How many bits, how fast, just how much resolution is enough?

[Jud]

There are dozens, likely hundreds. of published studies showing lack of statistical differences for the higher bit rates. Maybe they are all wrong? Personally, it doesn't cost me a lot to get that source material, but I figure I am likely getting more care in recording/mastering...

Similarly, there some studies showing that mp3 files cannot be distinguished from redbook. It costs me nothing to use Apple Lossless instead tho (well, ok - I guess it cost me a little more for a higher capacity iPhone).

There was a test done here at CA years ago by a member known as Julf, with the cooperation of the head of the Bis recording label. It included RedBook, mp3, and hi res files. The "winner" by a fairly large margin was a RedBook file 1dB louder than the others. This quite clearly shows even a loudness difference so small that it is not consciously perceptible is sufficient to sway preference in the context of an audio A/B test. (No tester identified from listening that one of the files was louder.)

 

By the way, if one was listening for a difference between a RedBook and a direct to DSD recording of the same material, what audio characteristics - phase, frequency response, time domain response, imaging, soundstage, noise - should one listen for, and what should these differences sound like?

[jabbr]

Thx, unfortunately only one of those is a peer-reviewed publication. They could still be correct, but your claims are not supported, esp. by the Stereopile 'thing'. I'll take a look at the one AES later on...

 

We are all free to decide how we spend our money. We are also free to decide which arguments sway us. I tend to reserve "argument by authority" for those I consider proven authorities, and as its often easy to get ones name on a paper, a single paper often does not convince me. Of course where there are more than a single paper, they are often contradictory.

There are dozens, likely hundreds. of published studies showing lack of statistical differences for the higher bit rates. Maybe they are all wrong? Personally, it doesn't cost me a lot to get that source material, but I figure I am likely getting more care in recording/mastering...

 

Yep. I figure its best to get the format closest to the recording. Of course I very well remember my first A/B test in a colleague's apartment between a CD --- touted at the time as the pinnacle of audio reproduction --- and an LP ... and well the CD sounded like cr*p in comparison. So is it the format or the ancient NOS brickwall filter? Miska's software goes the furthest among anything to convince me it is more the filter than the format.

 

That buy's him more street cred in my book than an AES Fellowship. Is that hard to get?

[Nelson_de_Leon]

It all started way back in about 1982 with the arrival of CDs originally providing a resolution or data rate of 44.1khz/16 bit. Now we can obtain or generate music files at a resolution of 384khz/24 bit or higher.

 

I've personally gotten caught up in the rush to higher resolution and have upgraded several times to be able to play back 384khz/24bit files and also getting into DSD64 and now DSD128. I'm not sure

 

Advances in electronics will, probably, allow for almost limitless increases in resolution and storage space to house those huge files is exceedingly low but at what point does the human ear stop hearing any improvement? Where will it end?

 

 

It ends when the differences in resolution is not already audible to the listener.

 

Sent from my ASUS_Z012D using Tapatalk

[Jud]

And your result still doesn't tell anything about audibility of the format differences alone, only tells about that particular DAC.

 

In addition, even in digital domain, there is no single way to do the conversion to/from RedBook, each of those also produce different results.

 

So overall this is complex topic and the results are not straightforward.

 

 

Word (as they say).

 

 

As a lawyer, what I nearly always find when non-lawyers talk about the law is that they tend to leap to sweeping over-generalized conclusions. I'm sure something very similar occurs to you as you read what non-professionals write about filters, modulators, audio software, etc.

[Ralf11]

There was a test done here at CA years ago by a member known as Julf, with the cooperation of the head of the Bis recording label. It included RedBook, mp3, and hi res files. The "winner" by a fairly large margin was a RedBook file 1dB louder than the others. This quite clearly shows even a loudness difference so small that it is not consciously perceptible is sufficient to sway preference in the context of an audio A/B test. (No tester identified from listening that one of the files was louder.)

 

this is well known and the usual spec. is 0.2 or even 0.1 dB - it has to be controlled for and usually not just by ear either

[Ralf11]

I very well remember my first A/B test in a colleague's apartment between a CD --- touted at the time as the pinnacle of audio reproduction --- and an LP ... and well the CD sounded like cr*p in comparison. So is it the format or the ancient NOS brickwall filter? Miska's software goes the furthest among anything to convince me it is more the filter than the format.

 

That buy's him more street cred in my book than an AES Fellowship. Is that hard to get?

 

 

early CDs were notorious for bad mastering; that is likely what you heard

 

a fellowship in a professional scientific or engineering society is a rather large chunk of cheese

 

you realize that I have asked for actual studies, rather than arguments from authority, but what the advocates of higher bitology have posted is almost nothing but arguments from authority ???

[jabbr]

early CDs were notorious for bad mastering; that is likely what you heard

[\quote]

 

That is entirely the point. When you insert A/B/A conversion software or circuitry you depend on the supposed fidelity. When you compare formats you are also comparing conversion software, you are comparing how your hardware accepts the format, or you are comparing mastering.

 

It's often easy to show lack of an effect in an experiment but it's much harder to prove lack of an effect.

 

a fellowship in a professional scientific or engineering society is a rather large chunk of cheese

 

you realize that I have asked for actual studies, rather than arguments from authority, but what the advocates of higher bitology have posted is almost nothing but arguments from authority ???

 

I wouldn't lump all the "advocates" together. I'm on the "not convinced either way by the arguments so err on the side of high res all else equal" but mastering trumps.

 

Also there are so so many commercial biases that I tend to be skeptical of what I can't hear.

[mayhem13]

Word (as they say).

 

 

As a lawyer, what I nearly always find when non-lawyers talk about the law is that they tend to leap to sweeping over-generalized conclusions. I'm sure something very similar occurs to you as you read what non-professionals write about filters, modulators, audio software, etc.

 

A failure of fundamental responsibility to serve the law, which serves the people. How we've gotten to where we are is a huge portion of the problem..........that it takes professionals to explain the difference between right and wrong.

 

You may agree.......or not.......inconsequential i suppose........but just a relevant as professionals suggesting we like A vs B. Debacle?.......I dunno?

[esldude]

snip.....

 

Then the first pro/audiophile asynchronous sample rate conversion (ASRC) DACs hit the market, most notably from Benchmark and Lavry. These DACs resampled to non-even multiples of the input rate (in Benchmark's case, 110 KHz; I don't know what Lavry used, though I recall the input rate was limited to 96KHz) as a means of jitter reduction, prior to putting the resulting bitstream through sigma-delta modulation to MHz sample rates. I recall Lavry in particular publishing white papers saying one certainly didn't need higher sample rates than 96KHz and suggesting higher rates were actually deleterious, while not mentioning their own DACs internally used the same MHz sample rates as everyone else.

 

But I'm sure these papers, which did as much as anything to spur discussion of how 44.1 or surely 96KHz rates were perfectly adequate (when even inexpensive non-audiophile commodity players used chips designed by non-audiophile engineers that upsampled internally to MHz rates) had nothing to do with Lavry's commercial objectives.

 

If you read Lavry's papers he isn't hiding anything. He talks about how his own converters are delta sigma and run in mhz range. Even then what he talks about as getting less accurate sampling at too high a speed and fewer effective bits is still in effect. He warns the reader to keep both sampling rates clearly separate in their minds when trying to understand how this works. Doesn't sound like someone pulling a fast one if you read it rather than what others say about the paper.

 

http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

[audiventory]

Most complex in bit depth issue, what we don't know exactly where threshold of our perception of "qualitative"/"non-qualitative" sound.

 

1. We can measure/calculate dynamic range. It's objective.

 

Objective here is "same for everybody, if use identical way of measurement/calculations".

 

2. Dynamic range impact to signal noise ratio of audio signal. We can calculate/measure it too. It's objective too.

 

3. But we can't say exactly what signal/noise ratio is allowable as qualitaive for us.

 

And subjective aspect of objective parameter are discussed.

 

Esldude before wrote:

 

Some revisionist history here I think. Before the CD became a standard various digital recording formats were around being experimented with for music. They ranged from 13 bits to 16 bits. Sample rates ran from 32 khz to 50 khz. Some used various pre and deemphasis schemes. When Philips and then Sony decided to back a new standard Philips had decided 14 bits was enough while Sony insisted it be 16 bits.

 

It fine illustrate what filst current technical abilities are considered.

 

And after it improvement of subjective perception of objective changes are discussed.

[hytechrednek]

When I read these comments, it reminds me of a guy who was my mentor when first entering the world of Liaison Engineering and the MRB. A German immigrant that was about to retire, he tried to warn me about our analysis groups and the kind of hypothetical arguments they could come up with. He told me I had to be able to assess their data, accept what was relevant to the problem at hand, yet keep a handle on reality. In his thick German accent, he gave me a metaphoric example that has always stuck with me. He said if you position a naked man and naked woman exactly 10 meters apart and have them close the distance between each other by 50% every minute, a statistician will tell you the man and woman will never make contact. There will always be some distance between them. However, a good Engineer knows they will get close enough for all practical purposes.

[kelly200269]

Rob Watts has always said that there is a lot more that 16-bit audio can give.

When you listen to the Hugo, you know exactly what he means. 16-bit files played thorough my Hugo sound better than higher bit-rate files played through my other DACs.

Which leads me to believe that 16-bit audio is 'good enough', and that the variable in all this is the type of decoding.

 

Sent from my E6653 using Tapatalk

[STC]

..... He said if you position a naked man and naked woman exactly 10 meters apart and have them close the distance between each other by 50% every minute, a statistician will tell you the man and woman will never make contact. There will always be some distance between them. However, a good Engineer knows they will get close enough for all practical purposes.

 

+1.

[semente]

Rob Watts has always said that there is a lot more that 16-bit audio can give.

When you listen to the Hugo, you know exactly what he means. 16-bit files played thorough my Hugo sound better than higher bit-rate files played through my other DACs.

Which leads me to believe that 16-bit audio is 'good enough', and that the variable in all this is the type of decoding.

 

That doesn't make sense...

The question is not whether Redbook with your current DAC sounds better than HR with your previous DACs but how do Redbook and HR compare with your current DAC.

 

R

[r_w]

Not so sure about that... the question is whether 16/44 is good enough as source data to properly deliver the content within the hearing range of humans (frequency & dynamic range) on DACs than are designed well enough to let that happen.

 

"How many bits, how fast, just how much resolution is enough?"

 

... if one of his DACs tells him redbook is enough, then his post is right on topic in my book. If his other DACs don't, that's not necessarily redbook's fault.

 

 

---on a another note---

 

If we (CA) agree that freq range 20hz-20khz is being captured and reproduced, then dynamic range shortfall of 16 bit is next to examine, which is fine on paper but not much music contains huge dynamic ranges like that... DR Database for example makes for some depressing reading.

 

 

;-)

 

 

 

That doesn't make sense...

The question is not whether Redbook with your current DAC sounds better than HR with your previous DACs but how do Redbook and HR compare with your current DAC.

 

R

[semente]

Not so sure about that... the question is whether 16/44 is good enough as source data to properly deliver the content within the hearing range of humans (frequency & dynamic range) on DACs than are designed well enough to let that happen.

 

"How many bits, how fast, just how much resolution is enough?"

 

... if one of his DACs tells him redbook is enough, then his post is right on topic in my book. If his other DACs don't, that's not necessarily redbook's fault.

 

 

---on a another note---

 

If we (CA) agree that freq range 20hz-20khz is being captured and reproduced, then dynamic range shortfall of 16 bit is next to examine, which is fine on paper but not much music contains huge dynamic ranges like that... DR Database for example makes for some depressing reading.

 

 

;-)

 

I understand your comments but I think that if he wishes to know if Redbook is "enough" with his current DAC he should compare it to HR converted by that same DAC and not by another one (which I assume didn't perform as well with either format).

This is what I would do if I wanted my findings to be meaningful.

 

I recommend using Mario/PlayClassics test files.

 

My comments from that thread regarding Redbook vs. HighRes comparisons:

 

Yesterday morning I spent some time listening to the Format Test Files kindly provided by Mario Martinez.

Even though I generally find listening evaluations a humongously tedious affair, I must have been in the right mood and the neighbourhood was very quiet which was also helpful.

 

The proceedings of my listening session were as follows:

 

16/44.1 track

24/44.1 track

16/44.1 track

-

16/44.1 track

16/96 track

16/44.1 track

-

24/96 track

16/44.1 track

24/96 track

 

First of all, it was a very interesting experience to listen to different performers and instruments playing with such a tonal and acoustical seamlessness, almost as if I were listening to a single musical event.

 

Then something happened which really surprised me: I had listened to two of the recordings included in the set a reasonably amount of times before in 24/96 and it didn't take too long to identify the shortcomings of the 16/44.1 version.

 

And once I nailed the characteristics of those shortcomings it was hard to miss them.

I can still enjoy 16/44.1 but I am convinced of it's lower fidelity.

R

[Jud]

Not so sure about that... the question is whether 16/44 is good enough as source data to properly deliver the content within the hearing range of humans (frequency & dynamic range) on DACs than are designed well enough to let that happen.

 

"How many bits, how fast, just how much resolution is enough?"

 

... if one of his DACs tells him redbook is enough, then his post is right on topic in my book. If his other DACs don't, that's not necessarily redbook's fault.

 

 

---on a another note---

 

If we (CA) agree that freq range 20hz-20khz is being captured and reproduced, then dynamic range shortfall of 16 bit is next to examine, which is fine on paper but not much music contains huge dynamic ranges like that... DR Database for example makes for some depressing reading.

 

 

;-)

 

 

Hi r_w, been meaning to get back to you regarding your earlier response and why I talked about what I did (the history of CD players, DACs, oversampling, filtering, etc.).

 

Frequency response is really not in question here. *It is the wrong thing to be looking at when discussing the adequacy of digital audio standards.* Why is that?

 

- First, it's a fact that a 44.1KHz sample rate is mathematically adequate to reproduce all audible frequencies. If you want a way to picture what's going on with the math, here's one:

 

Take a single sample point. There are an infinite number of curves that can be drawn through it.

 

Add a sample point, for a total of two. There are a smaller number of curves that will run through both points.

 

Now adjust your sample rate to just above double the highest frequency rate you are interested in. That adds a third sample point. As long as we are talking about frequencies that don't exceed just under half that sampling rate, there is one and only one curve that will run through all three sample points. So you have defined the sampled curve along its entire length and not just at the sampled points. That's a simplified English version of what the Sampling Theorem says. That's why more samples will not help to define the signal better. So a 44.1KHz sample rate will perfectly reproduce any signal of just under 22.05KHz or lower.

 

- Well then why upsample?

 

The problem is not the sample rate itself and its adequacy for reproducing the frequency range - as we've already seen, we can perfectly define any signal within our frequency range of interest with a 44.1KHz sample rate. The problem is that the Sampling Theorem requires a perfect filter that will pass all frequencies below half the sample rate, and completely eliminate all frequencies from half the sample rate on up. This perfect filter would have to operate instantaneously, in no time at all, would have to keep operating over an infinite period of time, and would have to operate perfectly as to frequency, i.e., would have to create an absolutely perfect vertical line between full signal and no signal on an oscilloscope frequency sweep. No such animal exists.

 

Because there are no perfect filters, all filters must balance two types of errors, frequency domain errors (aliasing or "leaking") and time domain errors ("ringing" or "smearing"). Mathematically, improving your filter's behavior in one aspect worsens it in the other aspect - that is, for a given filter, adjusting its parameters for less aliasing gives you more ringing, and vice versa. That's just sheer math, and it's unavoidable. So what you need to be looking at when discussing the adequacy of digital audio standards (such as sample rates) is not frequency response, but aliasing and ringing distortion.

 

The entire reason for upsampling is to make it more practical (easier and cheaper) to design filters with adequate distortion performance.

 

OK then, how should we identify adequate performance? What should we listen for?

 

Well, this comment is already long enough. Perhaps in my next one....

[jabbr]

I understand your comments but I think that if he wishes to know if Redbook is "enough" with his current DAC he should compare it to HR converted by that same DAC and not by another one (which I assume didn't perform as well with either format).

This is what I would do if I wanted my findings to be meaningful.

 

I recommend using Mario/PlayClassics test files.

 

 

You also should do the same comparison using HQP and/or AuI to upconvert to the native format of your DAC from these source files.

 

This will test whether the differences you hear are related to the filtering/sampling done by the DAC (vs capabilities of CPU/software)

[Jud]

You also should do the same comparison using HQP and/or AuI to upconvert to the native format of your DAC from these source files.

 

This will test whether the differences you hear are related to the filtering/sampling done by the DAC (vs capabilities of CPU/software)

 

Yes, and not incidentally it will get around two tremendous problems with such comparisons - they often involve different masterings and/or differences in loudness between the two files.

[semente]

You also should do the same comparison using HQP and/or AuI to upconvert to the native format of your DAC from these source files.

 

This will test whether the differences you hear are related to the filtering/sampling done by the DAC (vs capabilities of CPU/software)

Indeed.

There are so many variables: upsample to PCM or DSD, to a multiple frequency or maximum DAC sample rate, minimum phase or linear phase filters, SOX vs izotope vs HQPlayer vs Aul vs DAC ASRC, in line or off line, etc.

 

This is what I dislike about computer audio, it's as fussy as vinyl...or more.

 

R

[Jud]

Indeed.

There are so many variables: upsample to PCM or DSD, to a multiple frequency or maximum DAC sample rate, minimum phase or linear phase filters, SOX vs izotope vs HQPlayer vs Aul vs DAC ASRC, in line or off line, etc.

 

This is what I dislike about computer audio, it's as fussy as vinyl...

 

R

 

You can certainly choose a "plug and play" experience that can be very, very good. But there are folks (counting myself among them, or us) who would like to try to get the very best possible from whatever we have or can reasonably afford, and at least at this point in the development of digital audio that does involve some fuss.

 

You have your choice of whether the fuss occurs on the learning end or on the playback end, i.e., whether to try to learn about the reasons behind the way things sound or spend time on trial and error with playback settings.

[Jud]

early CDs were notorious for bad mastering; that is likely what you heard

 

a fellowship in a professional scientific or engineering society is a rather large chunk of cheese

 

you realize that I have asked for actual studies, rather than arguments from authority, but what the advocates of higher bitology have posted is almost nothing but arguments from authority ???

 

We certainly know it becomes easier/cheaper to achieve filters with levels of distortion acceptable to engineers at higher sample rate input. We're then left with the question whether levels of distortion considered unacceptable by engineers make differences that are audible, or even, short of being audible, have other conscious or subconscious effects (for example, "listening fatigue").

[Jud]

this is well known and the usual spec. is 0.2 or even 0.1 dB - it has to be controlled for and usually not just by ear either

 

Yes, exactly. Do you suppose if we'd been told one of the files was 1dB louder we could have identified which one?

[jabbr]

Indeed.

There are so many variables: upsample to PCM or DSD, to a multiple frequency or maximum DAC sample rate, minimum phase or linear phase filters, SOX vs izotope vs HQPlayer vs Aul vs DAC ASRC, in line or off line, etc.

 

The number of options is bewildering.

 

To simplify, my approach is:

 

1) Try to obtain recordings/works in the native format of the mastering engineer

 

2) Convert these files to the native format and resolution preferred by the particular DAC

[r_w]

Hi Jud,

 

Good post, thanks for the explanation (TBH... I get a bit lost when talk moves to impulse / ringing / etc... that's way over my pay grade).

 

The answer to your closing statement might be:

 

Q. play some tracks... do you love what you hear?

 

A. If yes then play some more, if no, amend system, and/or go hires?

 

:-)

 

 

OK then, how should we identify adequate performance? What should we listen for?

Well, this comment is already long enough. Perhaps in my next one....