John Curl's Blowtorch preamplifier part III

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It should be especially evident with headphones.

It may not be so evident unless one thinks to listen for it. Why would anyone think of doing that?

Many things may seem very evident when one knows what to listen for. Why bother? Some people say they avoid S-D dacs because they find them fatiguing and the find they stop listening to music as much, usually quitting a listening session much sooner. They say the still enjoy listening to R-2R dacs for much longer, more like their old listening habits. I don't think we know exactly what is going on with that, as I am not aware of any studies on the topic. As Jakob2 pointed, if a popular food was suddenly found to cloying and consumers started buying other products instead, there would likely be good cause to spend money investigating why. But, people don't buy dacs everyday like they do food, so decisions are few and far between. And it may never occur to people why their listening habits may have changed. If they do notice something to that effect, they may attribute it to other factors such as aging or changing tastes.
 
No illusion of stereo is very obvious.

Width of one instrument? Not the center of its location, just to be sure we are talking about the same thing.

Also, hearing different things in both ears is not the illusion, again just to make sure we agree, it the the perceptual source location between the speakers. With headphones there is always more of that, but speakers are probably a better way of experiencing the perceptual differences between different dacs in that respect.
 
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Regarding my way of designing a HPA or line level amp as outlined here and Not using IC Opamps w/external buffer and hundreds built are all like these comments --> --->

"Excellent sound. Very good depth and width, dynamics are superb and very good layering.

This is with a EX9028 dual dac, Focal Utopia, iBasso SR1 and Pioneer SE-Master1 headphones. Right up there with the best amps! And superb transparency.

and -

It sounds wonderful and effortless. It whispers and then kicks like a horse.

I have posted here several times. Thanks to Jack in NJ, I bought his PCB and matched parts kit.

I have built and tested many headphone amplifiers. This one tests better than all the rest.

Thanks DT

Typical responses to the sound quality and accuracy and independant test results.

The design is now being made for sale in Asia. DIY'ers got first crack at building at lowest cost. Commercially available product wont be cheap.

HPAmp pcb layout.jpg

THx-RNMarsh
 
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This is very good. I'd like to hear it compared to any particular high VFB IC Opamp. However, this is discrete and not an IC Opamp. Which is the way I go also. But certainly not a simple circuit in parts count. But thats OK.

What is the buffer PN? Cant see it on your schematic. And, the model of the buffer in your SIM? It Measured very poorly.


THx-RNMarsh

mk4 uses LME49600 or LME49610 as output buffer.
for complement input transistors could be used matched quad THAT340.
BR Damir
 

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Okay. Well, many dacs do it to some extent or another. With some dacs the effect is so extreme that perceptually the sound seems to emanate from two separate speakers and there is virtually no illusion of stereo even at the sweet spot listening location out in front of the speakers and in the exact center from left to right. It is so common that I asked how we are measuring it, not whether or not it can be measured. I take the latter point for granted that we could find a way to measure it if we were so inclined.
Localisation depends of several factors.
Phase and level coherency all over the dynamic range between the two channels are the basis.
If there is an absolute error of phase and the levels are equal between the two channels, we will feel the sound slightly moved to one side. If we try to compensate-it with the balance (levels) we will have the impression that the source is spreading in width and is no longer punctual.
If there is no delay between the two channels, but the relative levels of the two channels vary with the sound level ( R-2R ‎Accuracy), we will feel the same, but in a different way: like if the source was constantly moving.
So, a measurement could be done with cancelling (phase opposition) the two channels (50%) at various levels.

As always, simple listening is enough to figure out if a system works on this point satisfactorily. On my opinion, headphones are not the best way to figure out it. Because the localization in the middle of our head destabilizes (my personal ;-) brain.
The directivity of the speakers we use and the acoustic treatment of our listening room play an important role, one of the reasons I prefer (good) horns. Group delay too, one of the reason why I tend to prefer 2 Ways speakers to 3 ways (or more ;-).
 
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If we try to compensate-it with the balance (levels) we will have the impression that the source is spreading in width and is no longer punctual.

Thank you. However, it this case there is no attempt to adjust balance or volume level. It is merely a property of a dac when music is playing. Don't know about what might happen in circumstances besides playing music, haven't tried it. I would guess the cause might be something like inter-channel jitter of some type. Just a guess, is all.
 
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Do you think you could be over doing the critical listening and your brain is no longer easily creating the illusion of soundstage?

There is evidence to the contrary. Perceptual point source is very good with DAC-3, maybe an inch or less in perceived width of the illusory source located somewhere between the two speakers (apparent left right location being entirely dependent on where sound happened to be panned when the recording was made).
 
There is evidence to the contrary. Perceptual point source is very good with DAC-3, maybe an inch or less in perceived width of the illusory source located somewhere between the two speakers (apparent left right location being entirely dependent on where sound happened to be panned when the recording was made).
I agree. With a little remark.
People used with artificial head or "ORTF couple" recordings usually pretend localization is better with this kind of recording.
I think it is because both phases and levels differences are matched together in a more satisfactory (natural) way. I tried several times during mix sessions to do this with delay+balance, but it asks a big effort and a lot of concentration to optimize.
So, if localization is easy for all the sources in the middle and the sides, it is less obvious for anything situated between the center and the sides.
Fortunately, this point is not so critical for listening pleasure and nobody complain ;-)
 
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There is evidence to the contrary. Perceptual point source is very good with DAC-3, maybe an inch or less in perceived width of the illusory source located somewhere between the two speakers (apparent left right location being entirely dependent on where sound happened to be panned when the recording was made).

I'd say there is even more evidence to the contrary of your assertion, but we'll never agree and it's not worth debating. All I know is that I can listen to binaural audio on my HD800s from my iPhone XS dongle and the positioning is as it should be and just as spooky as it is with a desktop source and amp.

Scottjoplin's point is very valid, in my opinion.

Those labeled objectivists often get labeled as closed minded. What if there is no spoon, as the Matrix put it ;)? It might truly be the most open-minded position of all. I joined this forum many years ago without much knowledge and looking to design a DAC that included Jupiter beeswax caps, black gates, silver wire, common gate open-loop discrete I/V converter, etc. As my experience has grown, my perspective has shifted. So, for those who dismiss other points of view, just know that I've been there and I've tried R2-R DACs, NON-OS DACs, PMD100 filters, Black Gate caps, discrete Class A HPA with 2SK389 / 2SJ109 input pair. All of these parts that have achieved cult status here.

I like electronics, but I do believe you're better off sinking as much cash as you are willing to part with into speakers / headphones.
 
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So, if localization is easy for all the sources in the middle and the sides, it is less obvious for anything situated between the center and the sides.
Fortunately, this point is not so critical for listening pleasure and nobody complain ;-)
May-be, those defects are what somme appreciate and call "sound stage" ? ;-)

About the "reverb tail" ... It is something that many sound engineer complain with some artificial reverbs. Kind of a noise gate effect. decay curve is not natural enough.
One should be sure of what the source holds before making a judgment on one's own system. And also, maybe, we are missing the old tape hiss, somewhere ?
 
Tournesol, I can make reverb tails more or less audible depending on which dac I choose to use, and according to various configuration settings of my modded dac.

Regarding panning and apparent source width, I would agree that research shows there is an effect dependent on panned position and on frequency. However, I have previously described how I prefer to adjust volume levels when comparing dacs, rather than rigidly sticking to matched levels. In a sort of analogous way, when listening for perceived width of a source I like to turn my head and walk across in front of the speaker a few or several feet away as I turn my head back and forth trying to locate the panned position of a sound and listen for its apparent width. Not being confined with my head in a vise (or confined in a similar relative listening position by headphones), I find that there is an effect on apparent source width that is dac dependent. Depending, I can listen using a good enough dac such that sounds panned neither fully left or right, nor exactly in the center, can sound like they have a very small apparent width regardless of panned position, and that width can be dac dependent.

Obviously, some of the above things are well-known and apparently researched effects. Although I don't know of specific research, I have seen information describing the effects in enough detail to make me think such research probably exists.

For example, an excerpt from the Waves S1 Imager manual: "Shuffling is a stereo enhancement process that has few unwanted side effects, but which can add a sense of spaciousness and life to an otherwise ordinary stereo effect.

Shuffling is particularly suited to recordings made using standard stereo panning techniques made on most stereo mixing consoles or their digital equivalents.

The stereo effect from two loudspeakers creates the illusion in the ears and mind of the listener of illusory phantom images lying between the loudspeakers. This illusion, however, is not perfect, and one known fault is that for sounds panned to intermediate stereo positions, the apparent stereo stage width is narrower at bass frequencies than in the mid and high treble. This is known to be connected with the fact that the ears and brain determine sound direction by different methods below and above around 700 Hz.

Shuffling is a process that widens the bass frequencies so as to make the phantom bass and treble images seem to be of comparable stage width. This generally occurs when the shuffling level is set at around 1.6 and shuffle frequency around 650 Hz. This setting is recommended for sharpest stereo imaging. If you increase the bass width even further, so that the bass is over-wide compared to the treble, it will create less precise stereo images, but generally tends to create an illusion of “spaciousness” and sounds from beyond the two stereo loudspeakers. This helps makes up for the fact that the stereo image is normally confined to the space between the loudspeakers, rather than being all around as in real life. This effect is generally best when the shuffling level is between 2.0 and 2.5, and shuffle frequency around 650 Hz. This setting is recommended for best spaciousness on most material.

Such settings are especially useful on stereo material that otherwise sounds too narrow. Shuffling has no effect on the stereo imaging of central or mono images, but only on sounds panned away from the center. Its effect is not to enhance mono images, but rather to enhance the stereo effect already present in stereo images.

Unlike previous commercially available stereo shufflers, it is fully phase compensated, i.e. does not introduce unwanted phase errors between the stereo channels. This gives improved stereo sound quality with low listening fatigue. It is also carefully designed to minimize tonal alterations of the mix in the bass. The application of shuffling has no significant effect on mono compatibility."
 
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Width of one instrument? Not the center of its location, just to be sure we are talking about the same thing.

Also, hearing different things in both ears is not the illusion, again just to make sure we agree, it the the perceptual source location between the speakers. With headphones there is always more of that, but speakers are probably a better way of experiencing the perceptual differences between different dacs in that respect.
Only a few days ago you said 'what is that' about soundstage! Now I am very sensitive to placement and get annoyed with fat blobby instruments and 10ft wide vocalists. I listen to a lot of mono recordings so have a comparison. I also consider pan potting to produce a fake soundstage which is painfully audible as such. I also put store on the illusion of depth. Good classical records can fool the mind. But recording, room and speakers are 90% of this.

Easy enough to insert phase shifts into one channel to check things. That's worthy of some research.

wrt stereo shuffling this is a good read https://www.audiosignal.co.uk/Resources/Stereo_shuffling_A4.pdf
 
What about amplitude domain?
Transient settling domain (or whatever you want to call it)?
How do you measure what it is that makes one interpolation filter sound different from another? Do it with illegal signals that make them pre-ring? If not that then what?
Which time domain or frequency domain test do you use to measure the perceptual width of a sound panned to center in a multitone, multi-panned signal enviornment? What measurement shows whether a sound panned to middle is perceived as emanating from an approximate point source or a widely spread out elliptical source?
Check M/S vs L/R encoding (and un-matrixing at power amp input). If imaging changes (more stable etc with M/S), you have the dynamic channel imbalance you're speaking of. This is a very old and simple trick to enhance the imaging of lesser DACs but only few people seem to know it.
 
So, what is going between measurements and listening then? I will qualify my response right off as IMHO. For one thing we still don't know how to correlate measurements very well with aural perceptions as processed in the brains of different people. It could also be that we aren't measuring all the things we could be. S-D dacs are inherently very non-linear and I don't mean that in the sense of something like a power amp that may have a fairly stationary, but somewhat curved voltage transfer function. S-D dacs can be non-linear like fairly bizzare DSP algorithms. Measuring with steady tones doesn't reveal anything about how an S-D modulator reacts to dynamics.
But the disturbances are repeatable, aren't they? How long is the state memory of a typical D-S DAC? Not infinite and short (< seconds) in general, correct? So with differential techniques and heavy block-averaging it should be almost trivial to isolate the disturbance imposed on the D-S output after a "shock exitation" that isn't caused by simple analog settling. The analog settling characteristic is not a function of the level other than simple scaling, whereas the additional disturbance from that bizarre dynamic D-S error pattern might be different vs input level (different input bit stream)?
 
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