Cost no object dac/streamer

[bennetng]

Yes, but accurate to what, that signal has passed through a ton of equalizers and compressors already, by some mastering engineer, through his speakers, good for his hearing.
Some guys here talk like we are discussing for pacemaker devices, these things should reproduce music.
Have anyone here listened to live music? No dac or amplifier can reproduce it,
Accurate yes, but first is the listening..

You are comparing an human subject (mastering engineer) with an electronic device (DAC, CD player, streamer etc).

The mixing and mastering process require a mastering engineer to operate the effect processors and mixing parameters. Different music will use different parameters according to their own tastes.

An electronic playback device on the other hand doesn't understand music style and genre, it just process the data or signal without emotion, indiscriminately.

At least some sort of crude AI is required to make a playback device react to "music". AI is actually being used in some audio plugins, but still require a human subject to judge the quality and make adjustments. Example:

https://techblog.izotope.com/2017/0...lavalier-microphone-noise-with-deep-learning/

Therefore transparency or manual DSP adjustment is the way to go, at least until AI can analyze the taste of individual users and applying effects to playback material adaptively.

 

[Frank Dernie]

Interestingly the two units that ran through the attenuator were the Aune and the Merlot (not quite sure about the latter) because of fixed volume output. The others were adjusted using the DAC's volume. So for them the signal path was through a physical switch / cable directly to the brinkmann amps.
If output impedance was indeed a factor for the audible difference, then the high output impedance of some of these DACs would simply fall under the heading of bad design. The dCS seems to be good design (at least with respect to this parameter). Lucky I liked that one and not say the T+A, else I my credibility (if it exists at all) would take a serious hit.


Most modern DACs have switchable filters so I would suspect most of us have used one that has them. In our comparison we left all DACs at the default filters (which yes, is probably a fundamental flaw of the comparison).
But then comes the nasty question. Which of the filters (which are all compromises) is the most accurate?

I have a dCS P8i. I don't use it any more since my Devialet amp has a digital input.
The different reconstruction filters make a difference to the sound. When I compared DACs a few years ago they all used the same type of reconstruction filter and I heard no difference between them. If the DACs were not all using the same filter I would not expect them to sound the same.
I don't remember which was the dCS default. I preferred the sound with upsampling off though, it was too bright with upsampling.
When I tested a multitude of DACs, from the Linn Klimax DS down they all sounded the same to me. None of the ones I compared had a choice of reconstruction filter though.
The normal filter is the least compromised (unless you listen to a half cycle sample at 22.05kHz which is a frequently used test showing ringing) which is both amusical (actually impossible to exist as a sound) and only theoretically possible because in any real engineering situation it actually in the stop band, so it is what I would call a provocative test...

 

[jacobacci]

Thanks to @andreasmaaan for digging up the output impedances of the DACs. I have sent an email to Goldpoint to enquire about the behavior of the SA2X-I when set to "no attenuation". I will report back when I hear something.
For me this thread has been very interesting, because it has raised a number of fundamental issues, as I was trying to get to the bottom of why the issue of "I hear differences" and "no you don't" is such a contentious one.
One issue seems to be the question of scope of experiment. When a DAC is being measured with an analyzer these two pieces of equipment comprise the scope of the experiment. The DAC's output is connected to the high impedance input of the analyzer using a relatively short cable with limited capacitance. Unfortunately an analyzer does not have ears. So when it comes to listening to music, the scope of the experiment changes fundamentally. We are now dealing with a complete hifi chain, a room, ears, a nervous system and a conscious person interpreting the stimuli and deducing conclusions. While I find the measurements conducted in this forum extremely valuable (I really do), I am even more interested in the transferability of the conclusions drawn from the measurements to a live listening environment. This is not a trivial process, as described here.
In a hifi setup similar conditions to the analyzer experiment may apply, if the DAC is connected to a preamp with high input impedance using similarly short cables. If the DAC is connected to a monoblock amp using long cables, things become fundamentally different, especially if the DAC should have a high output impedance such as Chord DAVE in single ended mode. Even more so if a passive attenuator is used in the path.
So "all DACs that measure well sound the same" seems to be a statement that holds true only under certain conditions. While the behaviour of a DAC connected to an analyzer seems to be quite well defined, the picture becomes less clear as the DAC is introduced into a complete system and starts to interact with cables, inputs, power supplies etc. The concept of impedance architecture has certainly hit home with me and I will look at my friend's and my setup regarding the influence of impedance. I also connect the dCS Rossini directly to two Neukomm PA18 Monos with around 5m of Prefer PMK-206 cable. Luckily I don't need a passive attenuator, not even the attenuator of the dCS. I can run the dCS at full volume and adjust the volume directly with the Neukomm Monos' built in attenuation circuit. So one less element to worry about.
A valuable piece of information from my perspective would be a set of system architecture recommendations, under which the premise of "all DACs that measure well sound the same" actually holds true. In these conditions the conclusions from the measurements could then actually be transferred. Maybe one could also measure the DACs using different lengths of cables to determine if the output impedance of the DAC interacts in a significant way with the cabling. I was asked in one of the posts to name one or two additional measures for the test sequence. This could be one of them.

The second area that went through my mind was the one of digital filters. Most DACs these days contain several digital filters. They measure different, so they will likely also sound different. All of these filters are compromises, tuned to optimal frequency, phase or transient reponse or a mixture of these and other parameters. None of them is accurate, as no absolutely accurate transfer function exists when transferring a bandwidth limited digital signal to analogue.
Studies show that in order to achieve spatial sound localisation resolution humans (and other mammals) can achieve, a temporal detection accuracy of a few microseconds is necessary (this is apparently independent of our frequency limited hearing ability). In my view this begs the question, how different digital filter characteristics influence our ability to detect say 3 dimensional space in a recording. In our very subjective DAC comparisons the easiest way for me to differentiate was to judge the proportions of the 3 dimensional space of the recording. One explanation might be that the filters of the two DACs had differing phase errors across the audible spectrum and hence different ability to accurately provide the timing of spatial cues from which to derive a sense of space. Some DACs generated a deeper and more clearly defined 3 dimensional reproduction of opera singers moving about on stage. Call this audiophool nonsense if you wish. Anyway, I think it might be worth to look at phase errors across the audible spectrum as an additional measurement. I would certainly be interested to hear other people's thoughts about this. @Frank Dernie thanks for your comments re. filters above.
So, has this thread gone "down the gutter" - for me definitely not. Quite the contrary.

 

[jacobacci]

Btw, do you have a link for the speakers?

Sorry, only just saw your earlier post. No link for the speakers, but I have attached what I have. Hope you can read them.
See me above post for theories.
More advanced reconstruction filters would likely be the cause of the difference between Delius / Purcell and Rossini (close to 20 years of Moore's law).

 

[andreasmaaan]

So "all DACs that measure well sound the same" seems to be a statement that holds true only under certain conditions.

I think you're right, and maybe people who put forward this position need to attach more disclaimers.

It should be kept in mind though that your experiment took place under very unusual conditions. The amps in that system had an input impedance way below just about any other amp on the market, and I'm really not sure why that is. It's quite rare for power amps to present less than a 10 KOhm load, compared to 600 Ohm in the case of those Briknmann monoblocks. An enigma..

Maybe one could also measure the DACs using different lengths of cables to determine if the output impedance of the DAC interacts in a significant way with the cabling.

As per above, this will not be a significant factor if the amps have more typical loads than the Brinkmann. However, it would be good to measure what the output impedance of DACs are IMO, if for no other reason than to allow people to judge whether any problems might be foreseeable in whatever potentially unorthodox setups they intend to use them in.

Studies show that in order to achieve spatial sound localisation resolution humans (and other mammals) can achieve, a temporal detection accuracy of a few microseconds is necessary (this is apparently independent of our frequency limited hearing ability).

This is true, and one of the reasons that studios ensure that all digital devices are synced to the same word clock. I also agree that it would be interesting to see whether DACs create interchannel timing errors. Perhaps someone who knows more about DAC architecture could comment on whether significant errors in this regard are a real possibility?

However, I'm not aware of this ITD threshold being non-frequency dependent, however. Do you have sources you're basing that part of the statement on?

My understanding of the experimental data is that our are more sensitive below about 1500Hz. The typical detection threshold is given as about 30-50us in this most-sensitive range (source).

By my reckoning, 30us corresponds to about three samples at 96KHz, or about 1.5 samples at 44.1KHz, i.e. a very large amount in comparison to the levels of precision of the clocks in most DACs.

Most DACs these days contain several digital filters. They measure different, so they will likely also sound different.

I'm not sure I agree with that last sentence. Although of course what happens in terms of frequency and phase between the transition band and Nyquist is very different from filter to filter, plenty of filters (all good filters IMO) maintain flat phase and frequency response up to 20KHz, particularly when the sample rate is higher than 44.1KHz. Differences beyond the limits of human hearing, although present in the measurements, are by definition irrelevant IMHO.

So, has this thread gone "down the gutter" - for me definitely not. Quite the contrary.

I agree. It's an interesting discussion, thanks.

 

[jacobacci]

It should be kept in mind though that your experiment took place under very unusual conditions.

Absolutely agree. And I will need to take this up with my friend.

By my reckoning, 30us corresponds to about three samples at 96KHz, or about 1.5 samples at 44.1KHz, i.e. a very large amount in comparison to the levels of precision of the clocks in most DACs.

My reasoning was not concerning the clocks at all. It goes roughly as follows:

• Spatial resolution as determined in experiments requires a temporal resolution of some 10 microseconds
• A phase error in the reconstruction filter of 180 degrees at 15 kHz is a temporal error of around 30 microseconds
• Phase errors could possibly "distort" the spatial reconstruction re the original, because the spatial cues do not arrive at the same time as in the original signal

I may be completely off with this, but to me it seems like something worth investigating.
I really have no other theory why the spatial representation of different DACs is different, because spatial localisation is primarily about the timing of cues. And with a sufficiently resolving system in my experience they definitely are different. Now whether two DACs exhibiting two different spatial representations are both "accurate" is a completely different story.

 

[andreasmaaan]

Absolutely agree. And I will need to take this up with my friend.


My reasoning was not concerning the clocks at all. It goes roughly as follows:

• Spatial resolution as determined in experiments requires a temporal resolution of some 10 microseconds
• A phase error in the reconstruction filter of 180 degrees at 15 kHz is a temporal error of around 30 microseconds
• Phase errors could possibly "distort" the spatial reconstruction re the original, because the spatial cues do not arrive at the same time as in the original signal

I may be completely off with this, but to me it seems like something worth investigating.
I really have no other theory why the spatial representation of different DACs is different, because spatial localisation is primarily about the timing of cues. And with a sufficiently resolving system in my experience they definitely are different. Now whether two DACs exhibiting two different spatial representations are both "accurate" is a completely different story.

The thing is, that 10us figure is for interaural time differences. A phase error of 180° at 15KHz that affects both channels equally has never been detected under controlled conditions (not even close in fact).

That's my take on it, anyway.

 

[Sal1950]

I think you're right, and maybe people who put forward this position need to attach more disclaimers.

I don't think many objectivist put forward that position without some disclaimers either stated or implied. It's always been the subjective community that has tried to pigeonhole us as to having positions that "all amps, dacs, cables, etc" sound the same, PERIOD. Common sense would tell most people that there is always the possibility of variables at work in any situation. It's simply the fact that most properly designed modern electronics are at the level of being fully transparent and there is no magic dust out there. If a device sounds different than the next when inserted in system, the reason can be revealed by using modern measurement tools and knowledge.

 

[Blumlein 88]

Thanks to @andreasmaaan for digging up the output impedances of the DACs. I have sent an email to Goldpoint to enquire about the behavior of the SA2X-I when set to "no attenuation". I will report back when I hear something.

Probably at no attenuation the signal is straight thru with a high resistance to ground. So it would be fairly close to having no effect. If you have to turn it down at all then some resistance will be in series and could effect frequency response with longer cables etc. Using low impedance amps with passive control is really just a bad idea. The best way to use passive volume control is as close to the amp input as possible with as short a cable as possible.

For me this thread has been very interesting, because it has raised a number of fundamental issues, as I was trying to get to the bottom of why the issue of "I hear differences" and "no you don't" is such a contentious one.
One issue seems to be the question of scope of experiment. When a DAC is being measured with an analyzer these two pieces of equipment comprise the scope of the experiment. The DAC's output is connected to the high impedance input of the analyzer using a relatively short cable with limited capacitance. Unfortunately an analyzer does not have ears. So when it comes to listening to music, the scope of the experiment changes fundamentally. We are now dealing with a complete hifi chain, a room, ears, a nervous system and a conscious person interpreting the stimuli and deducing conclusions. While I find the measurements conducted in this forum extremely valuable (I really do), I am even more interested in the transferability of the conclusions drawn from the measurements to a live listening environment. This is not a trivial process, as described here.
In a hifi setup similar conditions to the analyzer experiment may apply, if the DAC is connected to a preamp with high input impedance using similarly short cables. If the DAC is connected to a monoblock amp using long cables, things become fundamentally different, especially if the DAC should have a high output impedance such as Chord DAVE in single ended mode. Even more so if a passive attenuator is used in the path.
So "all DACs that measure well sound the same" seems to be a statement that holds true only under certain conditions. While the behaviour of a DAC connected to an analyzer seems to be quite well defined, the picture becomes less clear as the DAC is introduced into a complete system and starts to interact with cables, inputs, power supplies etc.

As already noted the system you used was unusual. And it is necessary to meet certain conditions. For most systems this isn't a big issue. If you have a DAC output impedance of 100 ohms or less (not uncommon), even a few meters of cables (as long as they aren't some odd audiophile design with crazy high capacitance), and an input impedance to the amp of 20 kohm or more saying DACs that are linear and low distortion will sound the same should be what one finds in practice.

The concept of impedance architecture has certainly hit home with me and I will look at my friend's and my setup regarding the influence of impedance. I also connect the dCS Rossini directly to two Neukomm PA18 Monos with around 5m of Prefer PMK-206 cable. Luckily I don't need a passive attenuator, not even the attenuator of the dCS. I can run the dCS at full volume and adjust the volume directly with the Neukomm Monos' built in attenuation circuit. So one less element to worry about.
A valuable piece of information from my perspective would be a set of system architecture recommendations, under which the premise of "all DACs that measure well sound the same" actually holds true. In these conditions the conclusions from the measurements could then actually be transferred. Maybe one could also measure the DACs using different lengths of cables to determine if the output impedance of the DAC interacts in a significant way with the cabling. I was asked in one of the posts to name one or two additional measures for the test sequence. This could be one of them.

The second area that went through my mind was the one of digital filters. Most DACs these days contain several digital filters. They measure different, so they will likely also sound different. All of these filters are compromises, tuned to optimal frequency, phase or transient reponse or a mixture of these and other parameters. None of them is accurate, as no absolutely accurate transfer function exists when transferring a bandwidth limited digital signal to analogue.

A proper filter (which will have the dreaded and immaterial ringing that has been made a bugaboo about nothing) can have good flat response. Many others may allow low levels of aliasing, have drooping treble response or have a rippled up and down response in the upper octaves. These kinds of things can be audible.

Studies show that in order to achieve spatial sound localisation resolution humans (and other mammals) can achieve, a temporal detection accuracy of a few microseconds is necessary (this is apparently independent of our frequency limited hearing ability). In my view this begs the question, how different digital filter characteristics influence our ability to detect say 3 dimensional space in a recording. In our very subjective DAC comparisons the easiest way for me to differentiate was to judge the proportions of the 3 dimensional space of the recording. One explanation might be that the filters of the two DACs had differing phase errors across the audible spectrum and hence different ability to accurately provide the timing of spatial cues from which to derive a sense of space. Some DACs generated a deeper and more clearly defined 3 dimensional reproduction of opera singers moving about on stage. Call this audiophool nonsense if you wish. Anyway, I think it might be worth to look at phase errors across the audible spectrum as an additional measurement. I would certainly be interested to hear other people's thoughts about this. @Frank Dernie thanks for your comments re. filters above.
So, has this thread gone "down the gutter" - for me definitely not. Quite the contrary.

The available work into the area indicates above 1500 hz or so we don't much care about phase. And there has been a number of experiments over the years. Frequency response ripples and very small differences over an octave or more we are rather sensitive to however. Our ears might care about inter-channel phase differences, but most DACs have that under control to something like .5 degree phase difference or less between channels. You'd be surprised how plain old frequency response can change depth and sense of space even with rather small differences.

Interchannel timing is no issue. Digital audio can manage that within some low picosecond level. It is not limited to the length of time between samples.

So your listening tests are interesting. It may not be the total answer, but using a passive control with longer cables to monoblocks with unusually low input impedance is something of red flag that those conditions may be responsible for hearing differences. Or it could be something else.

 

[Blumlein 88]

I don't think many objectivist put forward that position without some disclaimers either stated or implied. It's always been the subjective community that has tried to pigeonhole us as to having positions that "all amps, dacs, cables, etc" sound the same, PERIOD. Common sense would tell most people that there is always the possibility of variables at work in any situation. It's simply the fact that most properly designed modern electronics are at the level of being fully transparent and there is no magic dust out there. If a device sounds different than the next when inserted in system, the reason can be revealed by using modern measurement tools and knowledge.

As you no doubt know Sal, on another forum when I say such things someone wants a list of properly designed gear. And if I give it to them they have other complaints. It is also an issue because some boutique audiophile gear is what departs from a normal regular design with normal parameters while the rest of the world more or less has this figured out.

 

[jacobacci]

The available work into the area indicates above 1500 hz or so we don't much care about phase. And there has been a number of experiments over the years. Frequency response ripples and very small differences over an octave or more we are rather sensitive to however. Our ears might care about inter-channel phase differences, but most DACs have that under control to something like .5 degree phase difference or less between channels. You'd be surprised how plain old frequency response can change depth and sense of space even with rather small differences.

Thanks, @Blumlein 88 you have just restored my peace of mind with an explanation that makes total sense.

The setup was an outlier, for sure and the impedance structure may well have been the reason for amplitude differences that impacted the sense of space. My system at home has a much more conventional setup (Neukomm PA18 Monoblocks, Piega Coax 90.2 speakers, Prefer PMK-206 cables and some headphone stuff. A Dodocus XLR switch, yes, but no attenuator.

I still can't help thinking the different filter concepts used in the DACs had a significant influence on the spatial representation as well (which was the main criterium I used when trying to differentiate between DACs).
The way we did the listening comparisons was to use the same pieces of music over and over. This means we got to know them extremely well (to a nauseating level, actually). We used mainly acoustic stuff (Classical Orchestra, Opera, some acoustic Jazz). How far back in the soundstage is the trumpet at second 37? It becomes extremely selective over time, as you know what you are listening for. For the life of me, I was not able to tell the difference between the Weiss and the dCS using Phil Collins. So Phil Collins was not taken into account for the results.

Call this listening for cues expectation bias. But the experiment is easy to fake in the "I don't hear a difference" direction (simply chose random), less so in the selective direction. We did not know with which DAC we were listening for the cues.

One observation that would point to not just the impedance structure (and its effect on amplitude) to be the cause of the differences could be the quite different results we noticed when using PCM or DSD material.
The dCS Delius / Purcell does not do DSD, so we used jRiver to convert DSD to PCM (yet another red flag, I know) for the dCS, while the Chord DAVE got DSD from jRiver. Of course we also had a setup where both the dCS and the DAVE got PCM coverted by jRiver from DSD.
While the depiction of space with PCM material was very comparable (but different enough to be able to differentiate) between the DAVE and the dCS, the story was completely different for DSD. The soundstage of the jRiver on the fly conversion / dCS remained quite convincing, while the DAVE's "direct DSD" soundstage became a lot flatter. There was much less depth. I jokingly attributed this to the fact that Rob Watts hates DSD and has tried to make the DSD filters as bad as possible.
The opposite was true for the Merlot. "Wonderful" (pardon the word) DSD, but sub-par PCM. Andreas Koch is a believer in DSD.
All that might show is that the developers of the DACs also have a bias towards the technology they believe in and put more effort into that technology. <end of tongue in check>


@Blumlein 88 Are you aware of any experimental data which would point to which parameters of a filter determine its ability to accurately reproduce a 3D sense of space? I would suspect that spatial representation could be one of the more selective scenarios, as it only works if the spatial clues are kept intact with respect to several parameters (similar to a visual hologram being sensitive to noise and phase errors).

The differences in spatial representation could also be a factor why certain DACs are more fatiguing than others. With a more realistic representation of space the brain would have to work less hard to create its illusion of the musical event.

 

[Frank Dernie]

The differences in spatial representation could also be a factor why certain DACs are more fatiguing than others.

I must say my experience of hearing "space" in the reproduction of a recording doesn't suggest it is a measure of equipment superiority necessarily. Two things lead to my scepticism.

Firstly very many years ago a friend of mine did some tests to try to find both why some people prefer analogue to digital, and also why LPs actually sound pretty good given how massively worse than CDs their performance actually is.
One test was raising the background noise when playing a digital file to the level common on LPs. This created an increase in apparent stereo depth, and a smidge more width too (!!!!).
The other results are not relevant here.

My second concern is with multi miked recordings. Here it seems to me that there is unlikely to be any real phase coherence between the various instruments and voices, so that aspect of spatial cues would be absent, and what was there would be a synthetic creation of the recording and mastering engineers.
When I listened to the improvement wrought by an active DSP crossover over a passive one the improvement was noticeable with a rock music but not huge. With an old simply miked opera recording the improvement in the stereo imaging was very marked.
I came to the conclusion that whilst modern recordings are quieter and probably have a better orchestral balance the old recordings, for example by Decca and Mercury, using a simple microphone layout carefully positioned are more enjoyable for me.

My opinion is that the quality of the recording makes very much more difference than that of the kit we play it on.

 

[jacobacci]

When I listened to the improvement wrought by an active DSP crossover over a passive one the improvement was noticeable with a rock music but not huge. With an old simply miked opera recording the improvement in the stereo imaging was very marked.
I came to the conclusion that whilst modern recordings are quieter and probably have a better orchestral balance the old recordings, for example by Decca and Mercury, using a simple microphone layout carefully positioned are more enjoyable for me.

After a bit of thought, I totally agree with you

• The quality of the recording is probably orders of magnitude more variable than the influence of the effect I was suggesting here.
• The fatiguing remedy would have to work with bad recordings as well and that is clearly not the case. Bad recordings sound bad when reproduced by a good DAC, as they should.
• I was basing my suggestion on a scenario where the recording is constant and the DAC is changed (by the way we did not use modern multimiked ones, but older simply miked ones, exactly as in your observation).

Some of the old stuff is simply great.

 

[graz_lag]

This is exactly why I use a Squeezebox Touch, with its internal DAC. The measurements are as good as they need to be, and they just work. Not that I use it much, but I'm also happy with the CD player's internal DAC.

S

I have the same, which I play via Daphile and or LMS, and Squeezer via the Android phone ...
I do not have the remote controller tough so I use the udap shell to configure it ...
Ohh ... and great customer service by Logitech, you get a reply within 24 hrs., they are very consistent on that ...

 

[jacobacci]

When I listened to the improvement wrought by an active DSP crossover over a passive one the improvement was noticeable with a rock music but not huge. With an old simply miked opera recording the improvement in the stereo imaging was very marked.
I came to the conclusion that whilst modern recordings are quieter and probably have a better orchestral balance the old recordings, for example by Decca and Mercury, using a simple microphone layout carefully positioned are more enjoyable for me.

This is very much consistent with our comparison sessions. It was much more difficult, if not impossible to differentiate between the different DACs using a multitrack produced i.e. Phil Collins disc than a simply miked orchestral or opera recording.
My interpretation (which is kind of obvious) is that spatial cues only get preserved as a whole in a simply miked recording and are distorted / broken apart / not present in multimiked ones. Denon one points would be a modern attempt at recreating the old magic.
Simply miked recordings with intact spatial cues seem to be highly differentiating re. the quality of the reproduction chain.

One test was raising the background noise when playing a digital file to the level common on LPs. This created an increase in apparent stereo depth, and a smidge more width too (!!!!).

This one completely throws me though. Human perception is a strange thing, really.

 

[Thomas savage]

You think he eats this?


View attachment 18857

Ha ha good work,, I’m kinda wanting to try it.. @svart-hvitt whats it like mate ?

 

[graz_lag]

This is very much consistent with our comparison sessions. It was much more difficult, if not impossible to differentiate between the different DACs using a multitrack produced i.e. Phil Collins disc than a simply miked orchestral or opera recording.
My interpretation (which is kind of obvious) is that spatial cues only get preserved as a whole in a simply miked recording and are distorted / broken apart / not present in multimiked ones. Denon one points would be a modern attempt at recreating the old magic.
Simply miked recordings with intact spatial cues seem to be highly differentiating re. the quality of the reproduction chain.


This one completely throws me though. Human perception is a strange thing, really.

@jacobacci @Frank Dernie
Pls. give us a break !!!
We have a debt of 2,000 cal a day or so on our weekly intake - just to follow up your this week's exchange ...
I do not want to think abt. the cal weight we'd need to reply to you ...

 

[svart-hvitt]

Ha ha good work,, I’m kinda wanting to try it.. @svart-hvitt whats it like mate ?

The taste is definitely neutral!

Not a big surprise, is it?

 

[graz_lag]

Pure fantasy.

https://www.caretobeauty.com/lb/durex-play-pure-fantasy-intimate-stimulator-x1/

 

[jacobacci]

Pls. give us a break !!!

No worries, I think things will calm down now ….

Just as a feedback that this forum actually does provide valuable input, my colleague has just informed me that he has removed the Goldpoint attenuator from his system and is now driving the Brinkmanns directly from the dCS' output. The Oppo (which was the reason for the attenuator / switch in the first place) has been retired to the archive.
He is happy with the change in SQ (something about veils.....he must have been listening to a simply miked recording of Salome)