In Search Of Accurate Sound Reproduction: The Final Word!

[gmgraves]

2 hours ago, fas42 said:

 

Accuracy to what's encoded on your copy of the recording. That's "locked in concrete" and IME the better the playback the more positive the impact 'artificial' music has - in particular, the spatial enhancements added to sounds can be huge, and if this information is not "perfectly" reproduced then the overall tends to sound cluttered, messy, congested.

 

Yes, the system needs to "sound good" playing this sort of stuff - and I gave up using my test CDs of this type of music when trying out unknown rigs a long time ago  - they generally made a mess of them, delivering a tiny, unpleasant, "wall of sound" - "what on earth were the musicians trying to achieve?" is the thought one would have. It takes 'correct' playback to allow one to understand what's going on ... and sometimes  they become your most treasured recordings.

 

 

That's pretty ridiculous if you think about it. How do you, as a listener know what's encoded on your copy of the recording? You can't hear it until you play it back, and then it sorta is what it is. You have no way of knowing what's actually encoded on the recording; just your system's version of what's there.

 

I agree that one needs "correct playback" to allow one to understand what's going on. My point is that unless the music you are trying to play back correctly actually exists in real space, you have no way of knowing what "correct playback" sounds like. Since people who listen to a lot of live concert music, they ostensibly know what it sounds like; they can hear a recording of a piece of concert music reproduced via recording (or live broadcast) and say: "Wait a minute, Those strings sound awful. They are way too bright. Real strings don't sound like that in a concert hall!" Or sometimes: "I've been in that hall and heard that orchestra. That hall doesn't have anywhere near as little bass as I'm hearing on my system. I need better subwoofers." 

 

How do you make those comparisons with music that doesn't exist outside of the studio? You can't. You don't know. If a recording sounds too bright, it might be that your system is erring toward the bright side and is thus not accurate, or, the band to whom you're listening wanted their performance to be what would be considered in the classical or jazz world "too bright".  But either way, you'll never know because there's no way for you to compare what the recording sounded like to the musicians, producers and engineers who made it and what you are hearing at home. Therefore the accuracy of your playback equipment is a moot point. You buy what sounds good to you when playing the music you like. If you don't ever listen to Beethoven or 'Dave Brubeck Live' why should you care that your systems sounds wrong when playing that kind of music? You listen to rock and/or Hip-hop, and your system sounds great playing those. What else would you be interested in?

[fas42]

22 minutes ago, gmgraves said:

 

That's pretty ridiculous if you think about it. How do you, as a listener know what's encoded on your copy of the recording? You can't hear it until you play it back, and then it sorta is what it is. You have no way of knowing what's actually encoded on the recording; just your system's version of what's there.

 

...

 

Therefore the accuracy of your playback equipment is a moot point. You buy what sounds good to you when playing the music you like. If you don't ever listen to Beethoven or 'Dave Brubeck Live' why should you care that your systems sounds wrong when playing that kind of music? You listen to rock and/or Hip-hop, and your system sounds great playing those. What else would you be interested in?

 

What you develop, over a period of time, is a knowledge what's there, because it always sounds the same - different systems, times of listening, all the usual variables; the signature of that recording becomes more and more clear, you build up a full understanding of the content, to the lowest levels. So, a competent system that you have never heard before should sound like that; if it doesn't, and worse, discards a high percentage of the detail thereon - am I likely to think I'm getting "closer to the recording"?

 

I make a point of listening to every type of recording - absolutely everything; no matter how bizarre, how "awful". Because, it may tell me something about what my system can't handle, reproduce well.

 

Most "rock" systems are excrutiatingly bad ... they "enhance" the sound in ridiculous ways, and are miles away from what the recording actually has. Typically, they try and turn what is obviously a studio creation into the "Listening to a PA at a concert" thing - well, if that's what turns them on ...

[Daudio]

27 minutes ago, gmgraves said:

My point is that unless the music you are trying to play back correctly actually exists in real space, you have no way of knowing what "correct playback" sounds like

 

Heartily agree !

 

28 minutes ago, gmgraves said:

How do you make those comparisons with music that doesn't exist outside of the studio? You can't.

 

Again agreement, but that wasn't quite what I read him saying (really skimming).

 

My response was "Yeah, that sounds familiar", in that I find that many of the mixing board, studio, and even electronic recordings (that never saw the light of air) sounded better and better, as my system was 'calibrated' (rhetorically) via well recorded, live acoustic music, in a good venue, compared against what I have (intentionally) heard live. (well... plus my decades old list of 'demo' cuts  )

 

Even through Classical is not my main genre, it serves double duty in my Library with some of the best recordings to measure system changes on SQ. The music lends itself to this task, by giving us about all the sonic elements we'd like to experience and compare (tonal, dynamics, freq. extremes, complexity, localization's, room acoustics, etc.), and it's not at all difficult, or unpleasant, to expose your ears to the real thing so you know what you're doing back in your audio room 

 

As the classical (and a limited number of other live stuff) recordings sound more like the reality I know (however imperfectly), most of my other music genres (Electronica !) sound better, more interesting and involving then before, making me a happy audiophile

 

[gmgraves]

5 hours ago, fas42 said:

Most "rock" systems are excrutiatingly bad ... they "enhance" the sound in ridiculous ways, and are miles away from what the recording actually has. Typically, they try and turn what is obviously a studio creation into the "Listening to a PA at a concert" thing - well, if that's what turns them on ...

 

But, those systems sound good to those rockers who own them, don't they? The fact that they sound terrible to those whose taste in music is more, shall we say, eclectic, doesn't mean that these rockers would or even should care that their system sounds terrible on other than their preferred types of music. I rest my case.

[semente]

7 hours ago, gmgraves said:

 

That's pretty ridiculous if you think about it. How do you, as a listener know what's encoded on your copy of the recording? You can't hear it until you play it back, and then it sorta is what it is. You have no way of knowing what's actually encoded on the recording; just your system's version of what's there.

 

I agree that one needs "correct playback" to allow one to understand what's going on. My point is that unless the music you are trying to play back correctly actually exists in real space, you have no way of knowing what "correct playback" sounds like. Since people who listen to a lot of live concert music, they ostensibly know what it sounds like; they can hear a recording of a piece of concert music reproduced via recording (or live broadcast) and say: "Wait a minute, Those strings sound awful. They are way too bright. Real strings don't sound like that in a concert hall!" Or sometimes: "I've been in that hall and heard that orchestra. That hall doesn't have anywhere near as little bass as I'm hearing on my system. I need better subwoofers." 

 

How do you make those comparisons with music that doesn't exist outside of the studio? You can't. You don't know. If a recording sounds too bright, it might be that your system is erring toward the bright side and is thus not accurate, or, the band to whom you're listening wanted their performance to be what would be considered in the classical or jazz world "too bright".  But either way, you'll never know because there's no way for you to compare what the recording sounded like to the musicians, producers and engineers who made it and what you are hearing at home. Therefore the accuracy of your playback equipment is a moot point. You buy what sounds good to you when playing the music you like. If you don't ever listen to Beethoven or 'Dave Brubeck Live' why should you care that your systems sounds wrong when playing that kind of music? You listen to rock and/or Hip-hop, and your system sounds great playing those. What else would you be interested in?

 

In other words you seem to imply that one cannot assess accuracy to the recorded signal by listening unless we use live unamplified acoustic music adequately recorded in a natural resonant space.

I agree.

[semente]

You also need a database of listening experiences of both live and reproduced music so that you know what to listen for as well as what can and cannot be achieve in domestic playback.

[fas42]

1 hour ago, semente said:

You also need a database of listening experiences of both live and reproduced music so that you know what to listen for as well as what can and cannot be achieve in domestic playback.

 

Yes, I seize every opportunity to sidle up to buskers, street celebrations, marching bands and the like, where there is not a whisper of a sound reinforcement device to be heard - the "bite and jump" of raw instruments is the goal.

 

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.

 

[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!

 

 

[gmgraves]

10 hours ago, semente said:

 

In other words you seem to imply that one cannot assess accuracy to the recorded signal by listening unless we use live unamplified acoustic music adequately recorded in a natural resonant space.

I agree.

 

I'm not actually thinking about the recorded signal, I'm simply saying that High-Fidelity, meaning a great degree of accuracy to the original performance, as a concept, is irrelevant when the listener has no idea what that performance should or actually does sound like. This is principally due to the fact that electronic music as produced by electric guitars, electronic keyboards, contact miked saxes, trumpets, trombones, whatever, have no sound. They are fed to a mixing console and other electronic effects generators where they are further altered, and don't become sound until they pass through an amplifier and emerge through some kind of reproducing transducers such as speakers or headphones. Due to this fact of life, people who listen to this kind of music exclusively have no need for the type of equipment neutrality and accuracy (to the original sound) that spawned the High-Fidelity movement in the first place. They just need a system that sounds good to them, playing the music that they like. There's nothing right or wrong about that feedback-less approach to music listening, it's just not High-Fidelity. 

[fas42]

4 hours ago, CuteStudio said:

 

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!

 

 

 

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.

[audiventory]

I suppose, that almost 100% fidelity will achieved after implementation of 3 conditions only:

 

1. Capturing by way that can be transformed to perception of each listener.

 

2. Playback directly to brain of the listener.

 

3. Music stored and transfered without distortions.

 

 

Also here matter of threshold of perception is important. The matter is researched and possibly we will get new information with time.

We are a long way away from that aim yet.

[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.

[AJ Soundfield]

6 hours ago, audiventory said:

We are a long way away from that aim yet.

"We" who? 

There is this insular "we" https://www.stereophile.com/asweseeit/1107awsi/

Then there is the other "we' who might not be quite as "far away" as thought.

http://www.linkwitzlab.com/Recording/acoustics-hearing.htm

http://www.aes.org/e-lib/browse.cfm?elib=9136

"Scary" results when the other "we" venture outside the bubble http://www.onhifi.com/features/20010615.htm  (please note dates too)

[mansr]

7 hours ago, CuteStudio said:

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.

What is this matrix you speak of?

7 hours ago, CuteStudio said:

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 rate converter is a digital low-pass filter. It is true that the sharp filter required for correct reproduction of 44.1 kHz material is more easily implemented digitally than with analogue components. That is why virtually all DACs upsample the input to 384 kHz or more.

7 hours ago, CuteStudio said:

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.

Do you accept that human hearing has a limit? Do you agree that once enough bits are used to exceed this limit, there is no point in going further? It appears that somewhere between 16 and 20 bits is sufficient to capture anything a human could possibly hear. The audible difference from adding another bit becomes increasingly subtle with each one, so going from 16 to 24 bits will be barely noticeable while the difference between 8-bit and 16-bit resolution is readily apparent. In fact, even 16 bits is plenty for most music in normal listening conditions. You'd need a very, very quiet room to have any real benefit from more.

7 hours ago, CuteStudio said:

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. 

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

7 hours ago, CuteStudio said:

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.

The magnitude of the distortion products from undithered quantisation depends on the bit depth, not on the amplitude of the signal. The distortion level at -60 dB signal is the same as full scale. The signal must of course be higher than the quantisation level. Anything lower is lost completely. With dither, even signals below the quantisation level are captured although at some point they are completely drowned out by the dither noise. In other words, the bit depth determines the usable dynamic range, and anything within this range is conveyed equally well. Dithered 8-bit actually has enough dynamic range that typical pop or rock music hardly suffers at all. With classical music, the dither noise becomes audible during relatively quiet parts.

7 hours ago, CuteStudio said:

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.

Speakers are indeed the least accurate component in an audio playback system. There we agree.

 

[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.

 

[AJ Soundfield]

11 hours ago, CuteStudio said:

There are two major issues with redbook:

441.kHz. No one can build a real filter that works for that,

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.

If you have reliable data contradicting decades of study, please present it (sorry, "I heard it, I said so" doesn't qualify). Thanks.

 

Quote

Which brings me to the terrible 16 bits. If 16 bits is Ok why is 8 bits bad?

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

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

[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?

 

 

[mansr]

18 minutes ago, CuteStudio said:

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.

I see what you're getting at, but that's not an accurate view of what's actually going on.

18 minutes ago, CuteStudio said:

" 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?

Recordings should absolutely be done in 24-bit. There are a multitude of benefits to that. Distribution may or may not benefit from a resolution beyond 16 bits. Then again, absent any constraints of a physical medium (e.g. CD), there's no practical reason to not use the full 24 bits.

18 minutes ago, CuteStudio said:

"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.

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

18 minutes ago, CuteStudio said:

"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%

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

[AJ Soundfield]

5 minutes ago, CuteStudio said:

 

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

I'll cite the study with all the references

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

Now the goalposts shift back to you and you get to cite evidence "441.kHz. No one can build a real filter that works for that" which would contradict the decades of studies cited in the article. Evidence please.

 

Quote

When you say 'consumer playback systems, do you mean

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

[mansr]

57 minutes ago, CuteStudio said:

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

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.

[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?

[Jud]

2 minutes ago, CuteStudio said:

 

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.

 

I really do wish people would discuss factual, relevant stuff, rather than having arguments between the non-factual (yes, it’s possible to build a final analog reconstruction filter for 44.1kHz) and the irrelevant (almost no one builds NOS DACs intended for 44.1kHz input any more).

[AJ Soundfield]

8 minutes ago, CuteStudio said:

 

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

Does what? You claimed 44.1kHz filtering can't be done, which means no Redbook.

Here, I'll remind you: 

Quote

 

There are two major issues with redbook:

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

 

Evidence please.

 

Quote

What is your objection to 24bits?

Ok, so no speakers, no noise measurements and no tracks.

So zero evidence against 16 bit consumer playback. Thanks, it was all rhetorical.