Purifi Class D

[PYP]

30 minutes ago, mocenigo said:

 In other words, it was only to make numbers better, not to solve a specific problem (which, in Bruno’s approach, is hysteresis distortion). 

Is the "in other words" part what they said?  I ask because the "only to make the numbers better" hypothesis seems silly to me.  The distortion was already so low that anyone could question whether better numbers = better sound.  And subjectivists use their ears, not the measurements (to the dismay of objectivists).  

 

One could just as likely interpret this theory as -- Mola Mola did less math but a lot more listening and finalized the result based upon listening rather than math.  I'm not saying that I believe that, but theories about intentions are just that.   Maybe both parties wanted to make the best product they could and found ways to improve upon the original design.  The world is surely large enough to accommodate two excellent companies innovating to improve the listening experience.  Some folks seem to want to turn lemonade into lemons.   

[PYP]

7 hours ago, mocenigo said:

You mean Hypex, not Mola Mola. Well, they used measurements, and trial and error, not listening.

Gosh!  Geek Gossip in the audio realm seems rather benign.  

 

I hope that after all the math and measurements there is extensive listening.  Since the Nilai amps include a discrete buffer stage, I'm assuming there was some tuning by ear.  

[PYP]

5 minutes ago, mocenigo said:

I do not know. Why design something so precise and then ruin it with some changes done by ear? 

Because the subjective reaction is the final measure of success?  But i would suggest asking the audio designers, including speaker designers who use extensive tests/measures, why they use their ears to determine the final product.   I know that the Tambaqui was tweaked according to listening tests.  

[PYP]

2 hours ago, barrows said:

The above is not totally accurate for the Tambaqui:  Bruno has stated that he used a slow(ish) rolloff filter because he thought it sounded better.  Some ASR types questioned this decision, as it resulted in a slight rolloff in FR before 20 kHz, those types at ASR might consider this (ruining its performance).  Digital filter response is a choice, as when considering 44.1 kHz sample rate input there is no "perfect" way to design a digital filter-one is trading off one aspect of performance for another(phase response vs frequency response, vs filter artifacts) and the resulting filter is always a compromise in one (or even two) area(s).  But in general, as above, I agree.

perfect example.

 

Also, for the Kaluga amps, Kubala-Sosna cable was used to connect the PCB to binding posts.  Why?  Did it measure better?  Or sounded better and measured no worse?

 

When one reads about respected designers who carefully choose certain capacitors, for example, the choice is stated not in terms of measured differences (which one assumes were used to narrow the choices, along with considerations of high rate of production meeting the design specifications, cost, availability, etc.) but preferred "presentation."

 

I'm certainly not advocating for designing by ear, nor making changes that measure worse.  But as Bruno has mentioned, sometimes listeners can hear noise (or choose a better word) that initial measurements missed (as @barrows noted).  He takes that as a challenge to find the measurements that identify the problem area.   Had he not done so, I don't think Class D would today be considered a topology that works for the high-end market.   He blazed the trail.  His work has greatly influenced Grimm Audio, Hypex, Mola Mola, Kii and Purifi.  

 

At any rate, I think there is more agreement than disagreement in how we perceive these issues.   

[PYP]

3 hours ago, leManu said:

After many hours of listening to the Nilai500, the "break-in" period seems to have do good.  Or I'm getting used to the sound of it.  Not as warm as my old amp, but I can get why people love this sound.  I also totally enjoy the power, dynamic range and definition of it.  I managed to have a better stereo imaging than I had at first by moving my speakers around.  But still the depth of the soundstage is not as good as before.

 

Which brings me to an issue I still have with the sound.  Every album sounds like I'm sitting in the front row.  Everything is "forward" or "in your face" for lack of better terms.  It's enjoyable for rock music, but for folk or world music it's just too much, and made it impossible for me to have the "suspension of disbelief" I enjoy when listening.  I've tried to pull back and forth my speakers in the room with no real difference.  Is this the way this amp is "tuned"?  Or is this also considered something people enjoy?  Cause I sure don't.  Can it be attenuated with cables or something else?

I'm not familiar with your gear (that is, haven't heard it), but "in your face" is not what I experience with Ncore-based amps (the version before the latest, improved Nilai).  Of course, this is all relative.  

 

Perhaps if you heard my setup you would react the same way.  I don't think so, given the strong preferences of the other listeners in our household (my wife and cat).  I've had systems in the past that were more forward and she didn't listen very often.  With the current setup, she makes her own playlists and (especially for her favorite vocalists) will crank the volume (previous to my current setup that never happened).  All of that is to say that it may not be the characteristics of the amp.  

 

Personally, I have found that power conditioning (which depends upon your location and power delivery to the home) and cables can make a significant difference.

[PYP]

2 hours ago, mocenigo said:

But I want to stress that I firmly believe that everything we hear in a properly controlled setting can be measured. The problem may be to identify the right physical quantity to measure, but our ears are, frankly, crap, so anything we can ear should be easily measurable with standard equipment.

I was nodding in agreement as I read your post until I got this the highlighted part.  Do you mean that gear can be designed that has better measurements than the ear can detect?  If so, what would be the point?  

 

[PYP]

4 hours ago, mocenigo said:

Well, this is already happening for many years, with DACs, and it started recently with amplifiers. I would say, bragging rights. You have a DAC with a THD+N (A) of 120Db, but I have the next iteration from the same company, with nearly the same circuit, and a THD+N (A) of 121Db. I win. Our ears have an internal distortion that is ludicrously higher, so any distortion level over about 100Db (and probably also much higher) cannot be discerned. Noise is covered by ambient noise except in an anecoic room, but the noise of your own circulatory system is already higher. 

https://www.dailymail.co.uk/sciencetech/article-2124581/The-worlds-quietest-place-chamber-Orfield-Laboratories.html

 

The longest that anyone has survived in the ‘anechoic chamber’ at Orfield Laboratories in South Minneapolis is just 45 minutes.

 

"When it’s quiet, ears will adapt. The quieter the room, the more things you hear. You'll hear your heart beating, sometimes you can hear your lungs, hear your stomach gurgling loudly.

In the anechoic chamber, you become the sound."

[PYP]

In this video (starts at 1:40), Eelco Grimm explains their approach to the DAC section of the MU2, which is both a streamer and DAC.  

 

 

 

[PYP]

For those who prefer the read the explanation in more detail:

 

https://www.grimmaudio.com/publications/the-mu2-major-dac/

 

 

[PYP]

41 minutes ago, barrows said:

Sounds like very similar architecture as the DSC-1/2, and Tambaqui.  Using a discrete FIR DAC for conversion.  Although each system seems to vary a bit in the way the oversampling and modulator is approached (in the Tambaqui onboard and implemented in three powerful DSP chips, in the Grimm an FPGA, and with the DSC-1/2, typically via HQPlayer's wide variety of modulators and filters).  Grimm's 11th order modulator seems like it might be unique.

Would love to see what @muski, Jussi, might have to say RE, Grimm's approach, and the noise profiles shown in the link.

 

Just realized we are way off topic for this thread, oops!  But the discussion is interesting, maybe we should move it somewhere else?

MU2 info under the MU1 thread for now (see last few pages):

 

 

[PYP]

8 hours ago, mocenigo said:

That’s why the Kroma speakers at Munich hiend were almost universally considered the best of show. There was a very different character and an obviously superior resolution and clarity. And it was all in the speakers, nothing else really mattered. But even that is rare.

Did this Kroma use Purifi (among other) drivers?  

[PYP]

31 minutes ago, barrows said:

If a current SOTA class D amplifier, like a Mola Mola has problems in your system, look somewhere else for the source of the problem, it is not in the amplifier, regardless of what class of operation it runs in.

It seems that some reviewers don't understand or believe this.  Some place an ultra-low distortion amp in their familiar setup and then start describing the sound.  It seems to me (my working hypothesis) that the latest Purifi and Hypex implementations that reduce distortion to previously unimaginable levels are very close to not having a sound.  

 

Can that level of low distortion be achieved by streamers and DACs?  

 

Meanwhile, Purif and other manufacturers are reducing the distortion of drivers, cabinets and the two together in the final design.   At what point does the lower distortion result in a presentation that is closer to the real thing than previously heard?

[PYP]

9 minutes ago, barrows said:

 

In my Purifi build, I changed out the OPA 1612 for OPA 1656, on the suggestion of Bruno Putzeys.  The amp sounded different (better) but any measurements for distortion would not show significant differences, as the measured standard distortion of these parts are both at levels which are not audible, at least by standard practices.

 

Are you trying to achieve a certain sound signature?  I'm assuming that would be what sounds natural to you in your setup.  

[PYP]

49 minutes ago, davide256 said:

 

 

Recommending a $17K amplifier does leave a good class D as unobtainable for most budgets. Well meantime I'll keep looking

for some science behind what I encountered and how to do improve next time.

I don't think one need spend $17K for excellent specs and sound (the Mola Mola has great innards, but the cost is increased by the fancy exterior).  The concept remains the same -- experimenting by working back from the low distortion amp to determine the contribution of upstream components and where improvements might be had.  

[PYP]

51 minutes ago, leManu said:

I've bought a hypex nilai500 amplifier, and it's the worst decision I have done in hi-fi. There is just no music coming out of it. Sounds yes, and pretty accurate ones, but not the beginning of a melody or an emotional connection with the musicians.

 

I then stumbled upon Soulnote brand with a possible explanation for what I was experiencing.

 

"Soulnote is a high-end home audio brand by CSR, Inc. headquartered in Kanagawa, Japan. CSR was founded in 2004 by the former members of Marantz Japan. Hideki Kato, the Chief Sound Manager since 2016, has his long professional career since late eighties, having deeply involved in the development of the legendary A-10 amplifiers by NEC as well as the LHH series by Philips, but having kept himself staying behind the scene.

 

Soulnote's design philosophy

 

Through his long carrer as a designer of audio equipment both in analogue and digital areas, Kato has been struggling to find out the reason why the improvement in static performance and the sound quality in playback did not always correspond. Kato made a hypothesis about a counter-concept which he called dynamic performacne. The adoption of the NOS (non-oversampling) mode for PCM playback in digital-to-analogue conversion as well as a non-NFB (without negative feedback) circuit in amplifiers come from this hypothesis. Indeed these worsen the measurement result in static performance, but a remarkable improvement can be recognised in dynamic one.  Soulnote believes dynamic performance keeping accuracy of the original waveform on the time axis as the most important for music playback, which is, however, still unmeasurable by any of the conventional methods. At Soulnote, only listening dominates the determination and improvement of circuit, selection of parts and mechanical construction. This approach is a kind of antithesis against the supremacy of static performance."

 

I'm not an audio engineer, but I can say without a doubt that Soulnote, and the many other bands that won't use any negative feedback loop in their products are doing it for a really good reason: music. I've been listening to a few amplifier in the past weeks and that point was always true. Negative feedback loop kills music. It's fantastic for the static numbers, but I rather listen to music with my ears than with my calculator or audio analyser.

 

If you have played 400 hours of music through the Nilai and it doesn't sound good to you, then this particular implementation of Class D isn't for you.  Other implementations might sound better to your ear.  

 

Which Soulnote amp sounds good to you?  And which brands have you listened to that use no feedback (I'm assuming you mean no global and no local feedback)?

[PYP]

15 minutes ago, barrows said:

The notion that feedback, generally, "is bad", is just plain nonsense.  All audio amplifiers employ feedback in some degree-there is no such thing as an amplifier without feedback.  

 

 

Sim Audio has this bullet point for the 760A that he tried:

 

Our “no overall feedback” amplifier circuit design which results in the following: real-time amplification; more accurate musical reproduction with respect to tonality; non-existent transient intermodulation distortion; the elimination of common phase errors resulting from feedback.

 

I wonder if the quotes mean that it doesn't literally use NO global feedback, but that it is negligible.  And if they didn't use global feedback, what about local feedback?  Off topic...but is there a good source of information for explaining how feedback is implemented in audio?  

[PYP]

7 minutes ago, davide256 said:

Actually my old modded class AB Hafler is far more relaxing/enjoyable to listen to if I'm not listening to "car crash" pieces.

So it appears there is no free lunch with class D, that getting acceptable results for both linearity, dynamics + mid range tone color saturation

requires more than a basic class D.

 

well, everyone is welcome to their opinion.  But broad generalizations are just that.  Or should we read the emphasis on the "basic" in your statement?

[PYP]

17 minutes ago, davide256 said:

And some people post word mish-mash which helps no one. Glad you like your $17k class D amps... thats a bit out of touch with the reality of amplifiers

in the $1~2k range

my experience with Class D started with a Hypex DIY amp (many years ago).  That showed me the promise of the Hypex module and I became curious about what else was out there.  NAD used an OEM version with their own power supply and that sounded much better than DIY.  Eventually, I did want to hear Bruno's best (at the time) and that demo resulted in my current setup.  

 

These days, however, it seems that advancements in the design of the module has lowered the price of truly great Class D (and by that I mean the enjoyment of music).   I mention this because there are folks who are just starting out in this hobby and it is helpful to be topologically agnostic when exploring entry points.  

 

[PYP]

42 minutes ago, Jud said:

People actually hear differently from each other, as a result of both inheritance and life experiences.

And, saying the obvious, hearing for any given person changes over time.  And experience with listening can lead to different preferences.  

 

Since I have had my amps and DAC for many years, eventually I started to hear what I thought were limitations in playback (meaning, less than realistic sound).  My assumption was that the existing gear was very transparent, therefore upstream components might be the issue.  That turned out to be correct.  

 

This has led me to the belief that very transparent gear is both a great advantage but also a challenge (it is like adding new, revealing speakers and then having to re-assess your electronics).   For Class D, if one is willing to experiment, the benefit can great bang for the buck.  It isn't for everyone, but if you don't know your listening preferences yet, it is worth trying.  

[PYP]

41 minutes ago, mocenigo said:

Emphasis and distortion can always be added somewhere else, and the power amp can just do what it does better: amplify and nothing else.

 

That is my starting assumption too.  Choose an amp that is transparent as possible.

 

At some point on the transparency continuum, the large differences across recordings (recording venue + recording engineer + mastering) becomes obvious.  That may be a desirable outcome for some and a bummer for others.   

[PYP]

44 minutes ago, mocenigo said:

 

Where are the ASR measurement simplistic? Have you seen the wealth of data they provide, or you have just heard word of mouth that they do only SINAD?

would you say that ASR is using the same measurements as Bruno does to design his gear?  I'm assuming the answer is no because Bruno has discovered some new ones along the way (otherwise, how does he continue to improve his amps, transducers, etc?).  

 

It seems that the ASR philosophy (or is it just the conclusion of some of the participants) is lowest distortion = best.  That wouldn't be a good conclusion if the measurements are "incomplete."

[PYP]

3 minutes ago, davide256 said:

umm, thats all obvious with even a modest system. What matters more to me is do my toes want to tap, am I tempted to sing along or does it leave

me un-engaged. Even bad recordings can be engaging if the system has "soul" and the artists performance is engaging. Lots of pop recordings from the 60/s 70s

that are poor masters but highly engaging.

agreed that all good systems reveal the recording.  As my system got more transparent, those differences became more apparent without a loss of enjoying the music.

 

For me the goal is being able to hear the musicians tell their story.   I enjoy hearing every bit of the timbre and tone that makes a musician's voice unique, and every little bit of nuance in how they interact with their bandmates.  I don't want to have to choose only the best recordings to enjoy music.  For me, a low distortion, highly transparent setup has been the means to that goal.  

[PYP]

23 minutes ago, mocenigo said:

Anyway, the owners of Purifi disagree with you since they lug around a Topping (and one of them asked me whether it was plausible that he could not hear a difference between the Tambaqui and the D90 non-MQA first series - I omit the name or Bruno will punch him :-)

 

I've never heard two DACs sound identical.  Or two pieces of any kind of gear.  

[PYP]

1 hour ago, Matias said:

What he does is ranking them by 1kHz 5W into 4 ohms SINAD, which we all agree that is not everything. 

 

 

Funny you mention that, as some months ago I brought my RME ADI-2 DAC (with iFi iPowerX PS, AKM 4493 version) to play in my main system (based on Thiel CS3.7, NC500 back then, treated room, FIR room EQ) in place of my Makua preamp+DAC. Non blind, non level matched, just subjective. There were differences, but I would say they were tiny overall. I own both devices, so no bias to favor one or the other. If anything made me impressed by the RME, and questioning the value of the Makua. Still I will not sell it, it is already paid for, I love it as well. :)

I've only read ASR briefly because many participants seem to have severe IBS.  Life is too short.  But if the comparison graph, which may be the only thing that non-technical novices look at, is not really relevant, what is the true utility?  

 

I think your comparison of DACs in one's own system and with your own ears is the only relevant one.  I have tried a Schiit Audio Modius DAC ($229) fed by the Grimm Audio MU1.  The MU1 is a DDC that does some of the complicated math (giving @mocenigo a few moments of rest) before handing off to the DAC.  The MU1 handled the volume control as well.  Result?  Sounded very nice, and if "pleasant and enjoyable" was one's goal, you could stop right there. 

 

After a week, the Tambaqui took its place.  If "much like the live event" is one's goal, this DAC is preferred.  Of course, the context is very favorable to the Tambaqui (I'm assuming Bruno used the Kaluga amps when he designed his DAC, which are the amps I use).  The Tambaqui replaced a PS Audio Directstream.  Once the Tambaqui was warmed up, the in-home demo took about 5 minutes before a decision was made to purchase the Tambaqui.  

[PYP]

25 minutes ago, barrows said:

I easily heard what I would consider vast differences between my Bricasti M3 and the D90 (first version), and my DIY DSC-2 DAC, given DSD 256 feed from HQPlayer and direct conversion of the DSD stream in the D-90.  This was here in my room under direct comparison.  My experience of the Tambaqui has been in different system contexts by memory only, but from those memories it outperformed any other DAC of my experience.  Main differences were in the ability to present a naturalness of tone and dynamic expression, while remaining extremely detailed.  The D-90, while almost as detailed, missed out on sounding natural, or dynamic, for that matter.  

While I don't believe YouTube is a good medium for comparing gear, I did watch a comparison of the Tambaqui and a similarly priced dCS.  What was very apparent from the video was the difference in portrayal of rhythm.  The difference was large and, for my tastes, favored the Tambaqui.  

 

Of course, I couldn't hear the differences in tone and timbre, which in my setup is a particular strength of the Tambaqui and a quality that enables me to hear what the musicians are expressing.