itself is a purely marketing term rather than an accurate description because if we are talking about resolution in terms of detail (of the sound which comes out of your DAC), 24bit has no more resolution than 16bit or indeed than 1bit! I saw one truly bizarre set of posts in this thread from someone who thought it was simple common sense both in theory and in practice that 24bit had more detail and sounds better than 16bit and that anyone suggesting otherwise must be essentially insane. This poster then held up SACD as the highest audio quality but of course SACD uses just 1bit and therefore, according to his common sense argument, SACD should sound many times inferior to the humble 16bit CD.
 
Some of the points raised in this thread (and others on Head-Fi) could do in my opinion with a little more explanation, so here goes:
 
Resolution
 
To help those still struggling with the concept that resolution does not equal detail, try turning your logic around and think of it this way: Exchange the word resolution with the words "less error". More bits = more accuracy and of course more accuracy = less error, hence higher resolution (more bits) therefore means less error. The next leap in understanding this question of resolution is understanding the fact that error in digital audio manifests as noise when it is converted back into an analogue signal. More resolution therefore results in less noise compared to lower resolution, the detail (fidelity) is always there and always the same (at any bit depth) but the more bits you use the less noise will accompany that detail. By the time we get to 16bit, the noise (due to error) is already many times below the noise which is already present in the recording due to other unrelated music production factors. Using noise shaped dither further reduces the audibility of that noise to well below the noise produced by even the very finest DAC, amp and headphone or speaker system. In other words you cannot hear this dither noise! So, if you already cannot hear the errors (noise) introduced by 16bit what is the point of increasing the resolution (to 24bit) and reducing the noise even further. In other words, what do you possibly imagine could be gained by taking inaudible noise and making it quieter? If this is true, why does 24bit even exist, what point is there? This brings me on to recording:
 
Recording
 
When recording in 16bit, we want to create the cleanest recording possible and that means making sure that the noise caused by our 16bit errors are below the noise of our microphones, mic pre-amps and the noise floor of our recording studio. This means setting our microphone pre-amps so that the loudest parts of our recording are somewhere near the maximum (0dBFS) level of our system. The problem with this is that we don't know what the loudest part of our recording is actually going to be until after we've recorded it. Sure, we can (and do) do a sound check but even the very finest artists cannot reproduce a performance exactly, so a sound check is only a very rough guide. Performers can create peaks 12dB higher or even more during recording compared to the sound check. So our problem is how to record loud enough to ensure our digital noise (errors) does not encroach on our recording but not too loud so that the loudest parts of the performance exceeds 0dBFS. In some recording situations this can be a quite small window of opportunity to hit. Recording in 24bit though ensures that our digital error noise is so far below the noise of our recording equipment and environment that we don't need to think about it. With 24bit we can afford to set our mic pre-amps so that the loudest part of our recorded signal is nowhere near 0dB, even -20dB is fine in 24bit, so it doesn't really matter how loud the performer performs, we're not going to ruin that brilliant and unrepeatable performance because we've hit the limit (0dBFS). We don't use 24bit for recording because it provides more detail, better quality or higher fidelity, a 24bit recording will have the same fidelity as a 16bit recording made within that window of opportunity, it just makes that window of opportunity far easier to hit! In other words, the advantages of 24bit for recording are all about workflow and nothing to do with quality or fidelity. Of course, when we mix/master and create the distribution mix, we already have the recorded material, already know where it is going to peak and already know what that peak value is, so we don't need that spare dynamic range, can go right up to -0.1dBFS without ruining our mix and are going to be so far away from the 16bit digital noise (errors) that it's guaranteed to be many times below the threshold of audibility. This brings up the question of mixing:
 
Mixing
 
When we mix music (or a film or TV program) we could be dealing with anything from a few channels of sound up to over 1,000 channels. On each of these channels we may have anywhere from zero processors, up to as many as 8-10, (EQ, compression, reverb and a wealth of others). Each of these processors process the audio, which in digital audio means runs a series of mathematical algorithms (calculation processes) and each of these algorithms is likely to introduce an error in the least significant bit (LSB). The LSB can only hold a 1 or a 0 but our algorithm (calculation) could easily result in say 0.6, so we set the LSB to "1" but we've introduced an error (noise). You're not going to hear this error in 16bit and obviously you're not going to hear it in 24bit but what happens when we've got say 80 channels of sound, each with say 2 processors and therefore a total of say 200 algorithms, the results of which are all being summed together? You will certainly hear the result of these combined errors at 16bit and even probably at 24bit. This problem is overcome by operating the processors and mixing at far greater bit depths, the errors in the LSB are now are so minuscule that even summing thousands of them together does not introduce anything remotely audible. It's common for many years for processors to be operating at 48bit, with mixing at 56bit and becoming even more common today to run everything (processors and mixing) at 64bit float. Once the mix has been made it can be brought back down to 16bit which is more than enough for every playback situation. I hear some audiophiles screaming: "I want the recording at it's original resolution". Well, you can have it! We record at 24bit (for reasons explained in the paragraph above) and we mix at various different bit depths, which generally we cannot print. The 56bit and 64bit float mixing environments common today are just for internal processing, we cannot actually write (record) files at these bit depths because they are useless, nothing can play them back and even if something could play them back you wouldn't be able to hear what was in the least significant 50bits or so (of a 64bit file) anyway. So when you see companies advertising "24/96 or 24/192 hi rez as it was created by the studio", that's a double lie! 24/96 (or 24/192) is not hi rez and is not as it was created by the studio. Which brings us on to mastering:
 
Mastering
 
Mastering is the process of taking a mix created in a recording/production studio and processing it so that it sounds good on the target audience's playback equipment, rather than sounding good only in the recording/production studio where it was made. This raises a number of important questions such as;what genre is the music, who is the target audience, what is their playback equipment and related to this, what format are we distributing? This obviously requires making assumptions and generalisations but if for example we are distributing jazz or classical on SACD we are most probably looking at an older target audience, who most probably take their music listening seriously (otherwise they wouldn't have bought an SACD player) and who are therefore the most likely demographic to have high or very high quality playback system/environment. So, we are far more likely to record, mix and master to a high standard and with a wide dynamic range. Instead of burning SACDs from this high quality master we could just convert it to 16/44.1, put it on a CD and this CD will sound absolutely identical to the SACD. For the record company there are two problems with this though: Firstly, this wide dynamic range, high quality recording might not be suitable for many playback systems/environments and secondly, how do you justify a significantly higher price for a "hi-rez" SACD which contains a recording indistinguishable from the much cheaper CD version? Of course there's an easy solution, you make a different 16/44.1 version which is distinguishable from the "hi rez" version! This brings us on to:
 
Dynamic Range
 
For practical purposes, dynamic range can be defined as difference between the highest energy in a signal (recording) and the lowest. As explained earlier in this thread, 16bit is capable of containing more dynamic range than you can safely listen to and even the best and most dynamic of SACDs have a dynamic range of no more than about 60dB and most recordings have a dynamic range of less than 40dB. To put this into perspective, 16bit is capable of roughly 1,000 times more dynamic range than even the most dynamic of SACDs. I can't understand this attitude from some audiophiles of wanting even more dynamic range than 16bit provide, enough dynamic range in fact to kill them if it were actually possible to use 24bits of dynamic range. Not only is this desire for 24bits (144dB) of dynamic range literally suicidal, it makes no sense in many cases to increase dynamic range of even some of the crushed music which many audiophile abhor. If you are listening to a recording in a quiet environment with very low environmental noise then yes, a decent dynamic range is a good thing and will allow the recording to sound breath and sound more alive but listen to that same recording say in a car while you're driving along the interstate and the high ambient noise will mean that you won't be able to hear half of recording without turning the volume up and nearly deafening yourself when a loud piece of the music comes along. I saw a thread on Head-Fi earlier where someone seemed desperate to get 24/192 playback from a galaxy smartphone. A quick look at the specs shows that at the headphone outputs this phone has a dynamic range of 92dB, well above the dynamic range of any commercial recording but well below the dynamic range possible with 16bit, why then is he wanting 24/192 playback? Even if his phone can handle 24bit format files it can't actually play more than about 15bits of them and for all intents and purposes ignores the other 9bits.
 
Coming back to mastering, why is it that many audiophiles seem to spend so much time, effort and money obsessing about aspects of their equipment which cannot possibly be heard but relatively little time and effort understanding and appreciating good mastering? I've heard the response to this question, which is "well the mastering is fixed on the recording and there's nothing we can do about it but we can do something about the equipment we use". Sorry, but this is horsesh*t, for two reasons: 1. If, as some fanatical audiophiles contend, power cables, digital inter-connects and ridiculously expensive speaker cables do actually make an audible difference, then every choice they make changes the mastering! For example, if an extremely expensive cable makes their system sound brighter, the mastering engineer almost certainly used relatively cheap copper cables and made the master exactly as bright as he/she intended, the expensive cable is changing the master! and 2. If these audiophiles spent more time on something which is easily audible instead of on what is patently not audible, they would soon learn what good mastering actually is and if they only purchased well mastered music, the record companies would soon take note and make sure their mixes/masters were of a higher standard to cater for this market! Let's not forget that mastering engineers are human beings, you can have an album mastered for $40 a song through the internet from a young newbie with little knowledge, experience or facilities or you can have an album mastered in one of the top mastering facilities by one of the world's great mastering engineers for $20k. Which album would you think is likely to sound better and which would you choose to buy if the price were the same or nearly the same?
 
In Summary
 
Can you hear a difference between 16bit recordings and 24bit ones? Yes, most definitely you can, I certainly have! There is however only 3 possible explanations 1. You are imagining the difference or 2. Some serious mistake has been made during mastering or the most likely 3. You are actually listening to different recordings/masters. 16bit, 24bit, 1bit (SACD) are just containers, what you put in those containers defines the quality of what you are hearing, not any inherent quality of the container itself. It's like trying a drink from a square bottle, liking it more than a previous drink you had from a round bottle and therefore deciding that you'll only drink from square bottles in future. In reality of course it's the drink which is in the bottle which makes you like the taste or not, not the shape of the bottle it's in.
 
One final thought: We've already covered that 16/44.1 is more than will ever be needed but going the other way, is there any potential problems with listening to the same recording at 24/96 or 24/192 or even 32/384? As far as bit depth is concerned, the answer is "no" there's nothing to be gained from higher bit depths but there's also nothing to lose, except storage space. This isn't necessarily true of the higher sample rates however! Most amps and speakers are not designed to reproduce any frequencies above about 20kHz, feeding them any significant amounts of frequency outside this range can cause them to create inter-modulation distortion (IMD), which is spurious tones or sounds within the range of human hearing. It's possible that some audiophiles might actually like this distortion or feel it is in some way "better" but for most sane people unexpected and unpredictable distortion is something to be avoided! Furthermore, once we get to 192k sampling rates and beyond it is impossible to correctly filter them according to the demands of the Nyquist Theorem. While it's extremely unlikely that this will result in any audible problems, it is in theory at least, lower digital fidelity. So don't get sucked into the marketing hype that 24/192k (or even worse 24/284) is somehow higher quality or higher fidelity than say 96k or 44.1k!
 
Gegorio

When I started this thread, the idea was to make as simple an explanation as possible, so it could be understood by Head-Fi'ers who may not have much interest in the deep technical detail. My last post followed the same vein. There's a broad spectrum of people on this site and it's a tough ask to write an article which covers all of them. In my last post I tried to balance as little detail as possible with enough detail to try and avoid rendering what I wrote too inaccurate.
 

There is one part I am struggling with, quoted above. How is it impossible to filter a very high sampling rate according to the demands of the Nyquist Theorem? In any case, we don't go to 192 KHz in order to try and record signals up to 96 KHz. We do so in order that we can use a simpler, lower order, low-pass anti-aliasing filter.
 
Nyquist demands that the signal is band limited. This means applying anti-alias and anti-imaging filters to remove the error signal above the Nyquist Point (fs/2). In the case of 16/44.1 it's relatively trivial to accomplish 120dB or more of attenuation in the stop band (the range of frequencies above the Nyquist Point) and therefore reduce anti-aliasing to below the digital noise floor. But with 24/192 we have a great deal more processing to accomplish but no additional time in which to accomplish it. At these very high sample rates and bit depths we start hitting the limits of the laws of physics in how fast we can perform the calculations required to implement a filter which reduces anti-aliasing to below the digital noise floor. The only way this is likely to change is with a new paradigm in processing, for example quantum computing could in theory solve the problem! All professional ADCs initially sample at incredibly high rates (many megahertz) but they do so with a greatly reduced bit depth, 5 bits or so generally. In other words, you either have more bandwidth OR more accuracy but not both! This is borne out in tests and in manufacturers' specifications; generally at 24/192 anti-alias attenuation is only accomplished down to around -80dB which results in distortion across the entire frequency spectrum, including the audible band! It's unlikely (but not impossible) that this failure to achieve sufficient anti-aliasing to fully satisfy the Nyquist Theorem is going to be audible but nevertheless, this additional distortion does mean that in theory at least 24/192 is lower fidelity than 24/96. For the same reason, 24/384 and 32/384 performs even worse than 24/192 and is even lower fidelity. Given the choice, no knowledgeable music recording engineer would ever record at anything higher than 24/96 but they are sometimes not given the choice by the record companies employing them. Unfortunately the audiophile world is driven by marketing more than by fidelity!
 
Gegorio


No, you're not misunderstanding, that's exactly the point when mixing and mastering, or at least one of the points! While you don't want listeners to specifically hear the noise floor, you do want them to hear the details of the recording all the way down to the noise floor. This means that the noise floor of the recording is hopefully roughly the same as the noise floor of the audiences' listening environment. Given that the average home listening environment is very roughly about 50dB, a recording which has a dynamic range of 60dB would mean the loudest peaks of the music would be at 110dB, which is extremely loud and would be uncomfortable for most people. In reality most people would play the music back quieter and simply not hear any of the details in the recording near the recording's noise floor because they would be considerably quieter than the noise floor of their environment. That's why very few recordings have a dynamic range as wide as 60dB and a 40dB or less dynamic range is so is much more common. Listening to music in something like a moving car, the ambient noise floor is way higher than an average home listening environment and therefore reducing the dynamic range of the recording even more is a good thing.
 
You've quoted the wiki article but you are confusing the container with it's contents! Yes, SACD like CD is capable of 120dB of dynamic range but for the reasons explained above, no one has or ever would release a recording with a dynamic range of 120dB, as even 60dB dynamic range is too much in the majority of cases!
 
Gegorio



I'm not quite sure I understand DSD. It was said in this thread and others as well by Giorgio that although SACD shouldn't sound any better than redbook, they often do because they have better masters. In that case, wouldn't that mean the only use of the DSD feature is to play back (mostly illegally) ripped SACDs? And what on earth is the difference between 2.8MHz and 5.6MHz DSD? Is there any point to using this crap or are these features all extraneous?

DSD uses 1bit rather than 16bit or 24bit. The downside of this is huge amounts of unwanted noise but with very high sampling rates there is plenty of space above the audible spectrum to move this noise, thereby making it inaudible. The SACD standard has a sampling rate of 2.8mHz and therefore a theoretical audio range of 1.4mHz, which is way more than enough to account for any difficulties related to reconstruction filters or other required process. This DAC obviously has a mode which exactly doubles that and indeed there are apparently some recordings available in this format and one or two DAC manufacturers are now offering a quad DSD oversampled rate of 11.2mHz. It appears that these oversampled DSD rates which have appeared are purely for marketing (snake oil) purposes. On the basis that if something has a higher number in it's specs, it's easier to market it as better, even if it isn't or even if it's actually worse!
 
Your basic premise is correct, look for the quality of the recording rather than for the distribution format. In some cases the best you will find will be a standard CD (16/44) in other cases it might be SACD or 24/96. You can always take a good master which may happen to be in 24/96 format, convert to a lossless 16/44 format to save space on say your smartphone and be secure in the knowledge that you're not loosing anything audible and are therefore listening to the highest quality available!
 
Gegorio

I shared this link in a different thread:
 
http://www.npr.org/blogs/therecord/2013/09/11/219727031/what-does-a-song-that-costs-5-sound-like
 
You can get some of the history of DSD from that article, and sense of why it is being revived today. The conclusion would appear that people are willing to pay more for "high quality", though what is and is not high quality is the hard part. I, for one, don't want to pay $5 for a file download and $50 for an album, but that's just me. The file sizes are atrocious, and there is no guarantee that a 'warm analog like' sound will be heard on each listen. The analog / digital snake oil bid drives me nuts.
 
When I complain about "sound quality" sometimes, I am not complaining about a specific format. What I am complaining about is clipped recordings, or recordings that seem stripped of dynamic range or for whatever reasons are harsh and nasty. An Mp3 has the same subjective properties (to my ears) as the CD it was ripped from, so I am not convinced that new formats and new equipment = better listening. Good headphones have made me aware just how far my recordings range (from amazing to disappointing). Are there ANY standards or guidelines for making good recordings these days?

I would expect that a higher frequency / bit accuracy could improve the relation to the original signal... To effectively implement the Nyquist Theorem, up to a point it can! But beyond that point the accuracy actually deteriorates simply due to the fact that the higher the sample rate the less time there is to process each sample. So, you can have high sample rates and few bits or lots of bits and a low sample rate but not both without compromising accuracy! In practise, modern ADCs take the former approach and initially sample at a frequency of many megahertz but with only a handful or so of bits. This initial sampling is then decimated down to the sample rate/bit depth selected by the user. Sample rates/bit depths from 16/44 up to 24/96 appear to fall into the optimum window for accuracy but with sampling rates beyond 24/96, accuracy deteriorates. I explained in a bit more detail at the end of this post.
 
Gegorio

[James Tanner]

Discusses 192:

http://www.audioaficionado.org/redirect-to/?redirect=http%3A%2F%2Fxiph.org%2F%7Exiphmont%2Fdemo%2Fneil-young.html

james

[Marius]

HI James.

indeed very interesting. But doesn't all of this obliterate most of your digital product line ? If all above 16bit 44.1 is useless in respect to audio quality, why bother. Even your newly presented BUC1 could be left unpacked 

I must say Ive always been very happy with the bcd1, but this seems to me to be rather disturbing for an audio manufacturer like you are, that always sticks to whats measured.

Thanks for keeping us informed!
Marius

[James Tanner]

HI James.

indeed very interesting. But doesn't all of this obliterate most of your digital product line ? If all above 16bit 44.1 is useless in respect to audio quality, why bother. Even your newly presented BUC1 could be left unpacked 

I must say I ve always been very happy with the bcd1, but this seems to me to be rather disturbing for an audio manufacturer like you are, that always sticks to whats measured.

Thanks for keeping us informed!
Marius

Its a great question Marius and one I wrestle with myself. 

A recent example is DSD which the market place is forcing us all to embrace for reasons that may have more to do with vested interest over scientific knowledge.  Offering a DAC today limited to 44.1 would be financially suicidal for a company regardless of your personal opinion on the merits of high resolution file playback.

I guess my position is I want to explore all areas without prejudice and ultimately the market will decide what is fact and what is fiction.

james

[Phil A]

Yes - Marius - good post.  I had a BCD-1 at one point and an old MicroMega DuoPro DAC (that only did 48k) and it sounded very similar to the BCD-1, however, it was obvious it did not do the best job stripping out jitter and the bass response was good but not quite right.  If I played them both on speakers that did not go below 100Hz, I'm not sure I could tell the difference.  I then moved forward with a BDA-1 for a bit and was able to hear the differences in higher resolution (was also using for a bit an HDMI audio de-embedder for a bit outputting 24/88.2 from an Oppo BDP-83 into the BDA-1).  Digital and video technology changes more rapidly.  Probably after I complete a move in a few months I'm going to get a new projector but probably not 4k as it is just too pricey now and if a few years I can replace a secondary older projector with the one I buy now.  It is the same for digital and that's why there was such a fuss about the SP1.7/SP2 not being future proof as originally advertised.  Manufacturers can't really do that. 

As consumers, we don't have a choice as to what record labels release on hi-rez PCM vs. DSD.  If something was recorded in 24/96 and they choose to release it on SACD then as a consumer if I want the highest quality from that recording I buy the SACD and have a DSD DAC.  Yes, it would be better if that recording were released in 24/96 but as noted consumers don't have that choice.  James also pointed to the market place forcing things. 

[HsvHeelFan]

My theory is that the most important part of the DAC is the output stage.

The DAC is just taking digital words of data and converting back into an analog waveform.  Jitter, of course, is an issue, but I think it's a fairly well understood issue from an engineering perspective and I think most manufacturers have strategies for managing jitter.

The number of bits just determine how many bits the word is going have.  Are you going to have 8/16/20/24/32 or more?  increasing the number of bits decreases the time width of the slices in the output, when the output is changing.

The Analog output stage where the analog signal is further processed to provide the drive capability to feed the pre-amp or amp is the key.  This Analog output stage can be located on the DAC chip itself or it can be located on the printed circuit board that the DAC chip is mounted on.  Component layout on the printed circuit board and signal path on the printed circuit board are very important and rarely discussed.

The world is an analog place and any digital system, at its core, is still analog.  Having "digital" allows us to store the media where we want and allows us to move it over a variety of mediums that pure analog systems can't (at least not easily).  But, at the end of the signal path, it's an analog audio signal that we enjoy.  None of us would like to hear  a digital signal driving our speakers.

HsvHeelFan

[Marius]

Well, speaking for myself and not trying to start a riot.. for me it means i can quite happily order ordinary cd's for I happen to like the looks of them and read through the actual booklets, and ripp those cd's lossless onto the bdp's drives for convenience sake,  without missing out on anything but the often much higher price for the same recordings in hires? (not even dearing to say one might be best off dl'ing the 16bit flacs that seem to be available more and more, and are even cheaper, albeit without the nice booklets and cases)

This must be a blow for HDtracks and the likes.

food for thought indeed.

Marius

[Russell Dawkins]

As a recording engineer, I record at 24/88.1 or 96, depending on whether the primary destination is CD or video.  I would record at 24/176/192 if my equipment would allow it. I use this level of resolution in the spirit of good engineering practice, and stay at high resolution through the various manipulations of mixing and mastering (for example, EQ, dynamics, level adjustments, spatial adjustment through MS, pitch and tempo adjustments). When all is done, the signal is dithered on the way down to 16/44 for CD.

When I listen in a 'consumer' frame of mind I don't really care whether the recording is 16/44 or 24/96 or 24/192. To me, the choice of loudspeaker completely obliterates these minutiae as a sonic variable, as does the quality of the mix - which also varies enormously.

The music itself, as usual, is the most important variable - and I will happily listen to 25% distortion if the music is good enough.

[Marius]

Good points Russel,

My main reason for playing 24bit in all its variations  is when it's the native resolution, thus avoiding any other unnecessary processing along the way. For this we need a great dac and digital player.

Or, if a previous good master was delivered to cd in a compressed or other suboptimal digitised way, and new remastering could bring back the good originals? Compare the various Solti Ring's and enjoy the reading.....

or the ring on Testament to name but two great older recordings. Then again, I am enjoying the latter on regular CD's, with accompanying booklets and pictures to hold and read....
And, maybe to underline all of this: I've never been able to agree on those that state the bcd1 sounds better through the BDA than on its own. I've used all possible connections, and it's only for convenience sake I play the Bcd through the BDA, being able to select it as source with the BR2....


Anyway,

Thanks for your input!
Marius

[jeffjensen]

Thank you for summarizing and sharing all this James!  It has been a fascinating read...

[Mag]

The finger is not appreciated.
Good for you for thinking you are able to hear extra harmonics with 48 over 44k.

48kHz sampling rate allows just more than one (1) semitone higher limit than 44.1 kHz. There are 12 semitones in one octave. Our hearing range is just shy of 9 octaves as adults; 10 as a child. How many extra harmonics do you think that allows??
48k sampling exists not because it's intended to be better than 44.1k, but to make it incompatible with 44.1k - to separate the pros from the consumers. 44.1k was the pro standard; 48k was the standard chosen when consumers were first able to record digitally - with DAT recorders and minidiscs and the like, so they could not make CDs from their recordings without going to a professional who could sample rate convert to 44.1 for CD release.
Now anybody and his dog can do this conversion - and plebes can argue over the audibility of harmonics with 44k sampling compared to 48k.
Here's my finger! 

On the other hand, 24 bit word lengths are useful for headroom in recording and the benefit of >176.4k sampling is audible.

Sorry you did not perceive my humor in the smiley, think of the Mugsy Character in the Bugs Bunny Cartoon.

Anyway, so when I play my Blu-ray concert video, what I'm perceiving as better harmonics is just a better uncompressed recording. Perhaps their best recording to date, with a new heavier bass sound?

[krikor]

Thanks very much for this thread... And very interesting educating reading. Since most (99%) of my library is 16/44.1 I find this heartening. Considering the purchase of a new DAC, but want one that does red book proud, yet most simply tout the latest high Rez capabilities.

[James Tanner]

Thanks very much for this thread... And very interesting educating reading. Since most (99%) of my library is 16/44.1 I find this heartening. Considering the purchase of a new DAC, but want one that does red book proud, yet most simply tout the latest high Rez capabilities.

Hi Krikor

Glad you found the above discussion helpful. Your comment about a DAC that does well with 44.1/16 Bit material though brings up a very good point.

When we were testing all the different DAC chips for our Bryston BDA - DAC we did find that some DAC’s measured better at 44.1 and 88.2 processing while starting to have trouble at 192. Other DAC’s measured better at 192 but were worse at 44.1. 

One of the reasons we choose the DAC we did is because it was the best across the whole band between 44.1 and 192. In fact the new AKM DAC we use in the BDA-2 had about a 5dB improvement in noise and distortion over the previous DAC we used in the BDA-1 from 44.1 through 192kbps.

That being said one of the main reasons the Bryston DAC’s perform and sound as good as they do has a lot to do with the “fully discrete analog stages” and the "independent power supplies for digital and analog sections" implemented as opposed to the smaller differences in any particular  DAC chip used.  Also Jitter (timing issues) are critical in a DAC and if you look at the latest test reports on our BDA-2 DAC you will see we are attaining jitter numbers in the 10ns range – which is at the threshold of even the best digital measuring gear available.

James

[Mag]

So I'm trying to wrap my head around the truth, information from this thread.

I conclude that listening to a Blu-ray in Lpcm 48k 16 bit uncompressed recording is perhaps the best that a recording is capable of achieving. 48k being only 1 octave difference from 44.1, not a significant difference.

Then playing a dvd that has been compressed to dts, I can hear a degradation in sound quality from the previous Blu-ray.

That means the typical cd of 44.1 16 bit has various levels of compression to fit the data on to the disk. A cd recording can sound pretty darn good but typically the difference that I claim to hear between dvd- 48k Lpcm 16 bit and cd- Lpcm 44.1 16 bit can only be due to compression.

[srb]

That means the typical cd of 44.1 16 bit has various levels of compression to fit the data on to the disk. A cd recording can sound pretty darn good but typically the difference that I claim to hear between dvd- 48k Lpcm 16 bit and cd- Lpcm 44.1 16 bit can only be due to compression.

Although dynamic compression exists on some CDs to participate in "loudness wars" (more prevalent on rock and pop CDs), even using the non-extended original commercial CD capacity which allows 74 minutes of non-compressed 16/44.1 data, unless the album exceeds that length, there would be no reason to use data compression on a CD to "fit the data on the disk".

Steve

[Russell Dawkins]

So I'm trying to wrap my head around the truth, information from this thread.

 ...48k being only 1 octave difference from 44.1, not a significant difference. 

I said 1 semitone difference, which is 1/12th of an octave. For an octave you would be talking 88.2k vs 44.1.

The musical note "A" to which an orchestra tunes, is 440Hz. A# (a semitone higher) is 466.16Hz; B is 493.88Hz.

48.0/44.1 = 1.0884. Multiply 440 by that and you get 478.91, which is nearer A# than B.

I think an octave is significant, but to get that you would have to be comparing 88.2k and 44.1k sampling rates.

Meanwhile, as I say, all of this is truly insignificant compared to the effect of the recording engineering quality and the speaker quality - even the difference between 44.1k and 192k is trivial compared to a number of other variables. Pursuing these details in this manner is to miss the big picture.