In Search Of Accurate Sound Reproduction: The Final Word!

[CuteStudio]

6 hours ago, fas42 said:

My belief is that there are no limits to domestic playback - given amplifiers of sufficient competence, and the whole chain sorted, any sane SPLs can be produced, with a full measure of the impact of the original performance.

 

 

In my view we already know how to build excellent amplifiers. Few actually seem to do so, but there are ways and some really good amps are possible, and some even built.

 

In my view we have two problems:

 

1. Dodgy source material

2. Imperfect speakers

 

We should really be on 24 bit 96kHz by now, but we're not and that's a bit sad, especially given computing technology and storage improvements in the past 3 decades of the CD format.

 

We can fudge the 44.1kHz by upsampling so real filters have a chance, but the 16 bit is still too low, despite people's faith in 'dither', a technology not of much use on transient waveforms in my view, and it 16 bit is so good, what's wrong with 8 bits, or 4 bits if dither is so great?

 

Then there is the terrible mastering of most of todays music, which is simply wrong and a deliberate reduction of fidelity.

 

But it's speakers were the most distortion happens and where the most work needs to be done. It's tricky due to weights, resonances, dispersions, and then we compound it by sticking it into a cabinet - a sort of rectangular version of a guitar body, and put in crossovers, another obstacle in the way of our sound, often sticking them in the loudest place in the room: inside the speaker box. Doh!

 

 

[CuteStudio]

10 hours ago, fas42 said:

 

I'm afraid I will have to disagree with quite a bit there - yes, excellent amplifiers are possible, Bryston is one that comes immediately to mind.

 

Dodgy source material? No, redbook is 100% OK; I realised this 30 years ago, and have looked with mild amusement at the enormous thrashing around that's occurred in the following decades, trying to 'improve' digital source.

 

Terrible mastering survives competent reproduction systems - but exposes shortcomings in the playback chain very aggressively. The loss of fidelity is not in the recording, but in the playback rig trying to reproduce it.

 

Speakers have got a bad rap, wholly unjustified! They can perform amazingly well, given some attention to smaller things. But I agree about crossovers being exposed to the full impact of the sound energy - should be handled in smarter ways.

 

Disagreeing is a good, usually the source f the best discussions

 

For modern recordings - especially of rock/pop - mastering quality means a CD of Californication sounds significantly worse than the 192k lossy pre-master version. So the reality is we rarely get close to taking full advantage of redbook even.

 

That was my main 'dodgy', but it's interesting that you think redbook is OK. Digital source has been improved, most DVDs and SACD has a better matrix. There are two major issues with redbook:

 

441.kHz. No one can build a real filter that works for that, unlike at 96 or 192 where the filter is trivial. So immediately we are into fudges to 'decompress' the waveform level-time matrix and various schemes are used for this. My favourite is a Behringer rate converter, which uses a Sharc DSP to calculate the missing points (the decompression phase) as accurately as possible and then the resultant 88.2k can be filtered with a HiFi filter. We still have an information limit of 22.05kHz which is low, but at least the filter can work. 

 

A 96k source doesn't need a DSP to calculate the approximate missing points, because that's supplied in the information, because like it or not, at 44.1k part of what you are hearing is the guess of where the intermediate points go.

 

Which brings me to the terrible 16 bits. If 16 bits is Ok why is 8 bits bad? In SeeDeClip4 I can switch the mastering to 8bit and (at least in Chrome) I can listen to that, correctly dithered with a nice gaussian dither, and you know what? It sounds Ok. A lot better than you'd think. But not as good as 16bit, so if 16 bit is better than 8 bit, surely 24bit is better than 16 bit, especially as many real work DACs are now 18-20 bit.

 

Many adherents of 16 bit then claim dither is the saviour, and point toward the maths that proves it's perfect. And the maths is right, for a 1kHz signal it pretty much IS perfect. But not the music. The maths is correct for a continuous wave, not for a short, transient one. If you study waveform shapes have a look at a quiet HF part and you'll see for say a soft cymbal strike there are shapes to the waveform. These shapes are wrong with a low bit + dither, and can only be correct with a higher bit rate. 

 

Linearity:

The 16 bit scale is 32767 bits per side of 0v, so you'd think that the resolution was 1/32767, but you'd be wrong for a reason that few people talk about: the logarithmic loudness of sound. What this means is that the distortion rises as the sound level falls. 6db down and you are at 15bit audio. That trailing ambience at -60dB? Welcome to 6bit audio.

 

Remember that 8bit music I said was quite good? Digital is 6dB per bit (each bit halves the signal) so 48dB down and most classical listeners will spend quite a bit of time oohing and aahing at basically 8bit digital music.

 

As for speakers, it's simply that the distortion is higher than all other parts of the chain (poor mastering excepted), 10-20% distortion is not unusual, there are some very good speakers around, but generally if people want a better sound the best thing to change is the speakers IME.

 

Perhaps some of the disagreement is due to the definition of HiFi. i appreciate lossy systems and I categorise the CD format as one such format, because the matrix is too sparse. However HiFi for me means reproducing the actual waveform, not just in apparent sound but in actual shape, from the air which hit the microphones to the air in front of the speakers. This definition means that fudging transient waveform shapes wit dither is not HiFi, it's in that class of audio which MP3 covers: fooling the ear and holing it will sound Ok.

[CuteStudio]

The XY matrix of time vs level. Imagine analog as a wave drawn on a whiteboard.

The digital matrix can then arrange dots in a grid that approximate that waveform. Then we store the dots, put them on a shiny disk and Fred plays that disk, which is where the dots are used to form an approximation of the analog waveform. If you imagine a 44.1 16 bits matrix as a grid of dots, then if you fill in twice as many columns (88.2kHz) you can see the grid is denser. Then add 255 levels between each row (24 bit), and you have a denser grid still.

This is why I use the term matrix, because what we are actually doing is putting a matrix over the waveform and choosing some squares/dots.

 

So 24bit/96kHz digital sampling has a far more accurate recording than a 16bit/44.1kHz as the matrix is much finer. That's just a simple fact, the argument is that some people are very attached to 16/44.1 and claim it's good enough. Personally I can't see why we want just 'good enough', 24bit/96kHz is technically trivial these days and there is no cost/technology reason to keep resisting this natural progression to higher fidelity.

The only benefit to 16/44.1 is that it's easier to download from the internet - is that what HiFi is now?

 

" It appears that somewhere between 16 and 20 bits is sufficient to capture anything a human could possibly hear. "

 

So what's the resistance to 24bit?

Why does it MATTER if 24bits is better than our hearing?

Is that really the game: to just up the tech enough to fool the average ear? Why can't we just standardize on something better? It's hardly difficult. We casually buy 1TB disks, 3GHz multi core processors and phones with 2GB of RAM and we're seriously arguing that we only need 16/44.1? Why are we doing this?

Why even the discussion?

We don't apply the 'just good enough' criteria to any other part of our Hifi so why apply it to the recording format?

 

"The maths you speak of is generic. It applies to all waveforms."

The maths applies to all continuous waveforms. Try it on a waveform 4 samples long. Not good. Worse than no dither. Dither is a way of averaging level errors over time: but on transient events you have no time, you want them accurate right then, not 200 samples later.

If we all listened to church organ music I'd agree that dither was a good answer, on transients it is demonstrably inadequate: there is no Free Lunch.

 

"The distortion level at -60 dB signal is the same as full scale."

Mmm - I'm not getting that here: 

At 0dB you have 16 bits, with a quantisation distortion of 1/65535 = 0.0015%

At -60dB you have 6 bits, with a quantisation distortion of 1/63 = 1.59%

 

BTW I'm talking about a quiet sound, not a quiet sound in the presence of a louder sound. For instance a soft flute solo on a dynamic recording with a full orchestra.

 

[CuteStudio]

10 minutes ago, AJ Soundfield said:

Umm, yes that can be done, transparently too. I can cite a dozen studies from the recent one Bob Stuart perhaps accidentally mentioned showing this.

 

That's fantastic, could you please name some gear that does this?

I Always though a 96dB filter in 0.05kHz was impossible but I'm willing to learn!

 

10 minutes ago, AJ Soundfield said:

 (sorry, "I heard it, I said so" doesn't qualify).

Please don't be sorry, I think we can all agree on this!

 

10 minutes ago, AJ Soundfield said:

Red Herrings are bad. Again, if you claim 16 bits as insufficient for consumer playback systems, list:

When you say 'consumer playback systems, do you mean accurate sound reproduction: the final word?

I'm not trying to pretend the car stereo needs more than 16 bits.

 

10 minutes ago, AJ Soundfield said:

1) Your loudspeakers

2) Broadband ambient noise measurements of room.

3) The track(s) used with 16bits of dynamic range above your noise floor at 1-3kHz

I'm not sure the 'the rest of the system isn't very good' is sufficient reason to not use 24bit/96kHz.

'Because we have no space', or 'CPUs are too slow' would be valid reasons, but I'm not getting this one. Are you saying 24/96 is too good so we shouldn't use it?

 

 

[CuteStudio]

57 minutes ago, mansr said:

We are always dealing with band-limited signals. There are no discontinuous waveforms.

Isn't dither a statistical mechanism?

How will we know if there are enough samples in a transient waveform for dither to work?

 

57 minutes ago, mansr said:

The absolute level is the same. If it is low enough to be inaudible, it doesn't matter what the signal level is.

The absolute level may be the same, but for classical can be pretty high, meaning that the -60dB level would be plainly audible. 

 

I'm not sure why we don't just go to 24bit for music, DVD movies appeared to go there some time ago. Ironically their sound is often better mastered too

[CuteStudio]

2 minutes ago, mansr said:

Where did you get that transition bandwidth? If the audible limit is 20 kHz, that gives you 2.05 kHz within which to achieve the necessary attenuation. This is easily done with digital filters.

 

Yes you are right, 2.05kHz if one is aiming for 20kHz. 0.05kHz was a math-typo.

 

I said:

 

"44.1kHz. No one can build a real filter that works for that"

 

Referring to the DAC anti-aliasing filter which has to be analog.

Aj Soundfield claims it can be done, hopefully he'll post a schematic.

[CuteStudio]

56 minutes ago, AJ Soundfield said:

I'll cite the study with all the references

 

Could you just name a CD player/DAC that does it please?

 

56 minutes ago, AJ Soundfield said:

Yours. Speakers, noise measurements and tracks with >16bits dynamic range above your measured noise floor. For the 2nd time.

 

What is your objection to 24bits?

[CuteStudio]

6 minutes ago, AJ Soundfield said:

You claimed 44.1kHz filtering can't be done

 

No I didn't, I said they didn't work: in the context of accurate sound reproduction (this thread)

They all have compromises which is why everyone started over and up sampling.

https://www.stereophile.com/features/106ringing/index.html

 

Whereas on the contrary 96kHz is far easier to build a filter for (and therefore a better sound).

 

I wonder if I misread  the title of the thread, I thought this was about accurate reproduction - if not, what's the point of the thread?

 

 

[CuteStudio]

Just now, AJ Soundfield said:

That's even worse, given that I just gave you a link by "HiRez" proponents no less, showing numerous studies falsifying your claim. That it was cited by Bob Stuart as evidence for the latest reincarnation, borders on the incredible.

http://journal.frontiersin.org/article/10.3389/fpsyg.2017.00093/full

 

 

Yes, another fantasy referencing Oohashis debunked nonsense about brainwaves showing "feeling better" due to inaudible HF (a word used often in the study I linked)

 

Evidence please

 

Thanks for the nice chat AJ, much appreciated.

[CuteStudio]

16 hours ago, jabbr said:

Yeah or upsampled PCM ... boy the quality of Redbook is so good upsampled either to DSD or PCM I think that's at least 90% of the problem with 16/44 ... I can hear differences with higher res files but it's really hard to know how much of that isn't mastering etc as opposed to something intrinsically better about high res files. For example, my favorite Led Zep files are 16/44 -- @bdiament -- now he hears improvements with high res ... but ... doesn't AFAIK, listen using the upsampling software that we do.

 

Surely upsampling involves digital interpolation (estimation)  of the intermediate values between the original data points? Is result then still redbook at all?

 

The process of mastering to 44.1 and then upsampling at the playback end could be described as a form of lossy compression because essentially you are throwing away data to create the 44.1 waveform, and then estimating what those values may have been during the upsampling.

 

The need to do this appears to corroborate my initial (oddly controversial) contention that anti aliasing filters don't work (very well) on the 44.1kHz redbook standard.

 

I'm not sure why we don't just store the music at 96kHz anyway, negating the need for interpolation and upsampling, 44.1 was only chosen due to the technical limitations we had 30 years ago.

 

[CuteStudio]

16 minutes ago, AJ Soundfield said:

I can help you there too

 

 

So AJ, have you found the filter yet?

Just post up the schematic, I don't mind if it's a long URL or a screenshot, but it would be nice if one didn't have to pay $30 to see it behind a paywall. For a 96k or 192k I suspect a basic RC would probably do a reasonable job, I'm not convinced that holds for 44.1, to my knowledge no one has created a perfect analog filter that does that but I'm waiting to be shown how mistaken I am.

 

It's funny but no-one who says "16/44.1 is perfect" ever seems to use it, have you noticed? They use upsampling and dither. Dither of course is a whole different can of worms and while it must be an intrinsic part of the correct downgrading to 16bit, surely if different dither techniques are audible that's a type of logical flaw in the argument that 16bits is enough? So we now have two uncontrolled processes (upsampling and dither) that appear to be band-aids to make 16bits acceptable.

 

So I guess the reality is more like "16/44.1 is perfect as long as you upsample and chose a good dither, and accept the temporal limitations of dithering".

Which may not really be the same thing at all as being perfect I suppose.

[CuteStudio]

9 hours ago, mansr said:

And here's the measured frequency response of that CD player:

 

It's a complex filter, several RC sections, two active parts and a band-pass with 3 inductors: not one for the purists!

Interesting graph, if real! It appears to show a level reduction from -78.5dB to -105dB which appears to be a full 26.5dB cut, which is marvellous.

 

AJ, thanks for the details where I can converse with this chap you've heard of who may have mentioned a transparent (glass, Lexan?) filter for the 44.1 that no one uses, I'm grateful to your references on this but I think that would be giving work to me to prove your assertions, which I don't agree with: so I'm not sure what would be in that for me. Any ideas?

 

It's interesting that as the CD fades into obscurity (This is COMPUTER audiophile, not OLD-CD-PLAYER audio right?) people desperately fighting to defend 16/44.1 become less able to justify why they do it. The filtering is largely solved with a simple up-tick in sample rate to 96 (or even 192) so I can't see the big retro attraction.

 

Additionally 16/44.1 is not the end of the processing chain today either, any digital room E.Q. or level control can only be described as Digital Signal Processing (DSP) which is far better done in 24 bits. Picture a speaker with a 3 way digital crossover feeding some class D chip amps glued to the back of the drivers, the obvious link then is a direct digital feed from your computer/device. 16/44.1 does nothing to help the sound quality here, it is just a tired remnant of a rather poor format that was needed 30 years ago because tech was a lot slower and smaller then. Every DSP devices first action on meeting a 44.1/16 format is then to escape from that format in much the same way it would deal with an MP3, decode it, guess the missing values and then process it.

 

So in a thread about accurate sound reproduction (which should probably include room correction) I find it disturbing that people are still living in the past clutching their obsolete favourite format despite the fact that decent formats are not only attainable but already used widely outside the 'HiFi' audio sector.

 

Crazy.

 

[CuteStudio]

On 6/21/2017 at 0:06 PM, mansr said:

Of course it's real. I measured it myself. The player may be 35 years old, but it's still in perfect working order.

 

Ah excellent, I wondered where it came from, hadn't realised you'd measured it. Cool!!

 

So I noted it seems to have about a full 26.5dB cut, but if I look at the 22.05 position (as close as I can) it looks to be only about 13.25dB down at the point, rather than the required 96dB, can you confirm that the vertical scale is in dB?

 

35 years and still going is good, obviously a quality piece of kit!

[CuteStudio]

On 6/21/2017 at 0:37 AM, AJ Soundfield said:

no one can build a 44.1k filter

 

Hi AJ, yes, that's what I was saying, I'm glad you have finally accepted the limitations of real world analog filters, although that particular one WAS real, I'm impressed with mansr's ability to pop up with a schematic + real measurement. It's an impressive filter but as you noted; it still appears to be a little over 82dB short of the target.

 

If you can post up any evidence to support your earlier assertion that real 44.1k filters are possible that'd be great, TIA. BTW interesting anecdote about the French Suite No. 5 by J. S. Bach.

[CuteStudio]

2 hours ago, AJ Soundfield said:

What target? What does that have to do with a 44.1k filter?

Ah I see, this explains a lot of your earlier posts.

 

Allow me to educate you: In order to avoid aliasing errors the 44.1k filter is required to have a cut of around 96dB between 20k and 22.05kHz

 

Mansr very usefully posted not only a schematic but actual measurements of his D50's filter that showed that it was possibly to get within about 85dB of the target using quite a complex filter, which was interesting.

 

We are waiting for you to produce evidence of an anti-aliasing filter that cuts 20k to 22.05kHz by 96dB to support the outlandish claim that a filter was possible.

 

Quote

hallucinations.

I'm sorry to hear about those, everyone has problems, no need to air them here.

 

Quote

See the fake one Mansr posted.

What makes you think mansr posted anything fake? He's made the best contribution so far!

 

Quote

 

 

[CuteStudio]

19 hours ago, AJ Soundfield said:

this looks close

 

 

 

A 3.25dB cut is a little short of the 96dB required.

 

So Aj you are now reduced to something akin to a robot on Star Trek who Kirk has locked up with a logical dilemma, and just stands there, lights flashing, issuing intelligible gibberish.

 

Perhaps next time you'll think twice before casually dismissing statements that clearly exceed your limited knowledge and grasp of the subject. It's not even a slightly tricky one, it's been known for over 30 years that this anti-aliasing was impossible to build correctly, that's why everyone suddenly started upsampling: duh.