Article: LessLoss Echo’s End Reference DAC Full Review

[Louis Motek - LessLoss]

11 hours ago, Ralf11 said:

He's got 2 stmts. in that sentence, and stmt 2 seems to preclude making stmt. 1...

Thank you for the discussion. I will explain this statement from our website:

 

We always know that sonic performance has primarily to do with Jitter reduction, and that Jitter is always going to be contended, since it is impossible to measure with authority. We work by ear where the lab equipment can’t follow. 

 

Jitter content in digital streams is reliably measurable only in approximate amounts. Even while it is being measured, it fluctuates in real time. As a general rule, approximately 100 pS amounts are the steps which can be meaningfully measured. It is impossible using any available lab equipment to make authoritative statements about jitter content smaller than this, repeatable from lab to lab by independent researchers. Anything you may have read published in smaller amounts is only marketing hype and nothing else. 

 

Let me give you a visual example. In the lab, when you set up a sensitive jitter test, and you do nothing more than wave your hand around the digital cable, the scope shows wild fluctuations of the data readout. But when you listen to a sound system, and you have somebody wave their hand around the digital cable, you don't honestly say you can hear wild fluctuations in the sound quality. 

This goes to show how this particular measurement is far from the end of the story. 

 

Having said that, it is easy to show the correlation of jitter reduction to sonic quality when you use extreme amounts. Comparing 1000 pS to 100 pS jitter content in digital streams is easily measured, shown repeatably on scopes, and easily heard by your average audiophile on any half-decent system. But even this large difference in jitter content can be masked by horrible ambient listening conditions, for example when the floor is made of ceramic tiles or the room has cement walls. 

Now let us suppose you have a good listening room, a carefully tweaked system, and "authoritative" listening talent. Let's say you've been at this for years. For such a listener, far smaller jitter content differences will have proportionately more and more meaning, until you get to the point where two fanatically determined audiophiles will heatedly argue unto the wee hours about even the smallest changes in jitter content, far smaller than those that are meaningfully and repeatably measurable. 

 

When we developed another of our creations, the Laminar Streamer, we tried all manner of oscillators. Here's a picture of a portion of those we critically tested:

 

 

One of the things we learned from these crucial tests was that Jitter numbers don't say anything about the actual Jitter spectrum, and each Jitter spectrum will produce upon conversion to analogue some sort of sound coloration of its own. This completely apart from the Jitter amount as expressed in pS. So now we have two parameters: Jitter amount and Jitter spectrum. And when listening to all these different clocks under the same conditions in the circuitry, the subjective listening experience again does not seem to correlate with the data. Yes, you can "like" the sound of one spectrum of Jitter more than you "like" the sound of another. This will go all the way deep down the  system synergy rabbit hole. These are extremely fine distinctions.

Therefore, it is useless and deceitful to publish tiny jitter numbers as some sort of "proof" that your digital solution is better than any other one. It is only useful as a general tool to make sure you are not making any blatant design mistakes. But in the end, it is the ear that decides which solution is ultimately preferred and therefore, hopefully now, the statement that "we work by ear where the lab equipment can’t follow" makes sense. If you ever see Jitter numbers published smaller than 100 pS beware of shameless marketing. At these levels, three labs can give you three different numbers, and a single lab can and most likely will give you three different numbers on Monday, Tuesday and Wednesday.

[Louis Motek - LessLoss]

8 hours ago, AudioDoctor said:

Wow...

 

The DIY part linked above lists that it is capable of 24/384 and Native DSD. The LessLoss Website says the DAC is capable of 24/192 and DSD.

 

The S/PDIF standard only goes up to 192 kHz sampling rate, and that is the limit we published on our website. The USB input, however, does play 384 kHz sampling rate files, not that any truly exist. 

This type of talk about sampling rates has absolutely no correlation with sound quality. One can easily devise ways to create lower sampling rate files which sound obviously superior to their higher sampling rate counterpart. All you need to do is tweak the upsampling/downsampling algorithms in order to do this, and the market is chock full of available algorithms. Each has its own sound.  The unsuspecting listener often never knows, nor even takes the time to try to inquire, what the originally recorded sampling rate was in the first place. The general mentality and experience in this regard is so narrow and fragile that it is an embarrassment to the entire art of audiophile culture that this topic ever exploded the way it has. Remember the scandalous sampling rate hacks on HDTracks? The publishers would upsample to a higher rate and charge more for the downloads just because somebody passed the file through an upsampling algorithm, something that most any DAC today does in real time anyway, including Soekris. 

These days, most people listen to conversion being carried out at 384 kHz without their even knowing it. They play what they think are different sampling rate files (not knowing the original recording's sampling rate in the first place, nor having any way of finding out), then listen as their DAC upsamples in real time to 384 kHz, without even knowing it. 

 

Those who are quickly excited about sampling rates very quickly get turned off by the math and engineering behind it. It is ironic. 

 

Meanwhile, we and like-minded audiophiles are still discovering deeper and deeper depths in good 'ol 44.1. The whole question of sonic discovery in digital always was and always will remain the further and further reduction of jitter. It is just that simple.

 

The whole numbers race in digital audio can be traced back to the analogous numbers race in the competitive field of computer processing. The big difference is that the concept of audio quality is strictly a real-time process, whereas computer processing is always a break-neck speed of churning out of crunched numbers with error correction algorithms with no recourse to perfect timing in real time. Like, why do I have to wait for my cursor on my screen to show me the word I typed half a second ago? I think you get the picture. Latency and multi-tasked resource allocation vs. the smooth flow of real time. The prior easily marketable with faster and faster speeds. The latter boring as hell from a marketing perspective. 

 

This is why the higher sampling rate numbers are so much more attractive to those in the selling business.   

 

 

 

[Louis Motek - LessLoss]

The reason we choose the Soekris was because it had the potential through our critical listening tests to outperform our earlier favorite, the legendary Burr-Brown PCM 1704. The Burr-Brown came in selected batches and we always used the best ones. The Soekris also comes in different levels of resistor precision and we only ever use the most precise ones. 

 

As for the better sound we are now achieving from the Soekris as opposed to the PCM 1704, I am convinced that much of this has to do not with oversampling algorithms, not with digital filter choices, nor even the exact oscillator chosen. The real reason for the great sound potential comes from the fact that the current and voltage at the actual conversion process, and the trace thicknesses and resistor sizes, are much larger than in a microscopic laser-etched silicon IC scenario. Compounding this, the small signal strength which comes out of the 1704 requires the use of subsequent current/voltage conversion and this means more parts, more powered parts, and thus less purity and more noise. 

 

When you listen to the signal coming out of the Echo's End, you are getting direct access to the converted signal. You don't get this from any chip-based converter anywhere.

 

 

 

[Louis Motek - LessLoss]

30 minutes ago, PeterSt said:

 

You are really overdoing it now.

Yes, we like to take our every concept to the extreme. That's how we understand the art. Some call it purist. Some call it ridiculous. 

 

This reminds me of a whisky label I once saw. 

 

 

[Louis Motek - LessLoss]

6 hours ago, mansr said:

What is that supposed to mean?

I mean that ADCs typically aren't used to convert at 384 kHz. 96, yes, this is ubiquitous, 48 kHz, too, but in recording scenarios it is not a professional standard to record at 384 and therefore anything you see on the music market which claims to be a 384 kHz sampling rate recording is likely a mathematically contrived version of the originally recorded material.  This is a heated discussion in the audio recording arena.

 

There might be some scientific applications for recording at 384 like recording bats, but in order for this to be justified, all the gear in the chain needs to have extremely low noise even at ultrasound frequencies in order for the intermodulation effects not to add even more noise to the audible spectrum that we humans can indeed hear. Maybe some rare labs have this capability but for the world of audio this type of extension of sampling rates simply does not add value and can even be (due to interpolation distortions) detrimental to the result. 

 

If you ever compare a high jitter recording at high sampling rate vs. a low jitter recording at a low sampling rate, you will always prefer the low jitter recording. In terms of hierarchy of importance with direct relation to sonic quality, low jitter is much, much more important than the difference between, say, 48 kHz and 96 kHz. Today there are even ADCs (AK5397 for example) which can do 786 kHz but it remains disputed as to its usefulness in real-world (human ear) audio applications. 

[Louis Motek - LessLoss]

3 hours ago, The Computer Audiophile said:

Just copied from the LL site. I’m sure they can better answer your question. 

 

To answer this:

 

11 hours ago, BrokeLinuxPhile said:
 

This statement confuses me, what are they trying to say?  I don't get why you need an acoustically dead material here.  Speakers makes sense but not a DAC.  Thick metal would shield better.

 

Depends on the frequency you want to shield. For instance, in microwave applications, a shield needs to be airtight. Even the slightest slit (for example a hole for a wire to get through) will let microwave radiation though.This is why some shields are soldered shut completely, from all sides. The importance of gaskets in shielding is a well established art. 

 

But then there is another thing altogether: each metal type has its own "sound". If you shield with aluminum, steel, stainless steel, iron, tungsten, lead, gold, nickel, etc., you will always have differing sonic results. This sonic coloration comes form the electro-magnetic reflections with said shielding material. Through many experiments, we have come to the conclusion that we don't like the sound of aluminum. It is used a lot in audio these days, and we simply don't like its sound. This is not only an acoustical phenomenon. It is also due to the near-field electro-magnetic interactions. Yes, simply being there next to the circuity, everything colors the sound to some extent. Hence, it is an art to create an enclosure that the designer likes. You have to try many things until you come to such understanding. Some speak of shielding "closing off" the sound. Then there are the famous experiments where you take identical circuitry and build two same-sized enclosures, built only with differing materials, and the sonic results differ quite astonishingly. We like Panzerholz as a build material in this regard, but also for its acoustic properties. When you have any current running through any wire, there will be at a very low level some acoustic vibration (charged particles moving are already a form of tiny acoustic turbulence anyway). When circuitry or any connector is mated directly to Panzerholz, the Panzerholz aborbs a lot of this tiny vibration. This is also why our C-MARC wire is over-braided with cotton fiber. High quality professional microphone cable also always has cotton in it, for the very same reason.    

[Louis Motek - LessLoss]

11 hours ago, Ralf11 said:

 

but maybe made in Romania or Bulgaria, no matter where the designer lives...

The Echo's End DAC is made by LessLoss right here in Lithuania. We have here a VAT tax rate of 21% and income tax rate of 15-35%, not to mention mandatory state social security tax. Whoever said that the retail price does not include international shipping is incorrect. Our prices include international shipping via 2-3 day courier with full tracking on every item we sell, including all versions of the Echo's End DAC. 

Kaiser acoustics currently use the twin enclosure Echo's End Reference Supreme edition, having compared it to other cost-no-object DACs costing $100k. Ours is about one third that price. 

 

How do we do it? By being very careful to spend exactly on those features and implementations which bring direct sonic advantage. This is done by a lot of experience and countless direct listening evaluations. 

[Louis Motek - LessLoss]

On 4/26/2019 at 10:04 PM, Nsummy said:

 

Not a chance.  And if it did influence it, it would be in a negative way.  Think about all of the high precision laboratory equipment, medical devices, quantum computers, space shuttle electronics, etc. that are not encapsulated in a wooden box.   Its fine if someone wants to make a product like this and jack up the price;  I get it, there is a level of craftsmanship here.  But to say the wooden box is there because of performance reasons is absurd.

 

Panzerholz (Tankwood) is used as the base of the core chassis of F1 cars. Here's a picture of one turned over:

 

 

We did not choose this material for eye candy. We chose it for its performance. 

 

We carried out a calibrated studio comparison of Panzerholz and aluminum and published the results here:

 

https://www.lessloss.com/page.html?id=80

 

On that page you can find audio examples which you can download yourself and run your own comparison. The difference is enormous. 

 

Any current running through any conductor will create some amount of molecular movement. Current is defined as the flow of charged particles. If it did not have any friction it would also not generate any vibration. We are talking not only about miniscule amounts; surely everybody has heard a transformer buzz with their ears directly. Just how that buzz is dealt with through the design of the enclosure will also influence the final results of the audio performance.

 

Here is another set of comparisons, with audio and video examples:

 

https://www.lessloss.com/video_demonstration_of_high_performance_audio.html

 

If the build of an enclosure had no effect whatsoever on the resulting sound, nobody would be found tweaking these things. 

 

Returning to the F1 application, you can find more Panzerholz inside the cars surrounding the car's timing electronics. Here is a sliced open F1 car. At around 3:30 you can clearly see the Panzerholz encapsulated onboard electronics:

 

https://www.youtube.com/watch?v=F9WVtZHYjds