IS EVERYTHING DEBATABLE, REALLY?

[fas42]

While people are hung up on magic words about the technology used in a particular case, then the "debating" will continue, ad infinitum ... ummm, It's the System, Stupid!

 

A bridge collapses, or doesn't, when a heavy load goes across it, not because a particular grade of steel was used, or the foundations were constructed by a certain company, or whether rope stays were used - it's the design, engineering  and construction of the whole that matters - and a seemingly trivial poor decision made about some aspect can be enough to completely undermine the integrity of what the bridge is supposed to handle.

 

Audio just happens to work the same way - everything that's part of it has to be "good enough!" for the reproduction to not be significantly degraded.

 

The good news is, that if the audio "bridge" is properly done then the subjective experience is, well, amazing - and never collapses ...

[fas42]

1 hour ago, gmgraves said:

 

 

That's the real answer. It's all subjective. In a technology where measurements mean less than audible impressions, and where it is understood that no system can possibly sound anywhere near as good as live acoustic instruments playing in a real space, there's many a slip twixt the cup and the lip, and everyone is free to interpret each component's strengths and weaknesses as they see fit. 

 

 

Not quite. It is possible to get a playback system to fool someone, say, that a grand piano is playing - and the illusion works outside with the sound wafting from a window, inside at the other end of the house, at the doorway to the room of the speakers; and finally inside that room, with an acoustically transparent curtain hiding the source of the sound - without altering the volume at any time, for any of these positions.

 

That this is rarely achieved is just an indicator of how the audio industry has been dragging its feet, for decades ...

[fas42]

The only worthwhile listening test is one where one deliberately attempts to make the system playback misbehave - you "stress test" the chain to see if it shows up audible flaws when reproducing certain types of material.

 

When a setup is never caught out doing this, then the competence of the system is confirmed.

[fas42]

1 hour ago, Tony Lauck said:

 

I have done this several times, with the exception of the curtain.   Apart from the different location of the speakers and the piano, no way to tell.  However, in addition to obsessing over playback setup it took quite a bit of work finding the right microphone positioning. 

 

The first time I did this was in the mid 1970's,  later in the 1980's.  Speakers the first time were AR-3a's,  the second time Snell AIIIs.  Two caveats:  the recording and playback were in the same room and the music did not use the bottom three notes on the keyboard.  In both cases a 7.5 IPS 2T Tandberg recorder was used.  In the 1980's I tried the same experiment using a Nak CR-7a instead of the Tandberg and the results were unsuccessful.  Using Dolby destroyed the dynamics and without it there was excessive tape hiss, unless the recording level was jacked up at which case the tape oxide compressed the dynamics.

Interestingly, the entire recording equipment came to under $500 in the mid 1970's.

 

 

Nice one! Yes, it requires everything to be in place - I don't have any sort of recording background, so I am reliant on normally available recordings - which is fine, IME. As you say, the technology decades ago was capable - but you can't just plug a few components together and expect it to happen - by far the most important ingredient is an ability to finesse the setup until it is sufficiently "bug-free".

 

Voice and piano are ideal test material - we all can recognise the qualities of a person vocalising, and pianos are everywhere, the sound of them in the flesh is familiar to most. Which means we are acutely attuned to any sort of giveaway that the sound is being faked.

 

The hardest test is the directly in front of the speakers one, with or without curtain - there is no place to hide for any sort of anomaly sneeking through. For the piano, transient impact is usually the undoing - can a crashing chord be reproduced with the necessary fidelity? Usually not, the 'dynamics' capability of the rig is way under par - and it's all a bit embarassing to listen to, really ...

[fas42]

And the silliness about DBT, etc, continues - it's all pretty simple, in my book: can the the rig fake, say, a piano in the way I've just related? If it can't, then the reproduction is faulty - and so then comparing it it to another faulty system is an exercise in futility.

[fas42]

2 minutes ago, Ralf11 said:

 

 

Do you think any system can fake a piano perfectly?

 

And, for those who cannot or will not pay for such a system, is it still possible for them to get a low fault system?

 

You seem to be equating near perfect and horrible

 

Any system of reasonable quality, that has sufficient headroom with regard to the amplifier combined the particular speakers - you can't combine a SET with ribbons, say, and expect the SPLs to happen - has the potential. As said earlier, the very lowest bass notes would be more difficult to perfectly mimic - but for 99.99% of recorded piano works there wouldn't be a problem.

 

At the moment, it is almost impossible to buy a combo that is good enough in raw form to get such a result - how to transition to it being more common I'm not sure. I suspect it will be a learning process; people will need to understand what areas to focus on, to get the job done. For myself, I have a background such that I've been happy to hack cheap gear to get what I'm after - expensive, bling components being fiddled with would make all concerned nervous, including myself!

 

Not quite sure what you mean by "equating near perfect and horrible" ... if a system can produce the necessary SPLs, but it sounds very sharp and unpleasant to the ears doing this then the signs are good; something that is terribly, terribly polite while playing everything is going to be far harder to "fix".

[fas42]

5 minutes ago, Audiophile Neuroscience said:

 

Well played my friend but Peter is still on record, Blue = Blue.....Evil laughter through the building - no resonances though coz I acoustically treated all the rooms !!

 

And,

 

 

[fas42]

17 minutes ago, Ralf11 said:

 

Thx for clarifying.

But I don't see a mechanism that would alter SQ with a re-save of the file.

 

W/o a mechanism, I'd suggest fairly strong evidence would be needed to get engineers involved.  Say, 19 out of 20 guesses correctly distinguishing the files by several people.

 

A resave could easily alter the situation - the original may not be overwritten, but retained as backup; and the new version saved on a different part of the drive, in a highly fragmented form - or completely unfragmented. This means that the process of accessing the latest version for playing has now altered, which may tip the balance.

[fas42]

22 minutes ago, sandyk said:

An old thread worth a look at by the Newbies of the forum.

It was one of the most popular threads of the time.

 

https://www.computeraudiophile.com/forums/topic/16174-where-is-audio-truth/

 

Takes me back ... a long time ago in a forum far, far away ... I talked about what I was trying to do, in a thread I started, entitled "A Search for Truth and Tonality" ...
 

[fas42]

Just now, beerandmusic said:

One of the most archaic pieces of equipment I ever worked on was an ECMU (extended core memory unit)...it was the size of a refrigerator, and held a whopping 64K of memory.  I also worked on disk drives that were run with hydraulics....i could go on and on...

 

Been there, done that ... did you ever mount a calibration disk, and then with great trepidation manually guide the heads to the right position - and hope to God that the hydraulics didn't take off, because you allowed the assembly to move too fast?

[fas42]

2 hours ago, Tony Lauck said:

 

I took some of these older piano recordings and digitized some of them.  Also some live concert material where there won't be "in the room" credibiilty since there will be venue sonics.  These are on a web site:  http://www.susanlauck.com/ Enjoy these free downloads.  On the studio recordings you may notice that the piano image is excessively large.  This will happen if the physical spacing of hyour speakers is wider than the spacing of the speakers that I used, where the setup had prevously been determined as a compromise with many high quality recordings that I used for playback setup at the time.

 

I presently have a small room in which I have two Focal twin 6 monitors and a single sub woofer.  These are all powered, driven directly from a Mytek Stereo 192-DSD DAC which also serves as a preamp for auditioning analog tapes.  As set up, and playing one of these recordings, a non-audiophile friend spontaneously observed that she had never heard a piano realistically reproduced.  This system will also reproduce Mahler Symphonies at row five live concert levels with adequate headroom, and the monitors come with a safety warning of ear damage, with peak sound level capability rated at 118 dB at my 1 m listening position.

 

When I got these powered speakers they sounded like shit.  It took many hours of adjustment to location, listening position and crossover settings to realize that this wasn't going to work with out adding the sub woofer.  And then when I got this I discovered it was absolutely impossible to get this balanced with the subs until I got a calibrated microphone and measurement software.  Once I did this, I was able to turn more knobs (mostly on the sub) and get good sound, but there was still boominess in some room modes.  I eventually used software parametric equalizer to get flat response in at the listening position from 30 hz up to 1000 kHz.  The regular tweeter adjustment provided a suitable high frequency roll off, and I had previously set this on a mixture of about three dozen recordings of acoustic music of various genres.  Basically, a fairly standard curve that is flat at 1 KHz and down about  -2 dB at 10 Khz did the trick, making the most brilliant recordings listenable (the Mercury Living Presence transfers) while none of these recordings sounding excessively dull.

 

All told, I put several weeks of my time into making this system sound excellent, but no more money once I bought the sub.   The alternative would have been to spend endless time trading equipment and never settling on something that provided realistic playback.   Setup is the most important part of any system, providing that you start with decent gear.

 

 

Thanks, I'll check them out!

 

Active monitors are a good shortcut for getting optimum sound, if they are done well - as will full "digital" speakers, the Kii Three type of thing. No excitement for people who want to play with combinations, but eliminates a whole lot of weaknesses in one go - and of course introduces the potential for others - vibration effects, proximity of all the electronics.

 

The big plus for your Focals is that they can hit the necessary SPLs with no significant audible difficulties - this is key, and why so many setups are not in the race for presenting realism.

 

Particularly interesting is that it required a subwoofer to allow the setup to shine. How I interpret this is that the performance of the Focal's electronics were compromised by trying to make its bass driver do everything - the current peaks required were generating too much interference, the power supplies were not good enough to minimise the impact of the sizeable current swings. Once the load had been taken off this area, by using the separate subwoofer, the Focal's electronics were operating in a far more benign electical environment - and the sound could be rendered cleanly.

[fas42]

1 hour ago, pkane2001 said:

 

It's not enough to say that bits are transmitted using an analog signal -- everyone knows that. Why this should matter to the output of a USB DAC is the real question. 

 

So, why are we even discussing this? What is the effect of this 'analog' waveform on the output of a DAC? Is it the effect of digital noise? Timing errors? Something else? It's not bit flips, I can tell you that. That problem has been solved for much faster data rates than audio, and over much longer cable lengths than 1m USB.

 

And the 'explanation' remains the same ... while the waveforms are treated as representing data, digital stuff, there are no problems, never will be - unless the error checking, parity correcting, etc, fails. But if the endpoint is for us humans to sense what that data means, via an "analogue" receiving mechanism, the ears - then the way to look at the whole shebang flips - it's now totally an analogue world we're considering, including how the "digital data" is being shipped around. Why? Because, high speed electrical signals generate a lot of noise, automatically - it's so easy for a tiny bit of stray capacitance, a minute amount of parasitic ground coupling, to mix in interference with the 'true' analogue side of things - and, degraded sound results.

[fas42]

1 hour ago, sandyk said:

In the capacitance multiplier section of the John Linsley Hood designed PSU add-on that I use there are 2 parallel 4700uF electro's.When the original was designed , Low ESR types were never mentioned. If a Low ESR type is used in parallel with a normal type of the same capacitance, the effective bandwidth of the original design is markedly increased. However,if both types used are Low ESR types, analogue audio sounds too detailed with accentuated treble and sibilance.

You wouldn't normally use very low ESR types as used in computers ,especially not in parallel in Analogue Preamplifiers etc. However , using a selected low ESR type in parallel with a normal type of electro in the JLH normally results in a more balanced SQ as verified by the 100s of people who have used the JLHs in Pre and Power Amplifiers.

It has been shown elsewhere in this thread , that even the waveform saved on a HDD is actually an ANALOGUE waveform with all it's vagaries before being processed.

Alex

 

Just to note, that paralleling capacitors should not be done lightly - there are precise combinations of values and types that should be fine, and others that can get you into trouble. The reasons are purely technical - the parasitic characteristics of the caps' constructions, such as ESR, and how they're mounted can mean there is a resonant impedance - the combo will ring, and this will most likely degrade the SQ.

[fas42]

1 hour ago, sandyk said:

Ceramic capacitors are particularly troublesome in this respect.

I don't use extremely low ESR types specifically designed for Motherboard use .

Neither do I use Panasonic FM types that John Swenson appears to like. I don't have a problem with Panasonic FC, just not 2 FCs in parallel.

 

Hmmm .... I'm not quite sure what the issue with the FM caps would be - in past efforts the FCs have done their job, and I note in the current catalogue that we're up to FS! FC, FM, etc, electros in parallel should be fine - did you find a specific problem with the FM?

[fas42]

38 minutes ago, gmgraves said:

 

One would have to find a solo piano recording where the recording engineer did not put the microphones on a piano extension bar inside the piano before one could try such a test. The way pianos are miked nowadays, they end up, in playback, 10 (or more) feet wide with the treble end of the keyboard coming out of the right speaker and the bass end from the left speaker. First of all, no real piano sounds like a piano miked from inside the piano, and secondly, nobody is going to be fooled by a room-wide piano! Perhaps if one put right and left speakers together, next to one another...

 

https://www.sweetwater.com/store/detail/PianoMicSys

 

I have very recent recordings - classical or otherwise. So, maybe I've been lucky with my piano CDs ... I don't ever sense a room wide keyboard. Which reminds me of an absolutely appalling rendition of the piano in a dealer's showroom, by of all things, a system with the word Steinway in its name - http://www.steinwaylyngdorf.com/en/products/speaker-series/model-d.

 

This was egregious to the highest order - in scale terms, the bass notes were coming from a pipe organ, and the treble from a recorder instrument ...

[fas42]

47 minutes ago, fas42 said:

 

I have very recent recordings - classical or otherwise..

 

Ummm ... I meant to say "I have very few recent recordings - ..."

[fas42]

16 hours ago, sandyk said:

Frank

 The Panasonic FM are VERY low ESR, and wherever I have used them ikn an analogue area they have resulted in a "hardening"  of the sound.

 Neither are they a good choice at the  output of some types of voltage regulators which don't like to see a low ESR capacitor at their output.

 

Alex

 

FM is significantly lower, but it's no deal breaker - or shouldn't be. In a particular use, did you directly try FM compared with FC - same rating, same capacitance, like for like? Can you remember what the particular part was, specifically, at all?

 

FM may use a different chemical cocktail inside, which could change the way the parasitics alter with time and use - these are the type of things that can make all the difference, and it's always handy to get a better understanding of what's going on.

[fas42]

Have never used LM317, my experience is with LM338 devices, and I gave the output an extremely low ESR, using multiple paralleled electro's - no issues. Every circumstance is different - so, something I'll keep in mind ...

[fas42]

2 hours ago, sandyk said:

 

 Hi Frank

 I found the info. It may be of interest to some other members too.

Regards

Alex

 

http://www.tnt-audio.com/clinica/regulators_noise1_e.html

 

Thanks, Alex, I'll give a read ...

[fas42]

Help! Arguing about whether digital is digital is a mighty big dead end - it's just a way of storing data about a musical performance ... it's how to get that data to the ear with minimum corruption on the journey, that matters ...

[fas42]

5 minutes ago, fas42 said:

 

Thanks, Alex, I'll give a read ...

 

Okay, found the key bit,

 

Quote

You won't read about this in these regulators' datasheets, but it has been documented in articles such as Erol Dietz' "Understanding and Reducing Noise Voltage on 3-Terminal Voltage Regulators", Electronic Design (issue unknown) , and Steven Sandler's "Spice Uncovers Regulator-Stability Problems", Personal Engineering, August 1998. From these you would also learn that an LM317's output inductance varies with its output current, and hence that the above noise peaking is load-variant to boot. Just what we needed: a signal-modulated noise generator!

 

I'll see if I can track these down, and what they add to the story ...

[fas42]

SoundAndMotion is saying that a circuit that is normally considered to be digital can also display analogue behaviour; and that parts normally used to perform analogue functions can be connected to behave as a digital component. No more than that. Using the particular electronic parts in that "perverse" way may not be the smartest way to achieve the result - but it demonstrates that there is no inherent difference between the two "modes" - as far as electrons, etc, are concerned.

[fas42]

5 hours ago, beerandmusic said:

why then, the need to create such a thing as a reclocker to feed a dac?  and why with such high praise?

 

Because designers of digital playback gear are very, very slow learners ... while mainstream manufacturers spat out cheap CD players with "fabulous" spec's, and everyone who was fussy in the listening thought that they all sucked, nothing particularly interesting happened. Finally, the general public, and those manufacturers got bored with the exercise - and interest in high quality sound lapsed. Now there is a resurgence of people who genuinely want to create components that deliver true high quality, and it's getting interesting again.

 

For the first time, playback of recordings in the digital domain are demonstrating the potential that was always there ...  a lot of people do struggle with the concept that in some areas one has to be very, very careful with implementation - it's not by the way stuff, it's of make or break importance ...

[fas42]

On 08/11/2017 at 12:20 PM, fas42 said:

 

Okay, found the key bit,

 

<<You won't read about this in these regulators' datasheets, but it has been documented in articles such as Erol Dietz' "Understanding and Reducing Noise Voltage on 3-Terminal Voltage Regulators", Electronic Design (issue unknown) , and Steven Sandler's "Spice Uncovers Regulator-Stability Problems", Personal Engineering, August 1998. From these you would also learn that an LM317's output inductance varies with its output current, and hence that the above noise peaking is load-variant to boot. Just what we needed: a signal-modulated noise generator!>>

 

I'll see if I can track these down, and what they add to the story ...

 

OK, located those articles, and they are not saying anything particularly significant, IMO. The stability problems are due to using a non-optimum bypass capacitor on the adjustment terminal - this area is equivalent to the feedback loop on a conventional amplifier, and as is well known circuits using a poorly chosen capacitor in feedback duties can suffer ringing, instability. Reducing noise voltage on the output is again about poor choice of value of output smoothing cap - too low a value may cause some issues; but just inserting very high value capacitance swamps any tendency to nasty behaviour. This is what I would always do anyway - a sensible choice of part automatically pushes the output impedance at higher frequencies way down.

 

[fas42]

19 minutes ago, gmgraves said:

I still maintain, that if you want to hear either analog or digital where full advantage has been taken of either format's potential, you have to "roll your own". Not even so-called "audiophile labels" have ever produced a CD, SACD, DVD-A or Blu-ray-A release that sounded anything even close to being as good as the master. I must tell you now that a well recorded 16-bit/44.1 KHz master will sound so good that most audiophiles will wonder why anybody would have felt that there was a need for High-Res! It's just that something is lost in the commercialization of master tapes during their journey to the silver disc!

This loss of quality is so profound, that many, after hearing these modern (and expensive) 1-to-1 1/2 track 15ips copies of master tapes from companies like The Tape Project and others, have come away profoundly changed. They mis-perceive what they have heard as proof that analog is better than digital, when what they have actually heard is the difference between the quality captured by the master, and what they got when they bought that same title on either vinyl, some commercial silver disc format or downloaded file. 

 

The CD or whatever release should make no difference, whatsoever. If the data on the medium matches that of the master, assuming the latter is digital, there should be zero loss of quality - if there is perceived to be an audible loss, then the playback chain being used to audition the release is faulty - it's no more complicated than that. Of course, if the master is analogue then losses via the ADC may occur, though these should be very subtle; if playback of an analogue master tape sounds "fabulous" then the chain being used at that moment is the reason for the superior subjective quality.