We already have had a complete review piece about DragonFly Cobalt for almost three years now at audioreviews.org, and based on shared appreciation within our team we decided to stick it onto our Wall of Excellence. As AudioQuest sent me a sample too, I am now sharing my own take on the device.

Very interestingly, AudioQuest recently repositioned Cobalt’s price to € 199,95 in EU (down from € 299,95) – and I can anticipate this is a KO move vs much of its direct competition. Cobalt is widely distributed, and can be purchased from multiple channels including Amazon and many other online platforms.

At-a-glance Card

Features and description

Externals

DragonFly Cobalt is the size of an old-school “USB pen drive”, with is encasing painted of a nice blue – or well, cobalt – color.

There are no controls, wheels or buttons whatsoever on the structure. Only after plugging it into a host PC (or Mac, or mobile device) one realises that the stylised dragonfly logo on the top side is indeed backlit by a colored LED, which color changes depending on the input stream digital sample rate:

Internals

The DragonFly product range is based on technology developed by a mr Gordon Rankin, a gentleman busy with seriously innovative digital and analog audio technology and products for the past almost 4 decades.

Gordon’s own company Wavelength Audio Ltd still holds the rights to the registered trademarks and of course the intellectual property at the very heart of AudioQuest’s DragonFl(ies), including DragonFly Cobalt of course.

Just for the sake of historical curiosity – and give Caesar his own, of course – StreamLength® is the given name of Gordon’s original setup which for the first time allowed for a plug-in device to take control of USB communication timing, which was normally exclusively managed by the host (the PC) until then. Such flip of perspective is the crux to the nowadays ubiquitous “Asynchronous USB mode”, the very base to start from and achieve adequate control over jitter when it comes to digital audio communications.

Once put the plug-in device in a control position over communications timing, the other step Gordon took was that of giving the device itself a high precision clock generator, which in DragonFly Cobalt case happens by taking it out of the very ESS ES9038Q2M chip at its core. Such setup was originally named “monoClock® technology” by Gordon.

Another fundamental architectural choice taken inside DragonFly Cobalt is to adopt separate chips for digital reconstruction (DAC) and amplification – as opposed to most of the direct competition relying on “all-in-one” chips doing both things on the same piece of silicon and – which is most significant to our discussion – without offering integrators and users any significant flexibility to change, fine tune, let alone customise the overall system behaviour. That’s why DragonFly Cobalt’s voicing, dynamic range and SNR won’t be apriori similar to that of other dongle devices relying on ES9038-line chips’ internal amping features.

Exploiting another feature on ES9038Q2M, DragonFly Cobalt comes with a custom designed minimum-phase slow roll-off FIR filter.

While I have no complaints about such choice, considering the hardware offers support for it I would welcome the chance to apply different filters, via good ol’ firmware flashing or even better via some sort of mobile app. Maybe there’s a chance this would come in the future?

USB communications are carried out via the good deeds of a Microchip’s PIC32MX274 IC, featuring extremely low power needs, also vis-a-vis its programming for support only USB-1on DragonFly Cobalt.

All such choices – the DAC chip, the AMP chip, and the USB processor chip – contribute to DragonFly Cobalt being amongst the lowest-power-demanding dongles on the market when it comes to host supply needs, which is why it is one of the very few to offer full compatibility with iPhone/iPad hosts, notoriously unable to deliver other than very low power off of their Lightning sockets.

Mind you though: free meals not being a part of real life, nor any divinity existing taking care of creating energy from nothing no matter how hard you pray, low input power draw means a few things that you do need to keep in mind to set the correct expectations about DragonFly Cobalt’s performance.

One: the USB-1 protocol drains much less power at the cost of a limited maximum transfer speed, which in terms of digital audio values turns into a 24 bit / 96 KHz digital resolution cap. And… PCM only! No DSD support.

Two: the ESS 9601 amp will prove limited in terms of maximum output power, with particular regards to current . So while it will reliably deliver a nice 2.1 V max swing on high impedance loads, DragonFly Cobalt will not (as it can not) adequately power low(er) sensitivity drivers, especially if featuring low impedances too.

So in practical terms you should not rely on DragonFly Cobalt to properly driver the likes of final E5000, or final B1, or final A3000, let alone pretty much any planar driver. 

And, no surprise should arise when Cobalt will seem to “struggle” (e.g. in terms of loss of treble control) with “mid-hard” loads. All that will be due to the device’s internal power circuitry “running out of current” in some situations, having it apriori been set up not to request more than a certain, very limited power from the host device to begin with.

Input

Like all pure “dongles”, DragonFly Cobalt only accepts USB input.

Very “classically” the device carries a USB-A male plug, so in itself it’s ready to plug onto any common PC or Mac USB port.

A USB-A to USB-C short cable adapter is supplied too, to facilitate connectivity with more recent smartphones. More on the adapter under Package, here below.

Output

DragonFly Cobalt’s sole output is its analog 3.5mm connector, of course accepting any 3.5mm single-ended terminated load.

Those who (also) own balanced-ended sources will most likely have a few or many of their drivers equipped with balanced terminated cables, and will need a balanced-to-single-ended adapter to plug them onto the Cobalt.

Host power requirements

DragonFly Cobalt requires very low power from the host (i.e., the PC, the phone or the dap it is connected to and therefor powered from). I’m talking about just 60mA when idle (i.e. when connected but “doing nothing”), and between 150 and 200mA when playing out on good volume on a mid-impedance driver.

This is of course very good news, but grounds for some caveats too.

Starting on the good side: DragonFly Cobalt will not suck your phone battery dry in no time like so many direct competitors and (!) it will perfectly work with iPhones and iPads, known picky fellas when it comes to the powering requirements of the devices you plug onto them. It will also not more than vaguely warm during operation.

On the flip side there are two important notes to make – which I indeed already mentioned above under “Internals”.

One: DragonFly Cobalt exclusively supports the USB 1 protocol speeds (USB 2 would require more power), which translates into a maximum supported input resolution 24 bit, 96Khz PCM (and no DSD).

Two: DragonFly Cobalt’s maximum output power will be, of course, limited, too: expect it to be good for powering high impedance (300 ohm) dynamic drivers (e.g. Sennheiser HD-series cans) and mid-impedance (20-30 ohm) not particularly demanding IEMs – which are, combined, probably >95% of the drivers out there anyway.

Volume and gain control

DragonFly Cobalt offers no physical control options so there’s no way to set the gain, and the sole way to manage its volume is via the host’s digital volume control.

On such latter front a point, as you may or may not know, the Android operating system divides the USB device volume range in only 40 steps (or even 25 for the latest Android releases…). When operating a device like DragonFly Cobalt this results in the last ticks of the volume control range converting into way too big SPL variations.

So if you are planning on using DragonFly Cobalt on a Android-based host just keep in mind that the way to “fix” this is using a better featured music player app e.g. UAPP or others – which is what you would normally do anyway for a number of other reasons one above all bypassing standard Android audio drivers – re-defining the number of steps Volume control is divided into (up to 250, on UAPP).

Other features

MQA Rendering

I won’t spend a word on what MQA itself is, of course. Google around if you wish and you’ll be overflooded with info.

What matters here is: DragonFly Cobalt is a “MQA Renderer”, so it can fully unfold MQA tracks on its own hardware, which is an upgrade vs the default represented by having the music player host do the unfolding, and only limited to the first 2 folds.

Firmware

For their DragonFly product line AudioQuest offers free software for users to autonomously carry on firmware upgrade operations when needed.

For DragonFly Cobalt no updated firmware version was (yet) ever released, however.

DragonTail

DragonFly Cobalt ships together with two complementary accessories: a leather sleeve, and a USB-A to USB-C short adapter cable – which, as always in AudioQuest’s standards, got its own given name: “DragonTail Extender”.

The DragonTail appear as nothing else than a digital plug format adapter, much needed of course to facilitate connecting the Cobalt to mobile devices like smartphones, or dap/transports. The version bundled with Cobalt is called DragonTail-C as it features a USB-C male plug at its end. AudioQuest also carries a DragonTail-Micro alternative.

What’s probably most interesting about DragonTail is its quality. I won’t take a digital audio cabling detour here, but I think it’s worth to share a very simple experience I had with Cobalt. When I first put it to work I connected Cobalt to my PC, where there’s only a USB-C port available, so I used a “nameless” USB-A to USB-C adapter cable I had laying around on my desk. No surprises: it just worked as I was expecting it to, and Cobalt sounded “right” off the bat.

One day for whatever reason I needed a USB-A to USB-C adapter for another application. Where do I have one? Meh… who knows. While watching around I noticed the one hooked to the Cobalt so I just took that one off, leaving the Cobalt disconnected for a while.

A few days later I wanted to use the Cobalt again, and did not want to “undo” the other cabling involving the other adapter. Time to think harder and try to devine where could I have another one – and that’s when I remembered there must have been one left inside the Cobalt box. Took it. Plugged it. It worked (of course). But… Cobalt seemed to be sounding different.

Mmmh – I thought – that’s very likely my wrong memory. So I carried on, for a day or two. Then, I decided to check it. Undid the other cable chain, recuped the “nameless” adapter cable, and organised a quick A/B test. And yes, there is a difference. When using DragonTail to connect it to my PC the Cobalt delivers fuller notes, and a darker background.

As I mentioned en-passant within my article regarding AudioQuest’s JitterBug, a passive cable cannot possibly “improve” a digital signal. However, it can deplete it. So what is actually happening on my case is that DragonTail revealed that the other cable was introducing noise…

DragonFly Cobalt sound

DragonFly Cobalt sounds detailed, dynamic and most of all clean, yet significantly musical.

In terms of cleanness in particular it trades (hard!) blows with the E1DA’s 9038SG3 and 9038D, arguably the “cleanest” – in the sense of most distortion-free – dongle-class devices one can find.

Which leads me right to articulate about the true crucial point of Cobalt’s sound: its stunningly spot-on compromise between resolving power, transparency and musicality.

I can name other more musical (“gracefully colored”) dongles. I can name more transparent ones, too. Very often, if not invariably, auditioning one of either group makes you soon want one from the other. Cobalt is not that. When listening to Cobalt’s clean notes you can’t fail noticing how expressive they also are, and, while going with Cobalt’s musical flow you’ll never feel you are really missing tiny beats, or soft nuances.

Cobalt’s output is masterfully “balanced”, not in the meaning we most commonly give to the word, regarding properly reciprocally calibrating lows mids and highs, rather is the sense of delivering as much of both – transparency and musicality, clean timbre and personal tonality – one can realistically hope to have at the same time.

Considerations & conclusions

DragonFly Cobalt is an absolutely remarkable piece of gear offering high quality reconstruction, reference-level amping transparency and delicious tonality, all near-magically mixed together at a unique mixture spot.

With its minuscule input power demands Cobalt is possibly the easiest in its class to pair with any mobile transport, iPhones and their (in)famous interfacing standards and power-out limitations. And, its single-ended output fully supports all those stock wires you may have a drawer full of, without leaving a balanced alternative to be desired.

Some may consider its modest maximum output power a limitation, and technically right so. However, in practical terms they translate into letting out possibly less than 5% of the IEMs on our Wall of Excellence.

DragonFly Cobalt was stuck onto our Wall of Excellence long ago, when its price was € 299,95. Now it’s been repositioned to € 199,95. Enough said, I guess.

The Editor: I have been using the Cobalt daily for the last three years...despite a drawer full of dongles: sounds great, has low power consumption, is small and light, and easy to use. The (monetary) value is mainly in its DAC part.

Our generic standard disclaimer.

The post AudioQuest DragonFly Cobalt Review (2) – Knowledge Is Power appeared first on Music For The Masses.

I have 3 units to test within my quite articulated home setup, and verify if / which sonic improvements are determined by the presence of one, or more, JitterBug FMJ units in line with and/or in parallel to my various DAC connections.

JitterBug FMJ retails in EU for € 69,00 and can be purchased from multiple sources, including Amazon. The manufacturer’s official information page is here.

At-a-glance Card

Introduction

I know it very well: there’s a thick population of “non-believers” who apriori refuse the whole concept of USB filtering.

Sole thing I can say about and to them is: fair enough ! If you are one of that lot I recommend you stop here, don’t go forward reading this article as I guarantee you won’t like it so why bother.

Some others are instead very confused about the topic.

Not about AudioQuest’s JitterBug FMJ per se, actually, or not yet about it, insofar as they “stop much before”, not having clear which “noise” are we talking about that a device like JitterBug FMJ (and others, in the same category) is supposed to do something with.

To all of them I dedicated the introduction to a past article of mine reviewing IFI’s Nano iUSB 3.0 filtered power supply, which I spun off into a separate post some time ago precisely to conveniently back-link to it from within other review articles (like this one), without dumbly “re-pasting” the whole thing again and again.

Long story (too) short: there’s of course no way nor need to “improve” on digital data quality. There’s however point, and need, to avoid that data depletes during transport, and/or that transport media (cables) “trojan-ride” spurious signals, together with legit data, which may, and will, perturbate the DAC’s activity.

General description and features

JitterBug FMJ looks like, and has the size of, a common USB-key drive. It’s in facts a sort of “passthrough” thingie: on one end is a USB-A male plug, on the opposite end a USB-A female connector.

The chassis is metallic, studied to protect the inside from nearfield RF interference. The female USB port is protected by a removable “cap” make of rubber mixed with carbon – that, too, aiming at tackling RF interference. Both such features (metal chassis and backport cap) are indeed inherent to this new “FMJ” version and were not present in the original JitterBug. FMJ standing in facts for “Full Metal Jacket”.

Finally, the ciruitry on the PCB inside the case is aiming at removing in-line RF interference, such as that generated (or transmitted) by pretty much any digital device e.g. a computer, a TV, an audio player, etc.

You can see JitterBug FMJ as a filter reducing / eliminating any signal coming out of a USB port which is distant from the working frequencies required by the digital data which are solely supposed to be managed by that port.

As a consequence the DAC will receive “just what it’s supposed to get”, with no, or at least much less “other spurious stuff”.

Of course electrical impurities might not be there in the first place in some case, or, they may be filtered/rejected by some circuitry built into the DAC device itself, and in these cases adding a Jitterbug FMJ may be simply… useless.

Another case where a JitterBug FMJ may be only partially or not-at-all beneficial, is of course when spurious signals and interference are picked up downstream of its location.

So in general JitterBug FMJ (and all other similar equipment) is not – as it cannot be – a guaranteed hit, nor a guaranteed complete solution.

As in all or at least most things audio, a try is needed to know if and how much it benefits each particular setup.

How to use it (in the manufacturer’s intention)

Quite simply, Jitterbug FMJ is intended to be plugged into a USB port on a music player host (a pc, a mac or a linux box). Then, the USB cable leading to an external DAC or DAC/AMP will be plugged onto JitterBug FMJ’s female connector.

There’s no driver to install, no options to set. Just plug it in and leave it there.

The removable rubber cap covering JitterBug FMJ’s female port is supposed to be put back in place when no USB cable is connected. That’s because the carbon mixed into the rubber material helps acting as an anti-RF shield.

Always according to AudioQuest there’s also another way to use JitterBug FMJ: install 2 of them in parallel on the same host machine, plugging them onto two different USB ports (partaking to the same internal USB hub).

Onto one of the two JitterBug FMJ the USB cable going to the DAC is supposed to be plugged. The other JitterBug FMJ will just stay passively there, with the back rubber cap installed, and may (or may not) add a further level of intereference removal from the USB line.

OK, but does JitterBug FMJ actually work ?

Simply put: yes, and well, too.

First things first, I tried Jitterbug FMJ at its main intended usage scenario: plugged in-line between a host and a USB DAC or DAC-AMP.

I tried this on all 4 different hosts I normally use (also) for audio application, which are

• an aging MacBook Pro 2012 reourposed into acting exclusively as a Roon server
• a Lenovo Y520 laptop with Windows 10 which is my main general purpose work platform, including Roon Remoting, and gaming
• a BananaPi M2+ box with Debian Linux acting exclusively as a Roon Bridge, and
• a RaspberryPi 4 with Dietpi Linux (a well packaged Debian distro) also exclusively acting as a Roon bridge.

DACs (DAC/AMPs) connected to those include my Questyle CMA400i, the Earmen ST-AMP unit I’m reviewing, and the main “dongles” I own, which include Apogee Groove, E1DA’s PowerDAC 2.1, 9038D and 9038SG3, Questyle M15 and AudioQuest’s own DragonFly Cobalt. Oh, and a Chord Mojo, too, every now and then.

“Dongles” (i.e. host-powered) devices are by definition those exposed at the highest risk of “inheriting” host perturbance carried over via digital interconnects, that’s why I expect JitterBug FMJ’s effect to be most evident on them.

I also expect JitterBug FMJ to be more beneficial on devices plugged onto my Y520 laptop, and less so when the host is one of the raspberries (you should know the rationales of such expectation if you know this stuff at the technological level, or if you read my article referenced above).

Long story short: JitterBug FMJ does work, i.e. it did deliver a sound improvement, in all my different install positions.

The effect on final sound has been more evident, at times totally obvious, in some cases, and more subtle in others.

I can hear improvement on two main areas: better, more rounded up, fuller notes and darker background. Both these improvements together also result in a better sense of macrodynamics (imaging), which, depending on musical genre, also improves on rhythm perception.

Expectedly, out of all my gear the device for which the improvement is most subtle (yet still audible) is Questyle CMA-400i, no matter the host it is connected to.

Again very expectedly, the cases where Jitterbug FMJ’s improvement is obvious are those involving dongles (all of them – yes, including AudioQuest’s own Dragonfly Cobalt), connected to all my hosts, and maximally when connected to my Y520 “gaming” laptop.

Of course I also tried the other manufacturer-suggested use case, which is that of adding a second JitterBug FMJ in parallel to a first one, connected to a free USB port on the same host transport as the one onto which a USB DAC is connected.

This time my experience is not positive. Not negative either, actually, but I could not perceive any “further” improvement over the one obtained by the first unit – the one just plugged in-line between the host and the DAC. This happens on any one of my hosts, be them the small ARM SBC’s or the “noisy” gaming laptop.

Comparisons

iFi iSilencer+ (€ 59)

iSilencer is marketed as a device pursuing totally similar aims as JitterBug FMJ, so we can see it as iFi’s direct alternative to it. I had the opportunity to test a (few of) iSilencer unit(s), and I must say that, unlike Jitterbug FMJ, they did not hit the spot in my case.

Sadly, in my environment iSilencer wasn’t merely transparent (read: useless) but actually made sound worse: it fundamentally makes tones brighter, depressing mids and bass, reducing stage depth and making imaging worse.

iFi iPurifier 3 (€ 129/149)

iPurifier 3 is another device falling in the general “digital signal filters” category, but instead of removing carry-over electrical noise it focuses on signal timing – which is something on which JitterBug FMJ is only “consequently” involved.

I will soon release a piece about iPurifier3 but long story short: (in my setup) it does work. I’ve in particular been using it in-between one of my ARM-based Roon bridges and the Questyle CMA-400i desktop DAC-AMP, and it carried an audible improvement in terms of better treble notes definition, and perceivable better room size definition.

What’s even more interesting is that iPurifier 3 synergises positively with JitterBug FMJ: if I plug JitterBug FMJ on the ARM’s USB port, and iPurifier 3 on CMA-400i’s USB input port, I get both improvements at the same time. Very nice!

Conclusions

To me, JitterBug FMJ works – and very well so. It makes now standard part of my home setup, and I see no downsides to its adoption as an in-line USB channel filter, also considering its quite modest price tag.

A great thank you goes to AudioQuest for providing me with 2 more units (in addition to the one I earlier had already personally purchased) to allow me for extensive testing in multiple configurations.

Thanks a bunch to coblogger Kazi for the nice title image, too.

The post AudioQuest JitterBug FMJ Review – It’s Not About Dancing… appeared first on Music For The Masses.

Cons — Questyle M15 drains the host’s battery when in high gain
– Somewhat narrow staging
– Unforgiving nature might not suit the bright or peaky IEMs
– Slight upper-midrange glare when driven from phones
– No volume or playback controls
– Prone to RF interference

INTRODUCTION

Had I been a YouTube reviewer, I would have littered a ton of “fire” emojis across this review title. The thumbnail would allude to something akin to “shut-up and buy it”, while a somewhat disturbing image of my agape face would round-up the level of bewilderment and bemusement that the M15 has evoked.

Sadly, in the written form, I am but slave to the words.

Questyle M15 is the flagship dongle in the brand’s lineup, and overall, perhaps the best dongle one can buy. Sadly, such sweeping generalizations do not favor anyone, and everything is reliant upon context.

So, this review is to contextualize the reasons why the Questyle M15 might be the best dongle ever, or why it may not be the right dongle for certain use-cases. Read on.

Note: the ratings given will be subjective to the price tier. Thanks to Questyle for sending the M15 for evaluation.

Price, while reviewed: $250. Can be bought from Questyle’s Official Website.

PHYSICAL THINGS AND USABILITY

PACKAGING AND ACCESSORIES

The packaging is minimal, while the accessories are basic. You get a type-C to type-C cable by default. For Apple users, the lightning cable is sold as a separate bundle for USD$20 premium. There is also a nice leather case that you can purchase separately.

BUILD QUALITY

Questyle opts for a CNC-milled aluminum chassis with a see-through acrylic window for the M15’s design. It’s a simple yet effective design decision to go for a see-through top, as it makes the M15 stand out without going overboard. Questyle is not new to this, as many of its desktop products offer an acrylic top for those so inclined.

In terms of inputs and outputs, things are decidedly simple. The type-C port allows USB connection while the 3.5mm and 4.4mm jacks offer unbalanced and balanced connections respectively. The balanced output sounds markedly better as an aside, but that is the case for nearly every dongle that offers a balanced output.

There is a button on the side for gain control, and that’s about it. No volume or playback buttons are there which might be an issue for some. There are two LEDs on the PCB that shines through the acrylic, one for gain level and another is the file type indicator.

Overall, a very simple yet elegant design that panders to my inner-geek thanks to that PCB that’s been laid bare.

HANDLING

At 61.8mm X 27.2mm X 12mm dimensions, the M15 is not the most innocuous of dongles in terms of size. However, I find it to be fairly practical on the desk and the low 25g of weight makes carrying it around easy enough. Even after prolonged usage, the M15 does not get hot which is another plus.

QUESTYLE M15 INTERNALS

Questyle has a knack for making pretty PCBs. Even the desktop DAC or amps have exceptionally clean PCB layout, and the M15 is no exception. Thanks to the acrylic window, all of it is in plain view. Apart from the ES9281AC DAC chip and the aforementioned status LEDs, you can also see the two SIP (system-in-package) current mode amp modules. Each module handles one channel.

There is also a TOREX power management unit that keeps the M15 inactive when no music is playing. In terms of specs, you get a really respectable 0.0003% THD and <-130dB SNR. Then you notice the output power specs and things just do not add up. A measly 22mW into 300ohms? Surely that cannot be right?

In terms of the actual “sound pressure” produced, that indeed seems to be misleading. The M15 can drive most dynamic driver headphones and nearly every single IEMs out there. Only issue is that for best performance, you need to use a laptop as the source. The higher current from the USB ports enable greater dynamic swings.

Speaking of dynamic swings, the SE out can have almost 2Vrms voltage swing from the single-ended out, and about 3.8Vrms from the balanced out. You can connect the M15 to a pair of powered monitors in a pinch and use it as a DAC/pre-amp combo. Just make sure to put the volume at max on the DAC side.

All in all, respectable measured performance, except for the amp specifications which do not really add up to real world experience.

TONALITY AND TECHNICALITIES

As always, it’s difficult to simply talk about the “tonality” of a source gear rather than specific pairing notes. That being said, the M15 has certain “characteristics” that shine through no matter which IEMs or headphones you connect on the other end.

The first thing you notice is the resolution, and how easily the M15 delineates between instruments. Rest assured, the amount of perceived details on the M15 eclipses any other dongle under USD$300. Minute details are pushed to the forefront, making it easier to analyze and dissect tracks. If it’s resolution and precision you want, M15 is practically peerless.

Such hyper-realistic rendition comes at the cost of two things: spatial qualities, and a tendency to be ruthless with poorly mastered tracks or bright/shouty gear. The M15 is unforgiving, though the lack of “etchiness” in the treble and upper-mids make it a potent option for borderline bright IEMs and headphones. The staging won’t be engulfing or stretched outwards, like it can be on some of M15’s peers.

Dynamics are good in terms of macrodynamic punch, though microdynamics are not as evident as they are on certain desktop sources (or even Questyle’s higher-tier DAPs).

Finally, the power output is ample for practically any IEMs out there. When connected to a laptop or desktop, the M15 is too powerful for most IEMs, in fact. I routinely found myself lowering the gain and/or lowering the volume on the desktop side. This is still not enough for power hungry monsters like Hifiman’s HE-6, for example, so for the pesky planars, you still need a more substantial setup.

PAIRING NOTES

I’ll try to keep this section short and sweet.

IEMs that paired well with the M15: most of them, but highlights include Sennheiser IE 900/200/300, SoftEars Turii, Final E3000/A5000/E4000, JVC FW1800/FW10000/FDX1, Campfire Holocene/Andromeda 2020/Solaris.

Headphones that paired well with the M15: not the absurdly power hungry planars, including the likes of HE-6 (and Susvara, by extension, though I fail to understand why anyone would try to run Susvaras off of a dongle), Sennheiser HD800S (too bright), and Beyers (same issues as the Senns). The HD650 had a good pairing though it lacked the liquid smoothness you get off of tubes or high output impedance sources.

Hifiman HE-400i and Arya sounded exceptional through the M15, and if you own the Arya Stealth (or even the newer Arya Organic), the M15 will be more than enough to do justice to their resolving prowess.

SELECT COMPARISONS

I have pitted the Questyle M15 against every single “hyped” or well-regarded dongle that has been released so far. None of them are as resolving, period.

Quloos MC01 gets close at the cost of sounding edgy in the treble and artificial throughout. Apogee Groove has better rendition (and sense) of space, but it sounds a bit veiled in the bass and treble comparatively. The Cayin RU6 are too smoothed out, while the Cayin RU7 opt for a more relaxed, engulfing, and timbrally-accurate presentation than going after raw details.

Lastly, the L&P dongles (W2 and W4) do better in terms of microdynamics but fall flat in every other aspect. The output power is lacking compared to the M15, and once again – not as resolving.

CONCLUDING REMARKS

I received the Questyle M15 back in November, 2022. At the time of writing this review (end of July, 2023) the M15 managed to ward off every single competition by the wayside.

It’s a remarkable achievement in the age of rapid-fire chi-fi releases, where even the parent brand makes its 6 months old “flagship” redundant by releasing something new and “improved”. The M15 is here to stay, and shall remain one of the best, if not the best DAC-Amp dongles out there for the foreseeable future.

The only caveat is the nature of the sound itself – it may become “information overload” for those accustomed to relaxed and laid-back tuning. With certain IEMs, the treble region can sound exaggerated and become bothersome in the long run.

These caveats apply to most, if not all products though, and the M15 achieves the one thing it set out to accomplish: the crown for the most “effortlessly resolving” DAC-Amp dongle out there. Questyle captured lightning in a bottle with the M15, and I hope the spark does not go out anytime soon.

MY VERDICT

4.5/5
THE dongle to beat.

Contact us!

DISCLAIMER

Get it from Questyle Shop.

Our generic standard disclaimer.

You find an INDEX of our most relevant technical articles HERE.

The post Questyle M15 Mobile DAC Review (3) – Dongle Par Excellence appeared first on Music For The Masses.

The latest Mac generation features a relatively good audio circuit that makes many budget dongles and headphone amps obsolete.

Introduction

Apple computers, unlike the company’s mobile devices, have traditionally delivered poor audio quality through their 3.5 mm headphone socket. And most Windows machines are not any better.

Plugging an earphone or headphone into my 2012 MacBook Air and listening to iTunes/Apple Music creates a dull and blunt listening experience. This low quality is a contradiction to the capabilities of digital audio, which goes back at least 20 years. Even the 2013 iPhone 5S had stunning audio quality. Similarly, Apple’s “Lightning to 3.5 mm Audio Adapter” delivers excellent sound quality.

In order to make computers sound better, digital audio pioneer Gordon Rankin of Wavelength Audio (and contractor to AudioQuest), invented the dongle DAC back in 2012, the DragonFly Black. A tiny device without its own battery, it drew power from the host. The DragonFly was restricted to use with a computer as it drew more than 100 mA, too much for iPhone to handle.

As of 2016, the next version of the “Black” was within the iPhone’s current draw tolerance. Many companies jumped on the bandwagon flooding the market with such devices. But not all dongles are equal and ALL of them are a compromise. Such that draw little current (and therefore drain your phone company slowly) have limited power, and the powerful ones empty your phone’s battery fast.

When it comes to dongle-DAC use with computers, current drain is largely irrelevant considering that desktop machines don’t have a battery at all, and notebooks have high battery capacities compared to a phone. Power is therefore no problem with computer application, sound quality has foremost priority.

Dongle DACs typically have 1 to 2 V, depending on impedance. Some, such as the Helm Bolt, automatically switch voltage depending on detected headphone impedance.

Apple introduced a new integrated audio circuit in their latest models:

• MacBook Air introduced in 2022
• MacBook Pro introduced in 2021 or later
• Mac mini introduced in 2023
• Mac Studio introduced in 2022

These models contain an audio circuit that is power wise very similar to dongles such as the Helm Bolt, ifi Audio GO Link, DragonFly Red and Cobalt. Below 150 ohm headphone impedance, the circuit provides a voltage of 1.25 V RMS, above 150 ohm and up to 1000 ohm, the headphone jack delivers 3 V RMS. You find Apple’s respective support article here.

Apple’s integrated DAC supports sample rates up to 96 kHz (just like the AudioQuest DragonFly Cobalt).

I calculated the resulting power and current drain as displayed in the following table:

What we have learnt so far is that Apple’s new audio circuit is as powerful as a standard dongle DAC of the kind that also works with a phone. Yes, there are more powerful dongles on the market such as the Apogee Groove, Questyle M15, or ifi Audio Go bar, which will work better with insensitive headphones.

The 2022 MacBook Air with the M2 processor has its headphone jack on the right-hand side.

Amplification and Sound Quality

Playing in-ears with 32 ohm impedance is a piece of cake, even the 70 ohm Sennheiser HD 25 headphones (on the title photo) get lots of volume. Just like many “standard dongle DACs”, the 300 ohm Sennheiser HD 600 brings the Mac audio to its limits. Yes, it principally works, but it lacks pizazz.

The current-hungry Final E5000 iems are a special case in that they do not run well with most current-conserving dongle DACs and daps, for example the ifi Go link, Helm Bolt or TempoTec V6. They play loud enough but lack bass control, an indication that hey don’t receive enough current.

My testing confirms the tech data comparison with comparable dongles.

And here comes the surprise: the sound quality of Apple’s new audio quality is…astonishingly good for what it is: crisp, transparent, clear. Very surprising. It sounds very similar (in terms of quality) to the ifi Go link or Helm Bolt.

Concluding Remarks

Apple’s new adaptive audio circuit finally sounds quite decent. Poor audio circuits in computers were the reason for the invention of the dongle DAC. The idea was to keep it small, down to the size of a thumb drive.

Some basic $50-100 dongle DACs have now become obsolete for modern Macs, which makes these devices even more compact. Where Apple are exaggerating is with the compatibility with high-impedance headphones. It principally works, but you are better off with a dedicated, powerful, headphone DAC/amp such as the ifi GO bar or the Questyle M15 on the go.

Testing this is easy: if you have a new Mac, try it out. If you don’t have one, don’t bother as you won’t buy one for this purpose anyway.

Until next time…keep on listening!

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Disclaimer

Our generic standard disclaimer.

The post Using Headphones With Your New Mac [Without An External DAC/Amp] – A Review appeared first on Music For The Masses.

At-a-glance Card

Features and description

Externals

GO Link is seriously small and lightweight, really barely more conspicous than a mere passive USB cable, and much more flexible thanks to the braided structure given to its external wires.

It comes with a single LED on the chassis, which is supposed to indicate its power ON state while idle, and the resolution of the digital content being played while working, according to the following scheme:

On the main chassis a single 3.5mm phone output port is available. On the opposite end of the white braided cabling, a single USB-C male connector is present.

Lastly yet very importantly GO Link comes bundled with 2 USB accessories: a USB-C to USB-A adapter, and a USB-C to Lightning adapter.

The Lightning adapter in particular is a quite expensive item when purchased separately, and it’s a crucial resource for Apple ecosystem users as it perfectly matches iPhone and iPad requirements, thus avoiding them the need for the bulky Apple Camera Adapter.

Internals

Not diverting from their habits, iFi releases just macro-information about GO Link internals, but hardly any finer details.

GO Link is built around an ESS SOC-class chip, namely the ES9219MQ/Q (data sheet) which, within the inherent limitations of a SOC chip, is a quite interesting model. It features for example “QUAD DAC+” internal structure, which iFi exploits big time within their smart “S-Balanced” scheme (see more below).

Another ESS9219 feature which GO Link puts to good use is Dynamic Range Enhancement (“DRE”). In GO Link case this results into a 6dB DR enhancement, which is quite significant for a product in this price class.

The rest of the components are coming from the usual manufacturers normally enlisted by iFi, first and foremost Murata and TDK (capacitors).

Input

GO Link is a “pure DAC/AMP dongle”, so its sole accepted input is digital, via the USB connection.

As previously mentioned, the package includes a native USB-C male plug, a USB-A adapter and a Lightning adapter, which considering the very modest total asking price are a huge added value.

Output

GO Link’s sole output is its analog 3.5mm connector, which of course accepts any 3.5mm single-ended terminated load… and something else, thanks to iFi’s S-Balanced technology implemented behind it.

I already covered this iFi proprietary technology multiple times on my articles regarding other iFi sources (here, here and here). You can alternatively refer to iFi’s own white paper, here.

Simply put S-Balanced is a smart way to route the analog signal inside the device such to realise a sort of (give me some rope here) “fake” balanced scheme even with a single amp stage (vs two independent, parallel ones as they are used within “real” balanced schemes).

S-Balanced offers the same benefits of a true Balanced scheme in terms of lower noise and distortion, and none of its drawbacks (to know why… RTFWP!) when it is connected to a balanced-wired (TRRS) load.

In addition to that, unlike a true Balanced scheme, S-Balanced is fully backwards compatible with single-ended (TRS) terminated loads.

And more: single-ended (TRS) terminated drivers plugged onto an S-Balanced port will get “some” improved sound quality (50% reduced crosstalk) compared to the same dac-amp circuitry organised following the “ordinary” single-ended flow.

So you understood well: GO Link’s 3.5 output port (same as Nano iDSD Black Label’s, for that matter) does accept both single ended and balanced ended terminated IEMs/Headphones.

But… how do I connect a balanced-terminated cable (typically ending in a 2.5mm or 4.4mm male plug) to GO Link’s 3.5mm phone out? With an adapter of course !

Now for the really odd part: iFi does not provide such adapter – neither within GO Link’s standard bundle, nor even as a separately-purchaseable part.

Be as it may, the 3.5 mm TRRS wiring you want is what is also called “Hifiman standard”. See here for reference.

I do have such an adapter (3.5mm TRRS male, 4.4mm TRRS female) from back in the days when I was using the Nano iDSD BL, and I can relate the “trick” does work: pairing the same balanced-wired IEM to GO Link once via a 4.4 to 3.5 TRS single ended adapter, and then via the 4.4 to 3.5 TRRS adapter, results in an obviously improved sound in the latter case.

Host power requirements

GO Link absorbs 1W = 450mA (max) while playing and 0.7W = 375mA while idle.

These are not the lowest figures in the industry – the champs always being the Dragonflys here, followed by E1DA 9038SG3 and 9038D – yet these are still to be considered quite modest values, which won’t make too quick shame of your phone’s battery. Yes, iPhones included (via the Lightning adapter supplied as a freebie).

Volume and gain control

GO Link does not offer a physical on-device volume control. Its volume control is interfaced with the one on the host platform (I positively experienced Windows and Android).

The situation is potentially problematic on Android which – by default – divides the USB device volume range in only 40 steps (or even 25 for the latest Android releases…), resulting in the last ticks converting into way too big SPL variations.

Luckily, those better featured player apps (e.g. UAPP) which you would normally anyway use for a number of other reasons one above all bypassing standard Android audio drivers do also allow for re-defining the number of steps Volume control is divided into (up to 250, on UAPP) – which solves the issue.

Other features

MQA Rendering

I won’t spend a word on what MQA itself is, of course. Google around if you wish and you’ll be overflooded with info.

What matters here is: GO Link is a “MQA Renderer”, so it can fully unfold MQA tracks on its own hardware, which is an upgrade vs the default represented by having the music player host do the unfolding, and only limited to the first 2 folds.

Firmware

Like all other iFi devices (well at least all those I know of, but the list seems quite comprehensive…), GO Link allows for the user to easily change/upgrade the device firmware.

At present time, on GO Link this option can only be actionated with the purpose of installing incremental firmware updates aimed at feature optimisations or bug fixing.

Package

I already covered this en passant above, but for the sake of completeness: GO Link comes in a small box but with the right bundle accessories, and premium quality ones at that too.

• USBC-Lightning passthrough adapter
• USBC-USBA passthrough adapter

Sound and power

GO Link sound quality is basically in line with what one would expect from a latest generation ESS SOC chip – such as what you get off of a Shanling M0 Pro, which I recently reviewed – with some little bit of further enhancement.

Who wants to go the extra mile is cordially encouraged to get a TRRS 3.5mm adapter and plug balanced-terminated drivers onto the GO Link. The improvement in terms of noise reduction, clarity, stage drawing and imaging will be quite obvious.

GO Link’s output power is not huge yet not weak either: it delivers 2V on high impedance loads (300+ ohms) and 1.5V (70mW) onto 32 ohm loads, making it an absolutely viable option for 95+% of the IEMs out there and much of the cans, too. Just avoid planars, and particularly current-hungry IEMs (E5000, B1…), and you’re a happy camper.

Considerations & conclusions

With the GO Link, positioned at just € 59 EU retail, iFi clearly aims at marketing a ultra-portable, not-necessarily-audiophile-tier “smartphone audio upgrade” device, offering all users of no-wired-analog-out phones an option to plug their wired drivers, and a sound quality and/or output power upgrade to all others.

After assessing it I can relate that iFi indeed went much, much beyond such intention.

While surely south of the top dongle market levels reached by the likes of Questyle M15, Apogee Groove, Dragonfly Cobalt and iFi’s own GO Bar, iFi GO Link offers very interesting output quality and power, full MQA rendering, instant matching with Windows, Android and/or Apple hosts, and even an extra option to pair with balanced-terminated drivers (via a third party adapter) for even wider compatibility, and even better sound.

What more an occasional user might ask for € 59 retail I frankly wouldn’t know.

Our generic standard disclaimer.

The post iFi GO Link Review (1) – What More ? appeared first on Music For The Masses.

It is customary for us in these cases to write a rather succint piece to avoid too much replication of the previous article’s contents but I’m going for an exception here. These little ones do in facts make me feel compelled to share my extended opinions with my few readers. I know, it’ll be boring. Few readers, however, means little damage. So let’s just get down to it.

Just for the record: Akoustyx S6 are currently on deeeep discount sale (like: 50% off) on Drop.

The manufacturer’s official page is instead here.

At-a-glance Card

Full Device Card

Test setup

Apogee Groove / Sony NW-A55 mrWalkman / Questyle QP1R / Questyle M15 / Questyle CMA-400i – Final E tips – Stock cable – lossless 16-24/44.1-192 FLAC and DSD 64/128/256 tracks.

Signature analysis

Tonality

S6’s timbre is natural, sculpted, muscular and well bodied, and stays so all across the spectrum. There’s above decent microdynamics, and no sign of “artificial” aftertastes. This, alone, would be worth closing my article with a glowing rec.

S6’s tonality indeed deserves some articulated comments. The manufacturer underlines S6 are intended for “studio reference acoustics”. Talking through with them they reported they tuned them closely following the Harman IE2017 target (see below) – and I must say it does show, big time. The presentation I hear from the low mids all the way up is definitely that. Sub-bass elevation is only more modest on S6 compared to the theoretical target.

Simply put IE2017 is not my personal preference, period, and this for two main reasons.

One: the circa 11+dB value gap between the 1Khz and 3Khz points results in highmids being slapped hard into my face when I raise volume beyond a very moderate level, and

Two: the depressed lowmids values convert in a very dry, too dry tonality – I do prefer bright neutral to warm balanced, but IE2017 is below neutral, it’s almost aseptycal.

This has to do with the target itself. Then, depending on the particular driver technology and/or implementation accuracy or lack thereof on this or that driver the “actual” result will be for me moderately bad, very bad, or downight unbearable.

Now, the good news is that planar drivers in general bear tonality corrections by means of equalisation with a certain ease. And, S6’s driver is very flexyble (pretty much in Audeze iSine or RHA CL2 league), so first thing I did of course was bringing tonality more in my preferred ballpark, and a bit off the effing IE2017 “thing”.

Here’s the scoop:

My experience with S6 refers to the first 2 corrections (3K and 4K) imperatively applied. I will outline differences when the optional ones are applied too.

Sub-Bass

S6’s sub-bass is fully extended and quite present. Typical snappy planar transients apply without distortions here so there’s little more to squeeze off the onion so to say. That said those who prefer an even more visceral rumble can experiment with a Low Shelf correction from 50-ish Hz, +3dB (or more) and a very wide badwidth (0.3 or so).

Mid Bass

S6 midbass is seriously good. Oh well, of course it’s good if you like technical acoustic bass as I do. Distorted overbloated bass lovers should never buy S6, period.

Transients are well managed here and while they stay in fast, precision-rendering territory as you expect from a planar, they are not overly snappy and do deliver some body and microdynamics.

Applying the aforementioned warmth correction (Peak 200Hz +2dB wide bandwidth) does exactly what it says: midbass (and not only) will heat up a good 20%, coming across as a bit more bodied and flowery.

Mids

Mids are spectacularly sculpted yet organic and detailed. Guitars and tenor sax benefit most of the situation delivering good nuances and microdynamics while staying precise and seprated (see Separation below).

Highmids is where the IE2017 – and S6 which follow that very closely – loses my personal approval and that’s why in my books S6 strictly require the EQ-based retuning I mentioned above.

Once that’s done however the magic happens in all its splendor: high mids are energetic, detailed, sparkly and controlled (!!), even when you pump the volume up significantly, which is indeed a way to open the presentation up and let S6 give its musical best

Male Vocals

Tenor vocal lovers will be those finding the Wamth correction (see above) most desireable. Without that there’s too much dryness to call delivery really organic.

Female Vocals

Female voices are natural and organic, although not flutey. Good texture available and good microdynamics for a planar.

Highs

On a corrected S6 trebles are integral part of the nice show. Well extended, quite airy, snappy without excesses, not zingy, not tizzy. Love them. Apply the “Extra Air” EQ correction to add further airiness. Don’t be scared nor shy: experiment. Try +3dB, +4dB, +8dB if you want. Only stop where you like the sound best : S6’s driver shall anyhow follow you like a doggie.

Technicalities

Soundstage

S6 cast a nice sized stage both accross and in depth – a bit more or a bit less depending on fit depth (the deeper the fit, the smaller the room).

Not the absolute widest projection I heard but very good anyway.

Imaging

Macrodynamics are beyond good. Intruments and voices are well scattered on the scene and there is nice air inbetween

Details

This is a point of excellence. S6’s detail retrieval smoothness is easily top rank for my experience on sub-500€ drivers. While I can name other “detail monsters” around, they all will “cost” some or a lot of fatigue and, before that, distraction from the music flow. S6 deliver fine and subtle details without slamming them onto your face nor covering you with “metallic noise dust” as other much leess refined drivers do.

Detail is also good from bass, although to a lesser extent: down there I guess planar-snappy transients do represent an apriori limitation to low frequency microdynamics. Something can be obtained with some light EQing but that’s it. Not “bad” however, just not so outstanding as to point it out as key plus. If you want special bass articulation and nuances get a high end DD.

Instrument separation

S6 execute separation very well. Crowded passages stay perfectly readable at all times, thanks to very controlled transient behaviour, and that glowing compromise mentioned above between snappyness and microdynamics.

Layering is top class: you can follow second or third voices with ease at all times and that’s not trivial to get – at any price, let alone with this small budget.

Driveability

In the “planar drivers” world S6 are probably the easiest to bias I found. You can even drive them from a phone, although you won’t have much headroom to compensate for low recorded materials (e.g. some vinyl digitisations, DSD conversions etc).

That said, their presentation opens up to more details and microdynamics when submitted to somewhat higher power. Once I apply my EQ corrections and the highmids excess goes for good, S6 offer a wonderfully smooth SPL progression. Indeed I find that even “dangerous” insofar as they cease any shouting, and you don’t get any “too high volume warning” so to say when pumping them up. Be careful… we all have only one hearing system you know that, once screwed you can’t fix it…

Like any bright/bright-neutral drivers S6 pair best with relatively warm sources, or at least with non-analythical ones. A special mention for Groove: the pairing with S6 is beyond spectacular.

Finally, a last important point of attention regards the equalisation requirements: your source need to be capable of at least “some” EQing.

Physicals

Build

The cylindrical part of the housings is in titanium alloy. The supersmooth outside finish is a titanium-oxyde based treatment. The backside is realised of a special polymer, in angled shapen, to properòy house the MMCX connector. Very stylish at least in my tastes, and covered by some patent too.

A red/blue colored ring helps easy identification of the right/left piece. Depending on fitting that ring might end up covered by the Earlock structure (see Fit below), however.

Lat but not least S6 housings are extremely lightweight: once selected the right size/type tips, and worn on with Earlocks etc they virtually “disappear” from your perception. Superb.

Fit

Stock silicon tips are not bad for the job. It’s not so easy to rotate others in as S6 nozzles are quite slim. In the end I settled for Final E (black) as they tend to tame trebles and bring mids forward a bit, which of course helps on re-estabilishing my preferred balance in this particular case.

Technically speaking a good alternative would be Spiraldots too, but their stem diameter is too wide so who wants to adopt them onto the S6 must be ready to apply a tight rubber ring onto the nozzles first, then the tips. Couldn’t be fussed personally, as I found Type-E’s more than good enough.

As you may reacall I am not in general a foamies lover but S6 is one of the few exceptions: here the typical foamies effect (“combing” thinnest treble notes and making bass a whiff “matte”) resolves in a very pleasant timbre nuance alternative.

Once again stock tips are of very good quality – very soft and quickly reactive material, classic cylindrical style – so you can easily go with those to begin with. My effing left canal is always creating problems though so in my particular case S stock tips is too small and M is too thick :-/ My best option is Comply TS200. YMMV needless to say.

Last but certainly not least in importance: the Earlocks. Those are totally brilliant. Think to the IEM version of those “comma shaped” rubber thingies you fit onto earbuds to help the stay put in your concha – and add twice the design accuracy.

These EarLock® silicon “rings” realise several contact points on the outer ear to (literally) lock the housings in place and fit the same way every time. The item comes from a company focusing on hearing protection aids for people involved in very loud noise situations (including law enforcement, army etc) aiming at guaranteeing that the noise attenuator/plug/intercom – whatever stays in-ear – won’t ever budge let alone pull out even in case of sudden hard movements, pullbacks, rush etc etc. And boy do they work!

Simply put: the Earlocks (provided in 3 sizes S-M-L) fit perfectly and “disappear” in/onto the outer ear, I don’t even perceive them as being there once worn, and S6 housings get a 100% firm stability in place, whatever I do however I move etc. This not only means that they won’t entirely slip off, but also and probably even most importantly that they won’t budge even as a consequence of mandibular movements while talking or eating which – in my case it does happen – may produce loss of seal and/or need to reposition.

Long story short: now that I tried them I want something similar for all my IEMs !

Comfort

Subjective differences apart, bullet shapes are normally considered “comfortable”. Amongst their downsides there’s typically stability which is totally fixed by the Earlocks in this case (read above). S6 are not particularly “long” in the bullet shaped category however they do support mid-deep fit, as a free choice user option.

As always: the deeper the fit the softer the trebles, the more relevant the bass, and the narrower the stage. Pick your poison

Isolation

Using foam tips and Earlocks to guarantee stable fit, S6 reach a whopping 34dB passive ambient noise reduction (NRR 28dB). That’s a lot! We are in professional NR aides territory indeed – these values are indicated for people working on tractors for example, or in some noisy industrial plants. Fantastic. Just be careful walking outside : you won’t hear traffic (!)

Cable

S6 stock cable is an unassuming-looking yet very sophysticated 16 core Oxygen Free Copper conductor. According to the manufacturer it is accuratly impedance-paired with the drivers. Be as it may, it sonically pairs spectacularly well with S6. I tried rotating some others – OFC is definitely the right choice, SP-OFC adds on edgyness which is not required here, Grafene does not pair well either.

As it often happens on low budget packages the cable has a fixed 3.5 termination only (the company is working on a multi-plug alternative to bundle on future versions but that’s on the drawing board yet).

Considering how well the cable pairs with S6 I recommend swapping only to those who are in dire need as all their sources sound best exclusively from their balanced otuputs. In such case a very inexpensive, decent option is the good ol’ ultracheap NiceHCK 16 core High Purity Copper (aka “Ugly Cable”). Alternatively a Linsoul HC08 will do well. Or, wait for Akoustyx to deliver their own

I guess something more is also worth saying about the cable.

One: the Kevlar sheath may easily be a love/hate thing. The material itself is beyond wonderful, super resistant etc. On the down side it’s badly microphonic (which is probably why the manufcturer strictly recommends over-ear cable install – RTFM…) and it’s quite springy at first. For the latter issue the good news is that the sheath gets obviously softer and malleable after a quite short time.

To quicken such “break-in” period you can frictionate harshly the cable in between your hands after coughly “coiling” it – don’t worry it won’t break – do it a few times and it will already get much better.

Two: the MMCX connectors offer a very firm “click-in-fit”. This may sound like a detail but for my experience it is not (!). Without going too far, this is one of the very few points of structural weakness I underlined on my Miyabi analisis (here). The down side on low quality MMCX options is of course micro-discharges resulting in subtle craclking noise while listening or worse.

Don’t take me wrong here, I’m not saying S6’s stock cable is the one and only good cable out there – I’m just saying don’t discard it quickly replacing it with “just any other one”, as – unlike what too often happens with cables bundled with budget-tier drivers – Akourstyx put a good one in here…

Specifications (declared)

Comparisons

7Hz Timeless ($ 199 Drop deal)

Simply put, S6 are miles better. Timeless have bloated, untextured midbass, a generally artificial timbre, scarce microdynamics (aka invasive “planar timbre”), very modest layering and separation. They also don’t seem to react particularly well to EQing, although some correction do make them a bit better. They do cast a wider stage compared to S6, there’s that. And they are more expensive.

TINHIFI P1 ($ 169)

P1 offer a smooth, nicely balanced and inoffensive tonality. Possibly a bit “too inoffensive” – one of their limitations for my tastes being that I find them a bit boring. S6 are obviously sparklier, much more engaging energetic and “brilliant” – they do require EQ correction ootb however, which is not an “absolute requirement” for P1 instead. Other major differences are the timbre – P1 being desperately “planar” vs S6’s much better microdynamics – and the driveability – P1 is much harder to bias.

Ikko OH1s ($ 74 promo on Amazon.com)

Recently price-repositioned by Ikko (I’d like to think: also after our suggestion), OH1S are based on different driver tech (1 DD + 1BA) but offer a general presentation and tonality similar to S6.

OH1S don’t require EQ corrections to deliver good bass, mids, vocals and some technicalities – all coming close to S6, which still has the edge on pretty much all counts, even if sometimes by not much. OH1S fall more evidently short of S6 in terms of imaging, and most of all energy. They are also very much tip dependent, and may not be so easy to fit.

final A3000 (€ 129,99 on Amazon.it)

By far my sub-300€ clear-timbre, bright-neutral tonality reference. A3000 are built on a custom-developed DD essentially sounding like a planar, and specially tuned prioritising equal clarity on sounds both closer and farther away from the listener position – which is particularly beneficial to acoustic music from large orchestras or groups.

As a direct consequence A3000 win big on sounstage drawing vs S6 – and pretty much any other sub 1K$ driver I heard tbh, solely bar their siblings A4000, which I find however less pleasant for my tastes on other counts (won’t digress here).

Tonal homogeneity, phenomenally nailed compromise on details vs musicality on trebles, layering proweness and well calibrated snappy transients are on par between the two. S6 offer higher note weight and whith that a more energetic, muscular, lively musicality while A3000 are obviously silkier. S6 sound if you wish… american, while A3000 so japanesely discrete-yet-deeply-sophisticated.

A3000 do not “require” EQ out of the box, their few shortcomings however can’t easily be fixed by EQing. Opposite situation on S6, which need to be put hands onto, but can be EQ-pushed/pulled/stirred in so many different sonic flavours, such argubaly being their most solid upper edge.

Considerations & conclusions

Building low priced, low quality products is not too complicated. Building equally low priced products carrying some more quality as to trigger a user’s attention on “price/performance compromise” grounds is already a bit less easy. Building, again, equally low priced products featuring however the same quality of a market-top product and just scaled-down features… this is a challenge. Taking and winning it requires serious, original industrial competence.

Some 2-3 years ago I auditioned my first planar IEM and I was kinda puzzled. Then I heard another. Then another. And I gave up. Most of all, they were drowning me into “planar timbre”, i.e. [almost] complete lack of microdynamics. A total turnoff for me. Simply put, I could see no reason why one would prefer one of those to a much more expressive and/or refined fast-transient DD or (quite rare, on low budgets) good BA.

Then in spite of my disappointment for the category last year a friend convinced me to audition a pair of RHA CL2, and that’s where I finally “got” planars: different beasts, indeed. And not at all “inexpressive” as the previous ones I tried.

Too bad that a) those CL2 babies cost a pretty penny, and what’s worse b) they are not in production anymore. “Alright too bad” – I said to myself. At least now I know “what” I look for “can” exist in a planar IEM, and that I was right on disregarding lower rank / quality alternatives.

Finally, in came Akoustyx.

Simply put, their S6 are truly hightech planar drivers built into a scaled-down, very modestly priced, stellar value package.

I sharply disagree on the apriori choice which as been made in favor of the IE2017 target. In my very modest personal opinion I don’t find it neither studio-neutral/reference, nor pleasantly musical. I was even more disappointed about stock CL2 tuning, however !

The outstanding things with S6 are their spot on native timbre, and their great elasticity vs EQ corrections.

No they do not deliver “precisely the same” technical proweness I heard on RHA CL2. They come seriously close however, with that indeed representing a credible, significant, differently flavored alternative to DD or BA technology budget drivers – that is, at a fraction of CL2’s price.

If you ask me, S6 are indeed worth their full 250$ MSRP, and then some. At their current deal price on Drop ($129) they are on “steal” category.

Our generic standard disclaimer.

The post Akoustyx S6 Review (2) – …This ! appeared first on Music For The Masses.

The ifi Audio Go bar was supplied by the company for my review – and I thank them for that. You find more information on the Go bar’s product page.

Introduction

My first portable DAC/amp was (and still is) the ifi Audio iDSD nano Black Label (which I treated in my hip-dac review). It was a safe buy as it had won quite a few awards – and it is still available. The nano BL is a microcosm of what ifi Audio stands for: classic shape, integrated IE Match, house sound. It is still my standard for headphone measurements.

The nano BL is a rather blocky device with limited portability, it is rather transportable. It is reasonably powerful and can drive headphones up to 300 ohm easily.

In contrast, the Go bar is a small device as it does not contain a battery. It draws its current from its source, that is a phone/tablet or a computer. This has its pros and cons as will be discussed below.

Alberto has already taken apart the Go bar’s technical and functional aspects in very great detail. I therefore would like to add and give my 5 cents where we differ – and possibly simplify some details. After all any review is to a large extent subjective.

What is most important to me is functionality, especially when it comes to miniature devices. How does the Go bar fare in the wide field of applications I’d like to use it for?

Specifications

Physical Things and Usability

The Go bar probably received its name from a pun derived from its portability (“Go”) and its form factor (“gold bar”). And it has the dimensions of a Bounty chocolate bar. In the box are the Go bar with a nifty leather case, 2 high-quality OTG (“On The Go”) cables with adapter, and the paperwork. That’s all you need to connect the Go bar to any computer, Android device, even iOS devices. No other accessories required.

The build quality, haptic, and mechanisms of all parts are outstanding. The chassis is made of alloy, workmanship is impeccable, the the button mechanisms are precise. Same with cables and USB adapter, which feel premium. Physically, the Go bar is high end.

Functionality and Operation

As to the Go bar’s “standard staples”: it features two circuits, a single-ended 3.5 mm one, and the ever emerging balanced with 4.4 mm socket. Although the 3.5 mm is S-balanced, the true 4.4 mm balanced circuit has better generally better specs and is more powerful. Try using mainly this one, that’s where the Go bar’s value is.

The Go bar features a 16-core XMOS micro controller with proprietary firmware to optimize the analog output quality through synergy with the Cirrus DAC. It features a precision clock to minimize jitter.

There are 4 different digital filter options available to minimize unwanted sonic artifacts:

• BP’ (Cyan): Bit-Perfect: no digital filtering, no pre or post ringing
• ‘STD’ (Red): Standard, modest filtering, modest pre and post ringing
• ‘MIN’ (Yellow): Minimum phase, slow roll-off, minimum pre and post ringing
• ‘GTO’ (White): Gibbs Transient-Optimised: upsampled to 352/384kHz, minimum filtering, no pre ringing, minimum post ringing

Ringing relates to an unwanted echo effect before (pre-) and after (post-) a note. Post ringing is actually a normal artifact of human hearing, pre ringing is not. Many claim pre ringing is not audible. This is a tricky topic and you are advised to rely on your ears.

In some aspects, the Go bar is the most complete dongle on the market as it has functionality no competitor offers: IEMatch, S-balanced, XBass, and XSpace.

IEMatch is an extremely useful tool for low-impedance iems in that it increases output impedance done by resistors dampening the amplifier. It removes hiss from very sensitive iems, for example the 16 ohm Dunu Zen. Check out Alberto’s detailed description of IEMatch in his Go bar review as well as his article dedicated to this tech feature.

XBass elevates the frequencies close to the sub-bass, adding a dry punch which can be quite enjoyable. The company calls it “an analogue bass boost to ‘add back’ lost bass response for more accurate reproduction of the original.”

XSpace, as you could imagine, adds headroom. It is, in their own words “a holographic sound field to open up your music to give you the spaciousness of a live concert.“

S-balanced (Single-Ended Compatible Balanced) means that the listener gets the benefits of a balanced circuit (2 amplifiers) with a normal 3.5 mm TRS plug (also with 3.5 mm TRRS).

Turbo is ifi Audio’s fancy term for high gain: it adds 6 dB to the signal. This is quite impressive considering the dedicated Helm dB12 adds maximally 12 dB.

Last but not least, the Go bar’s firmware is user updatable. It can be downloaded here.

Amplification and Power Management

Power management is not very efficient. The Go bar draws more than twice the current as the AudioQuest DragonFly Cobalt, and 50% more than the comparable Questyle M15. Most iPhones allow only a draw of 100 mA, which is below the Go bar’s 140 mA. My iPhone SE (1st gen.) does work with the Go bar, albeit with greatly reduced power.

In any way will phones not be the ideal partner for the Go bar. Any dongle without a battery can only be a compromise: those with low current draw (AudioQuest DragonFlys) will be easy on your phone’s battery, but may not be able to handle low-impedance/inefficient headphones well. Current hogs like the Go bar will have a more powerful performance, but will empty the host battery fast, or will not work with the host at all.

The best compromise is the Questyle M15, which has both acceptable current draw and much power. ifi Audio’s next step should be to reduce Go bar’s energy consumption, possibly even with a firmware update.

In terms of amplification power, the Go bar delivers 475mW@32Ω and 7.2V@600Ω on its balanced circuit, and 300mW@32Ω; 3.8V@600Ω on its single-ended circuit. Ignoring the single-ended circuit (it should only be used in emergency cases) the superior balanced circuit drives low-impedance iems such as my Final E5000 and Final A3000 very well, and it also handles the 300 ohm Sennheiser HD 600 with ease.

I have not tested any more demanding devices but would have my doubts that Go bar does justice to powerhungry planar magnetic headphones.

Sound

The Go bar follows the tradition of previous ifi DAC/amps in that it has a neutral signature with a light tinge of warmth. Call it “tepid”. It is less warm, more neutral, crisper and swifter than the nano BL. More like the excellent hip-dac. I took my time: >4 months of testing (apologies to ifi Audio).

The Go bar’s notes are like its build: accentuated, articulate, controlled, composed, cohesive, detailed, clean. The sound is from a single mold. The sonic image is of good clarity and detail. Extension at both ends is good however subtle, never overwhelming or intrusive. Treble is “sweet”.

The sound is substantially better with the balanced circuit. Comparing the sonic image to a picture means clear well defined lines with a good depth – and no overpixelation.

Comparisons

Go bar, Audioquest DragonFly Cobalt, and Questyle M15 have one thing in common: lots of proprietary engineering that elevates them from the mass of dongles. All of these are very good but have different purposes and different features. A direct comparison is difficult as all of them, being without battery, are a compromise.

AudioQuest DragonFly Cobalt

This is actually an unfair comparison – unfair to both. The Cobalt is designed for low current draw to work with iPhone, and it is therefore of limited power. It therefore lacks the “greedy” balanced circuit. The Go bar will not work well with iPhone but delivers much more power on the computer than the Cobalt.

Even before it comes to sound quality, the user will distinguish the two based on their different purposes. Both are no real competitors but complementary. In the limited overlap both have (for example 32 ohm earphones), the Cobalt is probably unbeatable, sonically, with its rich, textured, detailed sound. The Go bar is more composed but a bit more analytical, the Cobalt is more “musical”.

Questyle M15

The M15 is a more fitting competitor. It also features 3.5 mm single-ended and 4.4 mm balanced circuits. It cannot reach the Go bar in terms of its superb build and haptic, or in the quality of the included cables. In terms of design, the M15 features 2 standard ESS SoCs with two of Questyle’s own Current Mode Amplification modules.

The M15 has its tonal emphasis in the midrange, the Go bar more in the lower frequencies. I’d assign the Go Bar a marginally better articulation/accentuation, and the M15 a more organic presentation, although both come really close in terms of sound quality.

The biggest difference between the two are the features: Go Bar has XBass, Space, and selectable digital filters – but the M15 has a more effective power management (less battery drain) and works better with iPhone. Also different is the operation: the Go bar bypasses the phone/computers internal volume control completely…it is handled entirely by its buttons.

Both Go bar and M15 are less portable than the DragonFly Cobalt. Alberto also threw the Apogee Groove into the mix, which only works with single dynamic drivers and essentially not with phones. So it is of very limited use. But it offers an unbeatable spatial reconstruction – and no features whatsoever.

In summary, the Go bar may be the most polished and accentuated sounding of the lot, but, as always, it comes down to personal taste, which to choose.

Concluding Remarks

The ifi Audio Go bar is the equivalent of a comprehensive no-hassle package. Everything is of very high quality: the build including the button mechanism, the included adapter and cables, the functionality (including IEMatch, XBass and XSpace), the power, and, of course, the sound.

It is one of these rare things you can buy blind. Just I did with its older brother, the iDSD nano BL. Oh, in the meantime the Go bar has caught up to his older brother in terms of awards.

Until next time…keep on listening!

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Disclaimer

The ifi Audio Go bar was supplied by the company for my review – and I thank them for that. You find more information on the Go bar’s product page.

Our generic standard disclaimer.

The post ifi Audio GO bar Review (2) – Comprehensive No-Hassle Package appeared first on Music For The Masses.

Die ifi Audio Go-Bar wurde mir von der Firma für meinen Test zur Verfügung gestellt – dafür danke ich ihnen. Mehr Informationen finden Sie auf der Produktseite.

Der Author diese Artikels ist in Deutschland geboren und aufgewachsen, hat aber mehr als die Hälfte seines Lebens in Kanada verbracht. Ihm ging das geschwollene Gelaber der meisten deutschen Rezenten auf den Geist. Daher dieser Versuch: ist immer noch etwas hölzern, da aus dem Englischen übersetzt. Die Original Artikel findet man hier.

Einführung

Mein erster tragbarer DAC/Verstärker war (und ist immer noch) der ifi Audio iDSD nano Black Label (den ich in meiner hip-dac-Rezension behandelt habe). Es war ein sicherer Kauf, da er einige Preise gewonnen hatte – und er ist immer noch erhältlich. Der nano BL ist ein Mikrokosmos dessen, wofür ifi Audio steht: klassische Form, integriertes IE Match, Hausklang. Er ist immer noch mein Standard für Kopfhörermessungen. 

Der nano BL ist ein eher klobiges Gerät mit begrenzter Tragbarkeit, er ist mehr transportabel als alles andere. Er ist aber recht leistungsstark und kann Kopfhörer bis 300 Ohm problemlos ansteuern. 

Im Gegensatz dazu ist der Go bar ein kleines Gerät, da er keine Batterie enthält. Er bezieht seinen Strom aus seinem Quellgerät, also einem Telefon/Tablet oder einem Computer. Das hat seine Vor- und Nachteile, wie wir weiter unten besprechen werden. 

Co-blogger Alberto hat die technischen und funktionellen Aspekte des Go bar bereits sehr detailliert auseinandergenommen. Ich möchte daher meine 5 Cents hinzufügen und sagen, wo wir uns unterscheiden – und möglicherweise einige Details vereinfachen. Schließlich ist jede Bewertung zu einem großen Teil subjektiv.

Das Wichtigste für mich ist die Funktionalität, gerade bei Miniaturgeräten. Wie schlägt sich der Go bar in dem weiten Feld der Anwendungen, für die ich ihn einsetzen möchte?

Technische Daten des Go bar

Physische Dinge und Benutzerfreundlichkeit

Der Go bar erhielt seinen Namen wahrscheinlich von einem Wortspiel zwischen seiner Tragbarkeit (“Go”) und seinem Formfaktor (“Goldbarren”). Und er hat die Abmessungen eines Bounty Schokoriegels.

In der Schachtel befinden sich der Go bar mit einer schicken Ledertasche, 2 OTG (“On The Go”) Kabel mit Adapter und der Papierkram. Das ist alles, was Sie brauchen, um den Go bar an jeden Computer, jedes Android-Gerät und sogar an iOS-Geräte anzuschließen. Es ist kein weiteres Zubehör erforderlich. 

Die Verarbeitungsqualität, Haptik und Mechanik aller Teile sind hervorragend. Das Gehäuse ist aus einer Legierung gefertigt, die Verarbeitung ist tadellos, die Tastenmechanismen sind präzise. Das Gleiche gilt für Kabel und USB-Adapter, die sich hochwertig anfühlen. Physisch ist der Go Bar High-End.

Funktionsweise und Betrieb

Zu den “Standards” des Go bar: er verfügt über zwei Schaltkreise, einen “single-ended” asymmetrischen 3.5-mm Schaltkreis und den immer beliebter werdenden “balanced” symmetrischen Schaltkreis mit 4.4-mm Anschluss. Obwohl der 3.5-mm-Schaltkreis “S-balanced” ist, hat der 4.4 mm Schaltkreis allgemein bessere Spezifikationen und ist leistungsfähiger. Versuchen Sie, hauptsächlich diesen zu verwenden, denn darin liegt der Wert des Go bar. 

Der Go bar verfügt über einen 16-Kern-XMOS-Mikrocontroller mit proprietärer Firmware zur Optimierung der analogen Ausgangsqualität durch Synergie mit dem Cirrus DAC. Er verfügt über einen Präzisionstakt, um Jitter zu minimieren. 

Es stehen 4 verschiedene digitale Filteroptionen zur Verfügung, um unerwünschte klangliche Artefakte zu minimieren:

• BP’ (Cyan): Bit-Perfect: no digital filtering, no pre or post ringing
• ‘STD’ (Red): Standard, modest filtering, modest pre and post ringing
• ‘MIN’ (Yellow): Minimum phase, slow roll-off, minimum pre and post ringing
• ‘GTO’ (White): Gibbs Transient-Optimised: upsampled to 352/384kHz, minimum filtering, no pre ringing, minimum post ringing

Beim “Ringing” handelt es sich um einen unerwünschten Echoeffekt vor (pre-) und nach (post-) einem Ton. Nachklingeln ist eigentlich ein normales Artefakt des menschlichen Gehörs, Vorklingeln nicht. Viele behaupten, Vorklingeln sei nicht hörbar. Dies ist ein heikles Thema und Sie sollten sich auf Ihre Ohren verlassen.

In mancher Hinsicht ist der Go bar der vollständigste Dongle auf dem Markt, da er über Funktionen verfügt, die kein Mitbewerber bietet: IEMatch, S-balanced, XBass und XSpace.

IEMatch ist ein äußerst nützliches Werkzeug für niederohmige Verstärker, da es die Ausgangsimpedanz durch Widerstände erhöht, die den Verstärker dämpfen. Es entfernt das Rauschen von sehr empfindlichen Kopfhörern, z.B. dem 16 Ohm Dunu Zen. Sehen Sie sich Albertos detaillierte Beschreibung von IEMatch in seinem Go bar Artikel sowie seinen Artikel zu diesem technischen Feature an.

XBass hebt die Frequenzen in der Nähe des Subbasses an und fügt einen trockenen Kick hinzu, der sehr angenehm sein kann. Das Unternehmen nennt es “eine analoge Bassverstärkung, um verloren gegangene Basswiedergabe für eine genauere Reproduktion des Originals ‘zurückzugeben’.”

XSpace fügt, wie Sie sich vorstellen können, Headroom hinzu. Es ist, in ihren eigenen Worten, “ein holografisches Klangfeld, das Ihre Musik öffnet und Ihnen die Räumlichkeit eines Live-Konzerts gibt.”

S-balanced (Single-Ended Compatible Balanced) bedeutet, dass der Hörer die Vorteile einer symmetrischen Schaltung (2 Verstärker) mit einem normalen 3,5 mm TRS-Stecker (auch mit 3,5 mm TRRS) erhält.

Turbo ist ifi Audios schicke Bezeichnung für High Gain: Es fügt dem Signal 6 dB hinzu. Das ist ziemlich beeindruckend, wenn man bedenkt, dass der nur-Verstärker (ohne DAC) Helm dB12 maximal 12 dB hinzufügt.

Zu guter letzt ist die Firmware der Go Bar vom Benutzer aktualisierbar. Sie kann hier heruntergeladen werden.

Verstärkung und Strom Management

Das Strommanagement ist nicht sehr effizient. Der Go bar zieht mehr als doppelt so viel Strom wie der AudioQuest DragonFly Cobalt und 50 % mehr als der vergleichbare Questyle M15. Die meisten iPhones erlauben nur eine Stromaufnahme von 100 mA, was unter den 140 mA der Go Bar liegt. Mein iPhone SE (1. Generation) funktioniert mit dem Go-Bar, wenn auch mit stark reduzierter Leistung. 

Auf jeden Fall sind Handys nicht der ideale Partner für den Go Bar. Jeder Dongle ohne Akku kann nur ein Kompromiss sein: Dongles mit geringer Stromaufnahme (AudioQuest DragonFlys) schonen zwar den Akku des Handys, können aber möglicherweise nicht gut mit niederohmigen/ineffizienten Kopfhörern umgehen. Stromfresser wie der Go Bar sind zwar leistungsfähiger, leeren aber den Akku des Quellgeräts schnell oder funktionieren überhaupt nicht damit.

Das Beste aus beiden Welten ist der Questyle M15, der sowohl eine akzeptable Stromaufnahme als auch viel Leistung hat. ifi Audios nächster Schritt sollte sein, den Energieverbrauch des Go bar zu reduzieren, möglicherweise sogar mit einem Firmware-Update.

Was die Verstärkungsleistung betrifft, so liefert der Go bar 475mW@32Ω und 7.2V@600Ω in seinem symmetrischen Schaltkreis und 300mW@32Ω; 3.8V@600Ω in seinem asymmetrischen Schaltkreis. Ignoriert man letzteren (er sollte nur in Notfällen verwendet werden), treibt der überlegene balanced Schaltkreis niederohmige/ineffiziente Hörer wie meine Final E5000 und Final A3000 Kopfhörer sehr gut an, und er bewältigt auch den 300 Ω Sennheiser HD 600 mit Leichtigkeit.

Ich habe keine anspruchsvolleren Hörer getestet, aber ich hätte meine Zweifel, dass Go bar den leistungshungrigen planar-magnetischen Kopfhörern gerecht wird.

Sound

Der Go bar folgt der Tradition früherer ifi DAC/Verstärker, indem er eine neutrale Signatur mit einem leichten Hauch von Wärme aufweist. Nennen Sie es “lauwarm”. Er ist weniger warm, neutraler, knackiger und flotter als der nano BL. Mehr wie der ausgezeichnete hip-dac. Ich habe mir Zeit zum testen gelassen: >4 Monate (Entschuldigung an ifi Audio)…”gut Ding braucht Weil.”

Die Töne des Go bar sind wie sein Aufbau: akzentuiert, artikuliert, kontrolliert, komponiert, zusammenhängend, detailliert, sauber. Der Klang ist aus einem Guss. Das Klangbild ist von guter Klarheit und Detailtreue. Die Erweiterung an beiden Enden ist gut, aber subtil, niemals überwältigend oder aufdringlich. Die Höhen sind “sweet”. Der “balanced/symmetrische” Klang ist wesentlich besser als auf dem single-ended/asymmetrischen Schaltkreis. Vergleicht man das Klangbild mit einem Foto, so erkennt man klare, gut definierte Linien mit einer guten Tiefe – und keine Überpixelierung.

Vergleiche

Go bar, Audioquest DragonFly Cobalt und Questyle M15 haben eines gemeinsam: viel eigene Qualitätstechnik, die sie von der Masse der “Dongles” abhebt. Alle diese Geräte sind sehr gut, haben aber unterschiedliche Zwecke und Funktionen. Ein direkter Vergleich ist schwierig, da sie alle (ohne Batterie!) einen Kompromiss darstellen.

AudioQuest DragonFly Cobalt

Dies ist eigentlich ein unfairer Vergleich – für beide. Der Cobalt ist für eine geringe Stromaufnahme ausgelegt, um mit dem iPhone zu arbeiten, und hat daher eine begrenzte Leistung. Ihm fehlt daher ein “balanced/asymmetrischer” Schaltkreis. Der Go Bar funktioniert nicht gut mit dem iPhone, liefert aber viel mehr Leistung auf dem Computer. 

Noch bevor es um die Klangqualität geht, wird der Benutzer die beiden aufgrund ihrer unterschiedlichen Zwecke unterscheiden. Sie sind keine echten Konkurrenten, sondern ergänzen sich. In der begrenzten Überlappung, die beide haben (z. B. 32 ohm-Kopfhörer), ist der Cobalt mit seinem vollen, strukturierten und detaillierten Klang wahrscheinlich unschlagbar. Der Go bar ist ruhiger, aber etwas analytischer, der Cobalt ist “musikalischer”.

Questyle M15

Der M15 ist ein passenderer Konkurrent. Er verfügt ebenfalls über 3,5 mm single-ended- und 4,4 mm balanced Schaltkreise. Er kommt nicht an den Go Bar heran, was seine hervorragende Verarbeitung und Haptik oder die Qualität der mitgelieferten Kabel angeht. Konstruktiv verfügt der M15 über 2 Standard-ESS-SoCs mit zwei Questyle-eigenen Current-Mode-Verstärkungsmodulen.

Der M15 hat seinen klanglichen Schwerpunkt in den Mitten, der Go Bar eher in den unteren Frequenzen. Ich würde dem Go Bar eine geringfügig bessere Artikulation/Akzentuierung zuschreiben, obwohl beide in Bezug auf die Klangqualität sehr nahe beieinander liegen.

Der größte Unterschied zwischen den beiden sind die Funktionen: Der Go Bar hat XBass, Space und wählbare Digitalfilter – aber der M15 hat ein effektiveres Energiemanagement (weniger Batterieverbrauch) und funktioniert besser mit dem iPhone. Ein weiterer Unterschied ist die Bedienung: Die Go Bar umgeht die interne Lautstärkeregelung des Telefons/Computers komplett… sie wird vollständig über die Tasten gesteuert.

Sowohl Go bar als auch M15 sind weniger tragbar als der DragonFly Cobalt. Alberto warf auch den Apogee Groove in die Runde, der nur mit einzelnen dynamischen Treibern funktioniert und grundsätzlich nicht mit Handys. Es ist also nur sehr begrenzt einsetzbar. Aber es bietet eine unschlagbare räumliche Rekonstruktion – und überhaupt keine Features.

Zusammenfassend lässt sich sagen, dass die Go-Bar vielleicht der ausgefeilteste und akzentuierteste Klang von allen ist.

Abschliessende Bemerkungen

Der ifi Audio Go bar ist das Äquivalent eines Rundum-Sorglos-Pakets. Alles ist von sehr hoher Qualität: der Aufbau einschließlich des Tastenmechanismus, der mitgelieferte Adapter und die Kabel, die Funktionalität (einschließlich IEMatch, XBass und XSpace), die Leistung und natürlich der Klang. 

Es ist eines dieser seltenen Dinge, die man blind kaufen kann. So wie ich es mit seinem älteren Bruder, dem iDSD nano BL, getan habe. Oh, in der Zwischenzeit hat der Go bar seinen älteren Bruder in Sachen Auszeichnungen eingeholt.

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Disclaimer

Der ifi Audio Go Barwurde von der Firma für meine Analyse zur Verfügung gestellt – und ich danke ihnen dafür. Weitere Informationen finden Sie auf der Produktseite der Go Bar. 

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The post ifi Audio GO bar Test – Rundum-Sorglos Paket appeared first on Music For The Masses.

As there already is a comprehensive article about M15 on our blog, I will entirely skip a general description of the device features as it would be needless repetition. I will also be succint on most sound impressions.

I’m going to focus on some detail which are not covered in the previous piece, and/or on aspects for which I have a different opinion.

M15 costs € 249 + freight on the manufacturer’s website.

At-a-glance Card

Miscellaneous good stuff and caveats

Good High Gain option

Unlike what happens on so many other devices I heard, M15’s High Gain option is not chastising in terms of dynamic range compression. The effect is indeed very modest, which makes HG a totally viable option whenever one feels like adding some more early juice delivery to one or another driver.

Bad Single Ended output

It is so obviously duller and noisier compared to the Balanced option to be totally unworthy of such an otherwise outstanding product. To give a vague idea, it’s roughly on quality level of M15’s cheaper (120€) sibling, the M12 – which quality, at that price level, is trounced by the like of E1DA 9038D.

This means that one cannot elect the M15 as its “only” dongle if he/she has one or more drivers with single ended connections to support. Too bad.

Spectacular sound, if not totally uncolored

M15’s sound is first of all grand (spatially), and immediately after that it’s clear and detailed. Instrument separation and layering are just spectacular – which paired with its space drawing capabilities make for a really uncommonly good imaging and “sense of immersion” into the outcoming sound.

M15 is not uncolored in terms of tonality. There is some added accent on midbass notes – which is if you wish part of the “usual” compromise “musicality vs purity”. The situation comes out obvious when comparing M15 with E1DA 9038SG3: the two offer equal bilaterally extended sound ranges, with the latter’s bass staying faster, “more technical”, therefore also “less expressive” in a sense.

9038SG3 is however a step under M15 in terms of spatial reconstruction, with particular regards to depth. Layering is also a bit less refined – and that’s mainly why M15 comes across overall “more musically pleasing” compared to 9038SG3.

Powerwise 9038SG3 is better vs very low impedances: in the E5000-acidtest 9038SG3 beats M15 in terms of bass control and overall clarity. It’s fair to observe that 9038SG3 remains the best option around, and by far so, when the available budget is like half M15’s asking price.

In its being “exquisitely musical if unpure” M15 can’t but recall Groove in a sense. In Apogee’s dongle case sound colouring is even more marked, and comes with furtherly higher capability in terms of stage drawing – depth and height most of all. The two devices are not effectively comparable though – mainly due to Groove’s internal architecture making it the odd ball it is – read my piece for the full reasons why.

Annoying RFI sensitivity

When paired to a smartphone connected to the 4G cell network M15 easily picks up RFI (Radio Frequency Interference) when within approx 10cm from the phone. The “solution” is using a non-ultrashort USB-C cable for the connection, but a problem still remains when you walk around with your phone + the M15 in a coat’s pocket…

By comparison, E1DA 9038D behaves very similarly, E1DA 9038SG3 is instead virtually immune to such RFI.

Oddly lacking Direct-DSD support on non-mobile OS

While M15’s USB interfacing is fully supported by Android, not the same happens when the device is plugged onto a Windows or Linux host.

For Windows, Questyle does not make an ASIO driver available (yet?). M15 can therefore be used only on “WASAPI Exclusive” mode. Which means there’s not way to have access to direct DSD transfers (Wasapi only supports PCM).

For Linux the situation is even wierder (if not unique). M15 is apparently not fully supported by standard ALSA (Advanced Linux Sound Architecture) and the end result is that Roon Bridge on a Linux server does not offer Direct DSD support onto M15 when locally connected to it. Not the first time it happens to me – 9038D and 9038SG3 suffer of the same issue.

Considerations & conclusions

It’s very simple: Questyle M15 is “the” DAC/AMP dongle to have if one has 250-300€ to spend, and has balanced connectivity options for all its drivers.

It’s got superb sound clarity and body, very good spatial reconstruction capabilities, very good power management and strong power output.

In terms of overall sound experience M15 easily beats all standalone budget and mid-tier DAPs I happened to audition or own, the ones starting to represent an evident upgrade to it being nothing short of Questyle’s own QP1R, or Lotoo’s Paw 6000 and Gold Touch, or Cayin’s N8. Seen from this angle M15 carries a very reasonable price tag.

The sole serious caveat to mention for me about M15 is its Balanced output option being the sole one delivering good quality. Whoever wants or needs a Single-ended source wouldn’t be as satisfied with M15 (and its price).

M15 is rightfully stuck on our WoE, and determines the passage of Earmen Sparrow and L&P W2 to the relevant Past Excellences section therein.

As I mentioned at the very beginning I bought the M15 unit I am talking about in this article on my own budget after the package with the complimentary review unit which had been sent by Questyle got stolen at my door in my absence. So of course this still “counts” as a manufacturer-supplied unit in a sense, for which I am sincerely thankful to Questyle.

An M15 can be purchased from various Questyle distributors in selected countries, or from Questyle’s own web shop.

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The post Questyle M15 Review (2) – Best In Slot appeared first on Music For The Masses.

OH5 can be bought from Ikko’s website for approx $495 before promos. There’s a nice giveaway promo going on right now, and I have been hinted that a xmas promo is also coming up so stay tuned on their website in the coming days

At-a-glance Card

Full Device Card

Test setup

Sources: Apogee Groove / Sony NW-A55 mrWalkman / Questyle QP1R / E1DA 9038SG3 / Questyle M15 – Radius Deepmount tips – Stock cable – lossless 16-24/44.1-192 FLAC tracks.

Signature analysis

Tonality

OH5’s tonality is warm-balanced, and the timbre is bodied and polished.

Sub-Bass

Sub bass is there but doesnt shine enough. Looking at the graph it does not seem too much rolled off but from actual audition you can check that rumble may use some help to be more evident, and this also impacts negatively on spatial drawing of course. Mid bass elevation tends to cover it, too.

Mid Bass

OH5 have an evidently enhanced mid bass which is key to their global tonality in a positive sense on one hand, but paired to somehow “dampened” transients it also contributes to limitating overall resolving power.

Mids

OH5 mids are very well positioned in terms or relative relation with bass and trebles – not forward, not recessed – to the general purpose of obtaining a globally balanced, horizontally-calibrated presentation (much more so than the graph seems to say). Their tonality is very well “centered”. Highmids ramp up quite rapidly and deliver quite some energy, thus sometimes (although rarely) resulting in some minor inconsistency with the mid and lower ones. Those overly sensitive to 3KHz might be a bit “touched” on some tracks (I’m not in that category, rather the other way around), yet I cound’t hear sibilance which is great of course.

Male Vocals

Vocals on OH5 are good, with particular regards to male vocals. While midbass sometimes gets too close (and does sometimes overlap baritones) they come across very organic, especially on tenor registers.

Female Vocals

Female vocals are also good. A clear preference has been given to energy vs smoothness here. Purists of flutey sopranos may not consider OH5 as top of the block – but apart from that this is another spot where a good job has been made on the OH5.

Highs

OH5’s treble is nice, somewhat airy, and most of all energetic, sparkly while also staying combed, smooth though, so they are in the end not offensive while still staying engaging. The 4.5KHz peak gets hot at times, and depending on eartips selection and/or personal preference/sensibility it may want to be tamed by surgical equing. Other then that, a nice job was done here.

Technicalities

Soundstage

OH5 draw an average sized stage, with a decent width, some height but very limited depth.

Imaging

Macro dynamics (imaging) is good, although primarily in the sense of stereo separation given their flat-ish spatial rendering capabilities. Central panned instruments and/or mono tracks, suffer from OH5’s limitation in terms of layering/separation.

Details

Detail retrieval, like instrument separation, is dramatically sub-average for this price class. It’s quite evident that the entire tuners’ effort has been concentrated on delivering tonal pleasantness and a specific musicality tone, sacrificing resolution and analithical skills.

Instrument separation

Layering and instrument separation are the other major Achille’s heels of OH5, together with detail retrieval as previously noted. There’s little chance to appreciate various voices’ / instruments’ nuances in their singular identities even on acoustic, well mastered, uncompressed tracks.

Driveability

It’s not difficult to drive OH5 as their sensitivity is relevant, and their impedance is on a level where many amps deliver their best current, or near that. A decent phone should be enough, and surely not particularly powerful sources will be.

Physicals

Build

Housings offer a convincing impression of solidity, and sport a very pleasing design style.

Fit

In my case OH5’s shape and size are the right shape and size to fit my concha without difficulty, filling it up almost completely. The nozzles are not very long so long stem tips are in order for me as pushing the housings in beyond a certain point is a no go. In the end I settled for Radius Deepmount.

Comfort

As long as I adopt long stemmed tips, OH5 are very comfy for me once fit. Their weight is also “right” (not too light to “disappear”, not to heavy). They’d become unbearable however if equipped with short stemmed tips, as their housings would hit my antitragus (this is a common issue I have with similar shaped housings e.g. Final A and B series, Tanchjim Oxygen, etc especially on my left ear)

Isolation

As housings do fill my conchas quite well, some level of passive isolation is achieved in my case.

Cable

I found stock cable is quite nice. In addition to good sonic behaviour and very nice haptics, it comes with a modular plug system and 3 termination plugs included in the package (3.5, 2.5 and 4.4). Modular plugs miss a lock-in mechanism but they seem quite firm in position anyway so all OK on that front too.

After further experience with cables etc I must amend as follows. Stock cable is OK from the construction quality standpoint. In terms of sonic pairing it clearly contributes to OH5’s general “more musical than technical” presentation. After rotating lowend and less-lowend cables, I can for example say that pairing a Dunu DUW-02S impacts quite evidently in terms of snappier transients, wider stage, better note contour and layering. 

Specifications (declared)

Considerations and hints

What positively hits you about OH5 is its musicality. Somehow the tuners managed to reach a particular tonal balance, adding a quite personal color to the music being played, and such color is indeed pleasing. The sound coming from OH5 is bodied, vibrant, warm and enveloping. It’s energetic but also rounded off, smooth – there’s no sharp edge, no rigid brick wall, and no floppy surface either.

You got to love OH5’s particular color to appreciate that, of course, which might not happen to you. If you do like its timbre, however, chances are you might develop a particular affection for OH5.

On the flip side, I find OH5 compromise quite dramatically on key technicalities, first of all layering and instrument separation. I am no EQ guru, so I couldn’t (and I wouldn’t) find out inhowmuch the situation is due to aposteriori tuning or to the driver’s specific nature. What I did is play with Roon’s PEQ and after some fiddling I could devine some touch-ups wich make the situation a bit better (for my tastes of course)

Low shelf    55Hz  4dB   0.71
Peak        150Hz -3dB   0.5
Peak        950Hz  1.3dB 1
Peak       4500Hz -2dB   2

The 150Hz demotion helps making midbass much more polite and somewhat faster. The low shelf adds some missing “rumble tail” to bass notes. The 950Hz (or thereabout) pushup also helps de-dampening midbass and lowmids and the 4.5K dip takes some hotness away from metal notes.

All those figures are not carved in stone, take them as ballpark values, but if you try you will hear instrument separation and clarity improving, a more detailed bass, and a quite evident opening on stage drawing especially in the depth direction. Play with values to learn how sensible each one is to the final result.

An alternative possible intervention is adopting TRI Clarion eartips. In such case the pushdown on 150Hz or thereabout is not needed anymore, however a more generous dampening intervention gets required on the highmids – I would add a -2dB or so to 3Khz for example, in that case.

Lastly: some care is in order about ideal source pairings. OH5 do not welcome bass-strong sources too much. Questyle M15 or QP1R, and E1DA 9038xx dongles all OK. Groove on the other hand excites OH5’s “artistic” midbass too much, yielding a too dark result, thus not even being able to help OH5 on adding space depth which is amongst Groove’s specialties in general.

Comparisons

Final E4000 ($149)

The epitome of IEMs featuring strong musical personality (color) which grew on me since the day I got them and won’t ever leave me, even now that I have technically better alternatives, are Final E4000. And guess what: E4000 and OH5’s personal “voicings” offer quote a few common points.

Both are warm, smooth, musically “pop” and deliver a very particular balance between smoothness and strenght, energy and pampering. Compared to OH5, E4000 are… more japanese: silkier, a bit (even) more elegant in a sense. OH5 feel more energetic – in a good sense.

OH5 are braver on the trebles compared to E4000, even at the cost of minorly overdoing sometimes. E4000 on the other hand are very good at layering and separation, where OH5 is dramatically lacking, especially on the mid and low segments.

E4000 are much more demanding in terms of source power, and they have the not secondary advantage of costing one third of what OH5 do.

Oriolus Isabellae ($500 street price)

Isabellae’s musical personality is evidently different from OH5’s insofar as they deliver a V shape presentation, with relatively recessed mids and important, enhanced sub bass and bass (for the connoisseurs: something more in the ballpark of Ikko’s other model, the OH10). This alone of course already imprints a big part of the comparison between the two products.

Beyond that, Isabellae’s high mids are smoother, and trebles are less energetic, yet airier compared to OH5. Mid bass is definitely more textured and detailed on Isabellae, while still staying on the relaxed and buttery side in general. Sub bass is OK out of the box on Isabellae while it requires some help on OH5. Most of all, layering, resolution and detail retrieval is obvisouly better on Isabellae, which are also equivalently undemanding in terms of source power as OH5 are.

Dunu ZEN ($699)

Zen’s bass is arguably as good as the industry gets at least until pulling Softears stuff to the comparison table: perfectly calibrated mix of punchyness and body, volume and texture. ZEN also has beyond outstandind microdynamics and layering capabilities – on both fronts, that’s very much unlike what happens on OH5. On the opposite end, trebles are marginally but perceivably airier on OH5, although highmid sensitives as previously noted my consider that segment on the limit or even a bit beyond their preference. Taken as a whole, OH5 have a stronger musical personality (love or hate, of course), ZEN are more “technical”, and 40%+ more expensive. I’m referring to original ZEN model, not the subsequent Pro version (which I find less nice then its sibling).

Conclusions

OH5 are a nicely and coherently colored set. As such, they can be target of “unquenchable hatred and indomitable love” – like that other well known invididual from some 200 years ago, you know – depending on one’s own ego. Simply put, if you are the uncurably curious audiophile addict you may want to check these, knowing chances are you might viscerally love them, or find them as being “not your cup of tea”. I hope my article did convey at least some hints to educate your guess before you actually carry that out.

In summary OH5 deliver a bodied, vibrant, warm and enveloping musical experience. They are energetic while also smoothed. An evident accurate job has been carried out behind the curtains here to get to such point.

Their major downside is on resolving power and layering proweness, which can be helped a bit albeit not solved – without revolutionising the entire presentation – by means of some surgical EQ as I also tried to hint you about, here above.

The OH5 sample covered in this article has been delivered to me courtesy of Ikko staff which I warmly thank once again for the opportunity.

Our generic standard disclaimer.

The post Ikko OH5 Asgard Review – Music Better Than The Rest appeared first on Music For The Masses.

Gear of the Year: 2022 marks the blog’s fourth year. We collectively published 100-150 articles, mainly product reviews, but also technical information (such as earphone modding). Apart from receiving review units from manufacturers and sellers, we also purchased a lot…and we borrowed from audiophile friends and colleagues.

Our list of earphone reviews is going towards 400, which is a very useful database. And one of the world’s biggest. In the DAC and amplification department we have also reached a respectable 70.

While we shrank from 8 to 6 authors, we essentially doubled our viewer numbers and currently record well over 1000 daily individual blog visitors (1377 on Black Friday 2022). But we remain humble, continue viewing our hobby as labour of love and focus on information for you, the reader, while not selling out.

Another very popular list is our Wall of Excellence, which hosts gear not approved by one of us, but by the whole team. This is a useful filter for you and should give you confidence in your buying decisions.

And yes, we searched for and found an exclusive sponsor in HiFiGo, who help us with our basic operating cost (web hosting). We still chip a lot of our own money in for mailing between us reviewers, import charges etc.

By having a single sponsor (and not Google ads) paying us a moderate flat fee, we don’t rely on viewer numbers. For you, this means no popups and no ads between paragraphs, nothing in your way when reading our articles. All advertisement takes place in the top toolbar and the sidebar. Feel free to check it out.

We also continue refraining from affiliate links as it still leaves a bad taste in our mouths.

As at the end of the previous years, we list our our personal favourites of 2022 – the portable audio we personally enjoyed most. There are no rules, we just tell you what we like. It does not have to be the latest. After all, the gear we use most is the best for us. Between us, the Dunu Zen, 7Hz Timeless, Final ZE3000, and Questyle M15 received the most mentionings. The Questyle M15 appears to be the most highly acclaimed dongle in the blogosphere period and may as well be the “Product of the Year” all around.

We don’t publish any “best of” lists as we have not tested all competitors in each category.

Enjoy this read and we wish you a happy and successful 2023!

We thank

And here we go…that’s what we enjoyed in 2022…

Alberto Pittaluga… Bologna, ITALY

My 18 readers know that I’m much more into enjoying better sound than into getting excited about sidegrades or other small changes from my existing preferences. I am the polar opposite of a marketing hype target individual – on pretty much any topic by the way. I mean: try and sell me a “better” (?) Moka coffee machine…

With that in mind, and considering that I’m not into this since yesterday evening, it’s quite logical that the list of the really significant items out of all those I come across over a year’s time is short. Here’s what I found in 2022 which is worth recommending.

IEMS

Intime Miyabi – a great piece of engineering ticking most if not all boxes for my tastes. Beyond details, one of the absolute best sound delivery experiences one can buy (well… some personal initiative is required to fetch it from Japan) below 500$… spending less than half of it. My report here.

RHA CL2 – I heard quite a few planar IEMs, and not a single one “closes the door” for me. Long story short, none offers me a well-articulated-enough sound experience out of the box, and none carries a driver good enough to be equalised into something I really like. The sole exception I ever came across yet is this RHA CL2 – a dated model indeed, which I had the venture to audition this year for my first time. More on this on my piece about it.

DAC/AMP Dongles

Questyle M15 – no doubt the “best overall quality” battery-less DAC/AMP (“dongle”) at any price. Paired with a small transport easily makes all sub-800$, and most sub-2K$ DAPs pointless in terms of sound quality delivery (some may still prefer “some” DAP for better pocketability convenience or specific features). Firstly reviewed in depth by Jurgen here, you can read my incremental notes here.

E1DA 9038SG3 and 9038D – the M15 is what it is, but it also is relatively expensive (almost 300€ once delivered to Italy). My previous experience taught that me that exclusively higher-tier (and price) dongles are able to deliver sound qualities worth investing in more money than the 9$ price tag asked for an Apple Dongle ($9). Then I found these two gems which fulfill the need for incredibly clean and powerful sound for less than half M15’s price, and 90% of its proficiency.

Biodegraded…Vancouver, CANADA

7Hz Timeless:
Despite a somewhat elevated and loose midbass, these have great overall tonal balance, smooth mids, excellent instrument separation and layering, and speed. Macrodynamics are on the soft side, which might not be for everybody; and the fit will be problematic for people with concha bowls on the smaller side. They work best with warmer, dynamic sources.

Durwood…Chicago, USA

Swimming in dongles I don’t have a favorite, Moodrop Dawn has the selectable gain, others like the Shanling UP3 have single and balanced outputs and controls. Take your pick. I would like selectable gain AND single ended/balanced outputs.

Desktop DAC goes to the SMSL C200 for best value and sound that hangs near the top dogs. I personally prefer separate pieces for the ability to have more connection options and swapping but for near double the price I don’t see double the performance. The SMSL SU-9 is excellent for me I ended up buying one on the second hand market. The SMSL DO100/HO100 is great too and bit more user friendly for the size.

For earphones I still love my Shozy Form 1.4 for the bass tuning, warm signature and superb fitment YMMV. There are lots of excellent performers if just factoring in sound and technical merits including the 7Hz Timeless and Moondrop Kato, but I struggle keeping them in place. For those starting out or not a lot to spend, my budget pick for what is available would be the CCA Lyra.

For TWS, I only have 3 pairs to compare-but if I don’t need ANC, I would pick the Moondrop Alice all day everyday. Review coming soon, they are a wireless Kato and they nailed all the important features (sound, battery life, control, fitment), form follows function.

Jürgen Kraus…Calgary, CANADA

In terms of earphones, I continue liking the JVC HA-FDX1, the Dunu Zen, and the Final E5000. The Zen remain my go-tos. New on my list are the LETSHUOER EJ7M (great allrounders), the Final A3000 & E3000, and the very articulate Dunu Talos. But the earphone I used most in 2022 is the superb Final ZE3000 TWS.

As to headphones, my Sennheiser HD 600 have received company by the Final Sonorous III. On the budget side, I added the excellent KTXPro1 to my Koss selection (Porta Pros, KPHi-30, and KSC75).

I am principally a “portable guy” who used to operate his iPhone with a dongle, mainly the AudioQuest DragonFly Cobalt. Whilst the Cobalt remains a standard staple of mine, the excellent Questyle M15 (and here too) is a great alternative for my higher current-drawing headphones/earphones – and my personal product of the year 2022.

For driving full-sized headphones and single DD items, nothing beats the Apogee Groove. For Bluetooth DAC/amp, I exclusively use the very mature Qudelix-5K. And for earphone testing, I am still holding on to the totally underappreciated Earstudio HUD100.

The Hidizs AP80 Pro-X has proven a great dap for me, no matter if used alone or as transport with a dongle. The Sony NW-A55 with Mr Walkman firmware also remains in my collection of favourites. And the very crisp sounding Questyle QP1R remains my flagship dap. The biggest surprise for me was the Tempotec V6 dap…essentially a $500 device at half price.

For my desktop setup, I am holding on to EarMen Tradutto DAC in combination with the Burson Funk amp and AudioQuest analog and digital interconnects.

Kazi Mahbub Mutakabbir…Munich, GERMANY

Just like that, another year went by. This year was special for me as I got to visit High End Munich 2021 (which happened after a 3-year hiatus) and also got to meet some great reviewers and legends of the industry in person.

Best Headphones: This year, I have slowly transitioned from portable audio into desktop audio. A knock-on effect of that has been my growing headphone collection. What used to be a mere 4 headphones last year, has grown into a whooping 25 now.

Yup, I need another shelf.

Speaking of the best headphones, Hifiman HE-6se V2 got the most use this year from me. I still use the Sennheiser HD650 whenever I feel like relaxing, but for my playlist – the Hifimans absolutely slap!

That being said, they aren’t the best headphones that I’ve tried this year. That recognition goes to the venerable Stax SR-X9000. Still underrated in the summit-fi segment, these overtook Susvara for me in almost every aspect. They are build better, has better sense of space, bass is better defined and slams a bit harder, and of course – the ethereal treble is an absolute joy.

Too bad that the price tag (with energizer) makes these nigh-unobtainable for me. Doesn’t make them any less awesome, though.

Best IEMs: Dunu Zen remains my daily driver, and they scale tremendously with desk sources, so I am a happy camper.

Speaking of camps, Campfire Audio Holocene has sneaked their way into my daily rotation. Probably my most favorite Campfire Audio IEMs, ever. They deserve a lengthy, long-term review, which is in the pipeline.

This year I also had to wade through loads of “meh” IEMs, each mimicking a certain “scientifically perfect” (lol) curve in one way or another, and each sounding boring and unremarkable. But the one IEMs that kept the fire alive was the Softears Turii.

Exceptional in almost every sense, their fit is the biggest concern, but with Spinfit W1 tips (excellent tips btw, recommended) I found them stable enough to enjoy outdoors. Too bad that they’re discontinued.

The Softears Twilight is a spiritual successor of sorts. Doesn’t have the crazy sense of space that the Turii portray, but pretty much beats every other single dynamic I’ve tried under USD $1000 (and I have tried nearly all of the hyped ones by now). So yeah, Softears, take a bow!

Desktop DAC and Amp: Questyle CMA Fifteen takes the cake for the best all-in-one system that I’ve heard. Near-endgame for 99% out there I’d say. Drives almost everything with authority. What’s not to like? Oh yes, the price. Something’s gotta give after all.

As for standalone units, the best solid-state amp I’ve tried: Zaehl HM1. As for the best tube amp: Feliks Envy, or the Feliks Euforia AE. Two very different price-points, but both about the best tube amps you can buy right now.

DACs are a bit difficult for me to judge, and the ones I’ve tried this year couldn’t replace the Holo May L3 I tried last year, so that one still reigns supreme.

Portable DAC/Amp: Questyle came outta nowhere with the M15 (and here too) and seized the day. Nothing else comes close, really. On the higher side of the price, Chord Mojo 2 is an excellent device. For using with IEMs and moderately efficient headphones – that’s all you’ll ever need. The DAC section is kilobuck-level as well.

Surprise of the Year: Final ZE3000.

I do not like wireless stuff at all, mostly because of how poor and compressed 99% of them sound. This one caught be off-guard. I bought them only to review them, fully expecting to put them up for sale once the review is done.

Now, three months later, I carry them everywhere and despite the finicky touch controls, I keep coming back to them. Only the Sony WH-1000XM4 sound as good as them, and they cost 2x the price. To think that Final would come up with one of the best TWS IEMs around – didn’t see that one coming.

Then again, who else would be that methodical?

Loomis Johnson…Chicago, USA

IEMs:

Moondrop Starfield—my default rec to folks who ask me which <$100 phone to buy…big-sounding, with seamless coherence and as lot of PRAT.

7Hz Timeless—polarizing and imperfect (there’s some stridency at the highest frequencies), but technically very accomplished and highly resolving. The more I hear ‘em the more I dig ‘em.

Tin HiFi TWS Buds 3—light on features (no app, no ANC) but beautifully built and a really engaging listen, with very good imaging and instrument placement.

AXS Audio Professional Wireless Earbuds—generic-looking, with a modest soundstage, but probably the best-sounding TWS I’ve heard to date. 

Portable DAC/Dongles:

Xumee USB-C—lots of power and surprising finesse for less than the price of a six pack.

Hidisz S3Pro—a really refined performer which synergizes well with anything under 150 oHm.

Headphones:

1More Sonoflow Wireless ANC—you can pay a lot more for punchier sound or more tech features, but these are an awful lot of headphone for the money, with good noise-cancelling, an unforced natural tonality and fantastic battery life.

Koss KTXPro1—uber-comfortable and musical as hell; for $19 there’s no reason everyone shouldn’t own a pair.

Bluetooth Speakers:

Oontz Angle 3—diminutive, seemingly indestructible cheapo with surprisingly good bass output and an enveloping 3D soundstage.

Edifier R1280DB Bookshelf Speakers—I gave a set of these to the owner of a cannabis dispensary, whose customers rave incessantly about how good they sound even before imbibing.

Desktop DAC:

SMSL SU-6—a more than capable DAC, with excellent bass control and a very detailed (if occasionally overbright) high end; digital preamp and Bluetooth functionality are big plusses. Outstanding value.

KopiOkaya…SINGAPORE

The editor: although KopiOkaya retired a year ago (on his own terms), he keeps his eartips compendium up to date. His list has been a cornerstone of this block and its most successful article. It currently records somewhere between 80,000 and 100,000 views.

Best under $50 IEM:

7HZ SALNOTES ZERO

Best under $100 IEM:

DUNU KIMA

Best IEM of 2022:

SOFTEARS TURII Ti

Best dongle DAC/AMP of 2022:

QUESTYLE M15 (here and here)

Best portable DAP of 2022:

TEMPOTEC V6

Best budget DAC/AMP combo:

SMSL SH-9 THX AAA-888

Best desktop DAC of the 2022:

HOLO MAY KTE (KITSUNE EDITION)

Best desktop headphone amp of 2022:

QUESTYLE CMA FIFTEEN

Best eartips of the 2022

SPINFIT W1

Most outstanding audio product of 2022:

QUESTYLE M15 (here and here)

And This Was The Previous Year:

The post Gear Of The Year 2022 – Our Personal Favourites appeared first on Music For The Masses.

A device gets attached to this Wall of Excellence when based on our private and of course subjective experience it performs so well within its technical and price category as to even discourage considering homologous alternatives.

If it ain’t here, WE don’t want it!

Please note that our WoE will not be limited to devices we actually published a review of. Nonetheless, all WoE devices have for long time been or still are part of our operative gear.

We start small and plan to expand our wall according to merit.

This Site is being consistently updated…please bookmark it and keep checking back!

LATEST ADDITIONS: Qudelix-5K, Questyle M15, E1DA 9038SG3.

In-Ear Monitors

Past excellences, now discontinued or superseded

Headphones

Digital Audio Players (“DAPs”)

Desktop Amplifiers

Desktop DACs

Desktop Integrated Headphone DAC/AMPs

Portable Headphone Amplifiers

Portable Integrated Headphone DAC/AMPs

Headphone DAC/AMP “Dongles”

Dongles are little DAC/amps without battery that are powered by their source device.

Past excellences, now discontinued or superseded

Accessories

The post Wall Of Excellence DRAFT appeared first on Music For The Masses.

Cons — Lacks extension at both ends; not the best note definition, could be a bit edgier.

Executive Summary

The $600 Oriolus Isabellae is a warm, cohesive sounding, well balanced single DD iem with an inviting organic timbre that provides for a relaxed listen.

Introduction

Oriolus is a Japanese boutique manufacturer that has subscribed to high-end audio since 2015. They specialize in in ear monitors but also offer a digital analog player. Oriolus products are only available from hand-picked retailers.

The Isabellae is a well-perceived single-dynamic-driver earphone in the $500 price category. I could test it only for a few days as it was a private loaner.

Specifications

Physical Things and Usability

I did not receive the retail packaging but the bare iem plus cable. But what immediately stroke me was the earpieces haptic: I could not think of a better quality resin. The earpieces felt substantial and superb between my fingers. What’s unusual is the large nozzle thickness. I had to stretch standard eartips with 4.5 mm stems to get them on. Isolation was not great, but fit and comfort were good.

Cable comes with a 4.4 mm pentacon balanced connector. I used a Venture Electronics adapter to connect to 3.5 mm sockets. I could not find any tech information on the cable but it was very flexible and great to handle.

Tonality and Technicalities

Equipment used: Sony NW-A55; Macbook Air + AudioQuest DragonFly Cobalt or Apogee Groove; LETSHUOER EJ07M white stock tips.

The Isabellae has an overall warm tonality with a slight V-shape, resulting in a cohesive, organic presentation without irritating peaks. Extension at both ends ist somewhat modest so that the main focus is at the mid-bass to lower midrange. The frequency response is close to the plethora of personal target curves of the usual protagonists. Tuning is safe and there are no unpleasant surprises across the frequency spectrum.

This results in a relaxed, never fatiguing listening experience that is particularly well suited for orchestral/acoustic pieces.

There is a mid-bass boost that is not overbearing (it does not pound against my sensitive eardrums) which adds warmth to the image and adds richness to vocals in the lower midrange. This frequency region is the most obvious/characteristic of the whole presentation.

Mid-bass is of medium speed, well textured, but can come across as a bit fuzzy at times, and it can make vintage recordings sound thick. Although there is some rumble at the bottom, sub-bass extension could be better, which is reflected in decreased soundstage depth.

The lower midrange gets weight from the mid-bass boost and not much energy from the moderate 2-4 kHz area. This results in rich, slightly warm vocals reproduction and the complete absence of shoutiness, but at the expense of midrange clarity. Note definition could also be improved, that is adding some edge. Male vocals have a good weight, more so than female ones, which are more intimate.

The drop starting at 2.5 kHz and increasing just before 5 kHz excludes sibilance and adds to the overall smoothness, but it also adversely affects stage width. Cymbals and hi-hats are typically recessed, though reasonably well resolving.

Soundstage is of average width with average depth. I miss a deeper stage. Macrodynamics is a bit polite for my taste, notes are rather rounded and deserve more bite. This contributes to the opposite of fatigue…after two hours or so, I feel I need a bit more pizazz and a harder kick! Microdynamics could be better at this price tag. I’d appreciate a bit more sparkle.

Spatial cues, layering and separation are good however the usual compromise of a single DD, but the wonderful organic timbre compensates for that. The stage can be crowded when a symphony orchestra plays.

Oriolus Isabellae Compared

Of the iems I know, the Dunu Zen comes sonically closest to the Isabellae. The Zen has a better sub-bass extension, which creates a deeper stage and a more immersive and engaging listening experience. And it has incredible microdynamics. But it also has a glare in the upper midrange that can attenuate and sharpen vocals. Both lack notable treble extension.

The better resolving $600 LETSHUOER EJ07M tribrid at a similar price tag provides a stark sonic contrast with its much drier, leaner, more energetic, less weighty, and more brittle and neutral presentation. It’s bass is also boosted, but faster, tighter, and crisper/snappier/more dynamic, the vocals sharper, leaner, and more energetic. But the EJ07M has the narrower (however deeper) soundstage and the better resolution. You want the smoother Isabellae with its thicker bass and natural timbre for laid back, symphonic listening and the Shuoer for your energetic rock music.

The $350 Unique Melody 3DT with its three (!) dynamic drivers has a sharper, edgier/spicier, and a leaner, cleaner but more brittle and analytical presentation. Notes are better defined but the experience is less engaging though crisper than with the Oriolus. I had to tape the 3DT’s nozzle off to reduce its upper midrange and the danger of shoutiness. Both models could not be more opposite: warm and fuzzy vs. correct and sterile.

The $200 Moondrop KATO has the narrower stage but a clearer midrange and an overall “harder” sound than the Isabellae. It is not as warm and rich as the Isabellas, which probably stems from its comparatively less boosted mid bass. The Oriolus sounds smoother and more mature, the Kato more juvenile and energetic…and borderline shouty.

The JVC HA-FDX1 has a leaner, more neutral, and less bassy presentation. It offers a narrower stage, is more strident in the upper mids, has a somewhat metallic timbre…and is still less analytical than the UM 3DT.  

Concluding Remarks

The Oriolus Isabellae are rather polite playing single-dynamics earphones with state-of-the-art haptic. They will appeal to fans of Final Audio Design products (another Japanese company) and to those who enjoy natural timbre in acoustic sets.

Until next time…keep on listening!

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All Our 302 Earphone and Earbud Reviews (with Links)

Disclaimer

This private loaner was kindly provided by Super Best Audio Friend Rockwell – and I think him for that.

Get the Oriolua Isabellae from Musicteck.

Our generic standard disclaimer.

You find an INDEX of our most relevant technical articles HERE.

The post Oriolus Isabellae Review – Japanese Politeness appeared first on Music For The Masses.

Current-hungry DACs drain our DAP/Android Devices fast – and Apple’s current-draw limit frequently just results in an error message. The solution is an external power bank – which is easy in the DAP/Android case, but still problematic with iOS devices. Here’ s how to do it…with the E1DA splitter (2 kinds) and the Apple Camera Adapter.

This article is based on discussions with Alberto and I thank him for his insights…which have cost me lots of sweat in the meantime. Gordon Rankin or Wavelength Audio pointed the Apple restrictions out to me. I purchased all these cables myself. My testing refers to extreme cases. The E1DA splitters work as advertised as they are optimized for their own products.

Introduction

Dongles are little DAC/amps that draw their operating current from the host, which is either a computer, phone, DAP, or tablet. They have the advantage that they do not subscribe to planned obsolescence as they do not rely on an internal battery that dictates their life span. And they are small because of it. That’s why I like them.

There are principally two kinds of dongles, such that limit current draw to preserve the host’s battery, and such that…don’t (I wrote about this here). The first are limited in their performance, and the second drain the host (too) fast. Remember the 36 hours we got out of our iPod Classics?

Apple adds the “fun fact” of limiting current draw to 100 mA for most of their iOS devices. Reason is their fear of dissatisfied customers falsely claiming battery failure on warranty when their phones’s batteries drain “too fast”. If we connect our dongle with a current draw exceeding 100 mA, we get no music but a cryptic error message instead. Not good.

There are exceptions, when the iOS device is fooled to believe the draw is below 100 mA…by means of fudging the descriptor table in the software (according to Gordon Rankin of Wavelength Audio). Hidizs S9 Pro and ifi Go Bar (both around 140 mA), for example, appear to get around the restrictions, as both work with my iPhone SE (1st gen.), Astell & Kern’s PEE51 does not.

Luckily, some third-party Lightning cables exist that trick the iPhone to believe the current draw is lower than 100 mA…which is only a short-term solution, as it does not stop the battery from draining fast. E1DA have published a useful spreadsheet that tells you which Lightning cables work with which iPhones with their 9038SG3 and 9038D DACs (which draw above 130 mA).

Android devices and most DAPs are more forgiving – and work even with the biggest current w**res (excuse my Italian). But not for long in each case before you have to hook your device back up to the charger.

Therefore, in order not to run out of juice prematurely – and to entice Apple devices to play through our monster dongles – we need to use an external power source to drive those – and the readiness of the host to accept them.

The theory is simple: just separate power line and data line in the cable connecting host and dongle. As a result, the dongle draws its power from an external 5V power bank (or out of our 5V wall charger), and its data (“music”) from the phone/DAP/tablet. Computers are exempt in our discussion as they always provide enough power to any dongle (USB 2 up to 500 mA).

Stop! The theory may be easy for Android devices/DAPS, but things can be way more complicated for iOS devices, when the 100 mA current-draw limit needs to be circumvented.

What we need for DAPs/Android devices is a splitter cable (with separate data line and power line), an external battery, and the music host. This also works for iOS devices with <100 mA draw, but the Apple Camera Adapter is needed for anything higher. And that’s not all: you also need an MFI-certified USB-A (or USB-C) Lightning charging cable to make it work. Let’s test all possible cases.

DAPs/Android Devices with external Battery and E1DA Splitter

The easy case first to warm you up for things to come. What we need:

• Music source
• $4.99 E1DA USB-C to USB-C splitter cable
• Power bank
• Dongle (here the current hungry, 280 mA consuming, very powerful Apogee Groove…with micro USB to USB-C adapter
• Headphone

We plug it all in – and it works. The source device is not charged during music play and the Groove receives the required current from the power bank. Easy peasy!

iOS Devices with external Battery and E1DA Splitter vs. Apple Camera Adapter

1. Dongles with a Current Draw <100 mA (or a Software Manipulation pretending it is <100 mA) –> E1DA Splitter and Apple Camera Adapter work

Well, that’s smooth when the connected dongle draws less than 100 mA as it essentially works like the DAPs/Android device above. It just need a different E1DA USB-C to Lightning splitter cable that sets you back $19.99. Just in the case above, your iOS device is not being charged during operation.

2. Dongles with a Current Draw >100 mA (and without Software Manipulation pretending it is <100 mA) –> only Apple Camera Adapter works, E1DA Splitter does not

In this case, the E1DA USB-C to Lightning cable does not work. All you get is an error message. The power bank fuels the Groove properly, but there is no chip telling the iPhone it does actually not have to supply power to the Groove. Hence the iPhone reacts as it would without power bank: it does not want to do it. Thank you, Apple.

We still can drive the Groove but need different equipment to do so. Please fasten your seat belt! We need the following ingredients:

• iOS device
• $49 Apple Lightning to USB 3 Camera Adapter
• Power bank
• MFI-certifed Lightning charge cable
• Dongle (here again the current hungry, 280 mA consuming, very powerful Apogee Groove)
• Micro USB to USB-A cable
• Headphone

This works beautifully. The power bank funnels current into the Apple Camera Adapter with its controller chip that informs the iPhone that it does not have to supply current to the Groove. And the Groove draws its power from the power bank, which is controlled by the iPhone. On top of that, the iPhone charges while playing.

But there is still another hurdle: the charging cable has to be MFI certified to be able to communicate with the Apple Camera Adapter’s controller chip. Lightning cables by Amazon (Basics), Apple, IKEA, Startech and UGreen worked in my tests, OTG cables by ifi Audio, ddHiFi, and OE Audio did not (they are unidirectional “the wrong way” and not designed for charging). No power through the latter – and the well known error message appears on my iPhone. Bummer!

Concluding Remarks

Driving dongles with DAPS/Android devices and power bank using the E1DA USB-C to USB-C splitter cable works universally.

In contrast, iPhones (tested with SE 1st gen. and 13) do not like to draw zero current when they are not told to do so by a controller chip. As it appears, the Apple Camera Adapter draws some current from the iPhone but also from the power bank. E1DA’s USB-C to Lightning splitter works different from Apple’s Camera Adapter as it talks to the dongle and not to the iPhones’s power management. Therefore, iPhone does not charge while being connected to the E1DA splitter.

And I wished E1DA added a chip (as in Apple’s Camera Adapter) to make their Lightning splitter universally usable (Apple apparently makes this impossible for audio designers without an MFI license – and E1DA are probably not an accredited MFI contract manufacturer). E1DA’s USB-C to Lightning splitter is therefore of limited use for third-party dongles – but it works as intended with E1DA’s own DACs. This cable is not MFI certified.

As a rule of thumb, E1DA’s USB-C to Lightning splitter will only work with dongles that iPhone believes are drawing less than 100 mA current. And Apple’s Camera Adapter only works with MFI-certified charge cables. Nothing is perfect.

Until next time…keep on listening!

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The post Driving Power-Hungry Dongles With DAPs/Android Devices and iPhone (E1DA Splitter and Apple Camera Adapter Review) appeared first on Music For The Masses.

As I will try to outline, from multiple standpoints GO Bar fully partakes into iFi’s overall general product philosophy both in terms of components and general application. It promises outstanding results, and surely comes with a price tag (€ 329 in EU) positioning it at the top market level. It can as always be purchased directly from iFi, on their Amazon space, or from one of their distributors around the world.

At-a-glance Card

Features and description

Externals

GO Bar’s housing shape recalls that of their Nano / Micro iDSD line – just many times smaller. Within the “dongle” category, GO Bar falls more or less midway in terms of dimensions and weight: 65x22x13mm and most of all 28.5g are not certainly huge yet not entirely disregardable either when I have GO Bar connected to my transport. It’s more than twice the size and weight than an E1DA 9038S, yet much smaller and lighter compared to a Groove.

On one small ends we find two phone outs, on the opposite end a USB-C connector. On one of the long sides there are two + and – buttons, another multifunction button, and a microswitch dedicated to IEMatch.

Finally, on the bottom face there’s a raw of 9 little holes corresponding to as many LEDs, with engravings already hinting about those being dedicated to signaling which PCM sample rate is being played, or if DSD is being received, whether MQA is being detected and finally wheter XSpace and/or XBass+ are activated.

GO Bar is supplied in a small size carton box with adequate internal protections. The bundle is quite generous insofar as it includes a USBC-USBC short cable, a Lightning-USBC cable, a USBC-USBA adapter and a black leather pouch with enough room for the GO Bar itself plus at least one of such cables.

Internals

iFi traditionally offers quite some macroscopical information about the contents of their devices, but scarce or no fine details about them.

USB communication is taken care by a XMOS 16bit chip, which I would call an obviously good – for quality – and at the same time brave choice – when I think to the power that element alone requires to operate.

Unlike many other iFi DAC devices, “traditionally” equipped with Burr-Brown DAC chips, GO Bar is built around a 32bit Cirrus Logic DAC chip. The rest of the components are coming from the usual manufacturers normally enlisted by iFi: Murata and TDK capacitors, and Texas Instruments power supplies. I won’t go too much down through this as, in lack of better specifications coming from the manufacturers, I find this information to be useful essentially at the marketing level only.

Input

GO Bar offers a single input, being the USB-C port which is therefore supposed to carry in both digital data and power.

The range of accepted digital resolutions is nothing short of extended:

• PCM up to 32bit / 384MHz (ASIO drivers required on Windows to exceed 24bit / 192KHz)
• DSD direct up to 256 (DSD-direct requires ASIO drivers on Windows)

GO Bar is also an MQA full decoder. That means that it can both unfold non-authenticated MQA tracks, and (automatically) pre-authenticate with the MQA provider to guaranteer even higher quality and especially fidelity vs the digital content distributed by the publisher.

On the housing’s bottom face there’s a generous stripe of LEDs flanked by engraved markings.

One of the topmost 6 LEDs of the stripe will light up in white to indicate which digital resolution is being received by GO Bar: 44/48, 88/96, 172/192, 352/384, DSD64/128 or DSD256.

The 7th LED will light up of different colours depending on the situation:

• when an MQA stream is being received : green for MQA, blue for MQA Studio and magenta for “MQB” (Original Sample Rate).
• when a PCM stream is being received: cyan for BP filter, white for GTO, red for STD, yellow for MIN and off in case of DSD stream. (Read more down below about filters).

Output

GO Bar offers 2 distinct phone out ports: a 4.4mm Balanced and a 3.5mm Single Ended one.

iFi’s web site is quite stingy on information about GO Bar’s output power so I asked them directly. Here’s two tables giving a much more complete picture of the situation:

Balanced output

Single ended output

So it’s quite clear that GO Bar offers very nice output power on high impedance loads, while it is severely limited in terms of current delivery which turns into not particularly enticing figures from 16Ω down.

Don’t take me wrong here: GO Bar is still more powerful on low impedance loads, and delivers more current on medium impedance loads than many other “dongles” out there. However, GO Bar’s marketing tag line (“World’s most powerful headphone amp of its size”) is, simply, not true when taken in the absolute terms used by its very wordage.

Looking at the figures, it comes out that the device has a hard max current cap around 130mA which it reaches on a 32Ω (or thereabout) load, and to stay at safe distance from that it’s tuned such that drawn current is progressively reduced as load impedance goes down.

Running some simple math it’s also quite clear that GO Bar issues its max power on Balanced Ended output vs circa 55Ω impedance, that being a bit less than 1W, and from there on down it rapidly starts winding down both voltage and current flow.

Neither of the two output connections is configurable as a pure Line Out.

Output impedance on both BE and SE ports are declared to be equal, both < 1Ω. A nicely low value, although not a superlow one.

I also find it interesting to note that SNR and THD+N promised values on BE and SE ports are dramatically different. On BE port iFi declares 132dB(A) SNR and <0.002% THD+N. The former value in particular is really outstanding.

On the SE port they instead declare SNR at 108dB(A) and <0.09% THD+N which I would both call “unimpressive” to be generous. And match unenticing subjective audition experience (more below).

Lastly, the 3.5mm phone out on GO Bar offers iFi’s proprietary “S-Balanced” connectivity – which is a sort of superset of the usual 3.5mm Single Ended standards, offering some more cleanness, and full backwards compatibility to all existing 3.5mm terminated drivers.

If we except the case of fixed-cables earphones carrying TRRS 3.5mm plugs from factory (I don’t personally know of one), exploiting GO Bar’s 3.5 TRRS option would require swapping cables, and as such pretty much anybody at that point would swap onto a 2.5mm or 4.4mm terminated one, and exploit the full-blown Balanced port (the 4.4mm one) on the GO Bar.

So in the case of GO Bar – much the same as in the case of Micro iDSD Signature or GO Blu – I guess that the practical value of the S-Balanced technology applied behind the 3.5mm port is limited to the xtalk improvement – which is nevertheless nothing to bin.

Host power requirements

I didn’t analythically measure GO Bar’s power requirements, yet there are quite a few things that can be said based on iFi’s published data, and subjective experience.

First: iFi declares a (maximum) host power draw of 4W. That’s huge. It corresponds to 800mA which is far beyoned USB 2.0 limits.

That’s a maximum absorbtion figure – GO Bar will work on usual smartphones and tablets, and even on iPhones, but if the host (the smartphone, the tablet, or the PC) is not capable of delivering up to 800mA current / 4W power then GO Bar’s effective powering capabilities will be limited by the host’s cap. Or oppositely stated: to get the full power “promised” by its specs, GO Bar must be hooked onto a USB3-capable host, or a smartphone compliant with USB PowerDelivery standard.

Second: GO Bar does drain the smartphone’s battery quite fast, even when not “pumped to the max” power by the way.

As my FifteenReaders^tm know, the above is no big concern to me as I don’t use my general purpose smartphone as a transport, rather a separate dedicated device, paired to a dedicated small powerbank and wired with an appropriate custom cable. Nevertheless my personal use case is quite evidently not so common, so the average user looking into adopting GO Bar as a classic “dongle” to be paired to his main phone while commuting should keep its power absorbtion needs in due count.

Volume and gain control

GO Bar offers the user both the option to control volume by pressing the + and – buttons on its housing, and software syncing with the host’s OS.

When changing volume via +/- buttons the LEDs on GO Bar’s bottom faceplate briefly light up to give a visual representation of the volume level.

Host Volume Syncing is supposed to make such that when moving the system volume slider on the host (the machine onto which GO Bar is connected, be it on Windows, MacOS, Android…) then GO Bar’s internal volume changes, and viceversa changing Go Bar’s volume by pushing the +/- buttons will change the host’s volume.

Host Volume Sync is off by default. To turn it on one needs to keep the multifunction button pressed >5 seconds. The switch on is confirmed by an animation played on the upper 6 LEDs on the bottom faceplate. Same procedure to turn it off : keep button pressed >5 seconds, and a (different) animation gets played on the LEDs to confirm.

For my direct experience, Host Volume Sync works as intended on my Android devices, and on my proprietary-Linux small transport, the Tempotec V1. It works “erratically” on my Windows 10 laptop. It does not work at all on any of my different-distro Linux boxes.

After inquiring with iFi’s tech support I got no fix for the Windows problems, and I got confirmed that Linux is indeed not supported “by design”. This is bad, as this de facto prevents GO Bar to be used in pretty much any Linux Client-Server configuration e.g. in a Roon, or LMS, or similar infrastructure, unless by slamming its volume to 100% via its HW buttons, and then actionate on host’s Digital Volume which is of course far from ideal in terms of output quality.

GO Bar also has a +6dB High Gain mode. Unlike what happens e.g. on the GO Blu where gain selection follows an automatic system, on GO Bar it’s the user who has to manually set the device on High or Low Gain mode.

To toggle Gain H / L one needs to push both + and – volume buttons at the same time for >2 seconds. Similarly to the Host Volume Sync case, the “usual” uppermost 6 LEDs on the device’s bottom faceplate will play an animation to indicate the action has been carried out.

Other features

MQA Full Decoding

I won’t spend a word on what MQA itself is, of course. Google around if you wish and you’ll be overflooded with info.

What matters here is: GO Bar is a “MQA Full Decoder”. This means that GO Bar not only can (like any “MQA Renderer”) fully unfold MQA tracks on its own hardware, but that it can also:

• Authenticate the provenance of MQA tracks.
• Authenticate the Original Bit Rate of the MQA tracks.

Ifi GO Bar, Gryphon, HipDac-2 are all Full Decoder devices. Ifi HipDac, Micro iDSD Signature, Nano iDSD Black Label are Renderers.

Between parentheses: HipDac and HipDac-2 being virtually identical in terms of sound capabilities, power, etc, with the sole major difference represented by their different MQA capabilities, offered me the interesting opportunity to check the differences on a quite similar if not virtually identical situation and I could tell a quite obvious SQ improvement when listening to a few particular tracks just Rendered (HipDac) or Full Decoded (HipDac-2).

That said, I don’t personally care about MQA, nor about any of the existing digital distribution catalogues for that matter, due to the fundamental lack of good editions of the music I prefer on there.

XBass+ and XSpace

“XBass+” behaves like what an EQ expert would call a low shelf positive filter. By ear I would say it pushes lows up by 2dB-ish from 100Hz down.

“XSpace” is a “crossfeed filter”, i.e. a function that puts “some” of the right channel output into the left one and viceversa, simulating on headphones what happens when listening to loudspeakers. Within its limits (it’s not parametric, configurable etc – just a mere on/off) and situationality (effects are totally evident on some tracks, minimal on others) the trick is really nice, and I use it quite often.

My main application for XSpace are those original jazz masters from the 60ies where mixing tended to be executed by hard panning each instrument on a single channel only: crossfeed is almost magical in those cases.

Both features are according to iFi’s documentation entirely implemented in the analog domain. No DSPs are involved which promises the minimal impact on sound quality.

To turn either or both features ON or OFF all is needed is short pressing the Multifunction button on GO Bar in sequence. The two bottom LEDs on the stripe on GO Bar’s bottom faceplate will light up or down accordingly.

Alternative reconstruction filters

Reconstruction filters are an extremely technical topic and some fundamental knowledge needs to be acquired to even start to understand what the heck are they about.

If you are technically inclined a good starting point for your homework about what are Reconstruction Filters is actually this Wikipedia page.

If you alternatively would love a more vulgarised approach, there’s my article about this (or many others on the web).

Assuming you are at least somewhat familiar with these concepts, let me say that GO Bar offers the user 4 options :

• BP (“Bit Perfect”) – this actually corresponds to not applying any digital filtering. No pre nor post ringing is involved (of course).
• STD (“Standard”) – a modestly fast filter with modest pre and post ringing
• MIN (“Minimum Phase”) – a slow minimum phase filter, with minimum pre and post ringing
• GTO (“Gibbs-transient optimised”) – iFi’s proprietary filter, very fast, with no pre ringing and little post ringing. When GTO filter is applied all digital input is automatically up sampled to 352 or 384KHz prior to decoding c/o GO Bar’s internal hw.

To set the preferred filter, hold the Multifunction button down >3 seconds. The MQA LED on GO Bar’s backplate will start flashing of a specific colour. Short pressing the Multifunction botton once will cycle through the 4 possible filters, and the LED will start flashing of a different colour. Long pressing the Multifunction button again >3 sec will “set” the filter and keep it selected until the procedure is done again.

LED color codes are as follows:

• Cyan – BP filter
• Red – STD filter
• Yellow – MIN filter
• White – GTO filter

The first 3 options really sound like 3 of the 5 standard options made available on stock Cirrus CS43131 chips (full spec sheet here)

The “BP” option (Cirrus’ “NOS” filter) – will of course avoid the slightest risk of introducing any ringing, at the cost, however, of leaving artifacts all there where they are. Conceptually not recommended for redbook (44.1/48KHz) tracks, becomes a viable alternative to assess for highres (>96KHz++) tracks. But that’s theory if you ask me: you *do* want to cut the high frequency crap out. Always. I am not using this option myself.

The “MIN” option is a minimum phase, slow filter. More recommended on high(er) res tracks than Redbook ones, it introduces very modest pre and post ringing.

The “STD” option is a step in the direction of a faster filter. More recommended on Redbook tracks, although not really “vertical” as other fast filters are. Ringing is a tad more important then MIN’s one.

Finally, “GTO” is ifi’s proprietary filter called “Gibbs-Transient Optimised. It’s a very fast, minimum phase filter. I strongly recommend you read iFi’s whitepaper about why and how this may be technically desireable, or not.

As mentioned above, selecting the GTO filter also adds a pre-reconstruction systematic up-sampling passage, executed by GO Bar’s internal hardware. All incoming PCM tracks get up sampled to GO Bar’s maximum sample rate of 384KHz (if the original track’s sample rate is 48, 96, or 192Khz) or 352.8 KHz (if the original track’s sample rate is 44.1, 88.2 or 176.4KHz).

In general on higher resolution tracks one tends to prefer slower filters as those don’t risk to cut out on treble air nor spatial cues, their slowness not being a problem thanks to the higher sampling rate. iFi’s GTO filter is a special implementation offering super-fast operation, zero pre ringing and minimal post ringing. This, in addition to the up sampling conducted on the incoming stream, results in very sharp transients and “sculpted” notes. Which – as everything in audio – will then be subject to personal preference!

For my personal experience with iFi devices, and for my tastes, I found the GTO filter viable on all iFi models I tried it onto, including GO Bar, with the sole exception of Micro iDSD Signature, where I found it “excessive”. You may want to read my take about that case here.

IEMatch

For a through description of what IEMatch is, there’s my article which I recommend you to read if you are unfamiliar with the concept or I guarantee you won’t understand what follows.

Much like it happens on many other iFi’s models, GO Bar carries built-in IEMatch circuitry. The implementation does not exactly follow the same specs as the standalone IEMatch devices though.

Firstly, next to the IEMatch switch on GO Bar we don’t find the usual “Ultra” / “High” engravings, but rather “3.5” and “4.4”. The GO Bar manual quite smokily says that “iEMatch reduces the output level, so that even the most sensitive In-Ear-Monitors (IEMs) can be matched to the GO bar”. Which is only a part of what a full-blown IEMatch does. And does not offer precise figures in terms of attenuations nor output impedances to help the user anticipate what he will get by plugging IEMs of specific impedance or sensitivity.

Long story short, I asked iFi’s tech support and they provided me with the following table:

In spite of my repeated requests, iFi didn’t supply me with the other relevant information which is the Input impedance value on all those cases. Or at least they didn’t yet at the time of this article’s publishing. Looking at the figures, and comparing them with those of the standalone IEMatch models, I can only “guess” that input impedances might be in the ballpark of those featured by IEMatch 4.4, so around 40-50Ω.

Such “guess” is also corroborated looking at the Single Ended output figures: there, the lower output impedance option does not feature a much higher attenuation as it usually is the case on IEMatch devices, but rather a slightly lower one, with a phase flip involved.

After taking their sweet time, IFI did provide the missing bit of information in the end: the internal IEMatch module features the following internal upstream impedances (the impedance “seen” by the amplifier module when the relevant IEMatch setting is on).

Sadly, these figures mean that GO Bar may result not too powerful especially when switched onto its 4.4/4.4 IEMatch output combination, considering the -12.5dB attenuation involved.

Lastly, Going back to GO Bar manual’s recommendation to use the IEMatch switch to attenuate output in order to cope with extrasensitive drivers, looking at the figures it’s quite clear that the rec stands as stated only when the drivers also carry a not very low impedance (ideally, no lower than 30-ish ohm). Low or very low Z drivers (Dunu ZEN, Oriveti OH500…) will show some midbass bump due to reduced dampening, which shall have to be compensated by EQing – or just avoided by plugging a “regular” IEMAtch-2.5 onto GO Bar’s balanced output (via a 2.5-4.4 adapter of course).

Firmware

Like for most if not really all iFi devices, for GO Bar too iFi makes firmware package availables for the user to download and easily apply.

At the current time there’s only one package available, version 1.7 (in two sub-versions with just a minimal, almost “aesthetic” difference). I do recommend checking that is the version installed on the device when you get it, as the previous one (v1.48) which was installed on my review sample when I got it was quite buggy.

I won’t be surprised if iFi will make more alternative fw packages available going forward, e.g. offering different filtering options as it happens on other iFi models.

Package

GO Bar comes in a small box but with the right bundle accessories, and premium quality ones at that too.

Cables include:

• USBC-USBC 10cm cable
• USBC-Lightning 10cm cable
• USBC-USBA passthrough adapter

Cable quality is apparently top notch.

Same can be said of the black leather travel case, offering enough space for the GO Bar device itself and one or actually both of its USB cables.

Sound and power

GO Bar sounds well, and I should actually remark “very well” indeed, from its balanced output port.

As for voicing GO Bar definitely marks a diversion from that warm and midbass-accented iFi’s “house sound” typical of many other models e.g. Hip Dac, Nano iDSD BL, etc. 

GO Bar is much closer to neutrality (although still somewhat into warm-ish territory). Its sound is well bilaterally extended, with very good note body accross the board, good clarity and good detail, with very good but not over-accented bass presence and a good treble rendering.

About trebles it should definitely be noted that GO Bar delivers unoffensive high notes, and a nice, unfatiguing and nicely musical experience on one hand, while staying south of some competitor’s last mile in terms of treble energy and detail retrieval on the other. Pick your poison I guess, and as for all compromises appreciation for iFi’s choices on this will strongly depend on users’ preferences.

From the power delivery standapoint GO Bar is definitely a musclar device, although some notes are in order on this respect.

Regarding voltage swing into very high impedance drivers (600Ω) GO Bar easily promises (and delivers) the highest figure on the “dongles” market today, a whopping 7,2V.  That’s significantly higher even compared to Apogee Groove’s 5V on 600Ω. Ifi does not declare (and I couldn’t measure) the swing on 300Ω (Groove’s stays just a bit below 5V there).

Truth be told, as most if not all high impedance cans are equipped with dynamic drivers, I’m not sure to understand what the purpose of a 7V+ swing really is (“stunning” spec sheet figure apart, I mean…).

GO Bar delivers circa 1W onto a 55Ω load (always talking about the Balanced Ended output), which is definitely a huge lot for a dongle, and why it drives the likes of Shure SRH1540 wonderfully well, and SRH1840 near perfectly, too.

It delivers circa half Watt into a 32Ω driver, which is really a lot in a sense, indeed overkill for most DD, BA or other technology IEMS out there, yet (!…) not enough for higher demanding planars, which require even more current and/or they require it at lower load impedance values.

Going further down with load impedance GO Bar’s power drops rapidly (as noted above the device has a sort of hard cap on output current at approx 115mA), thus delivering “only” circa 200mW on 14Ω and circa 140mW on 10Ω. Again, such figures are higher than those on most of the direct competition, yet not quite at dongle market’s top (E1DA’s 9038SG3 delivers something similar to 600mW onto 10ohm…).

Consistently to this, GO Bar drives the likes of Final A3000, Tanchjim Darling, and even Final E5000 waaaay better then most other dongles, but does not have enough power for RHA CL2, nor of course any demanding planar overear.

As I repeatedly mentioned, all the above refers to GO Bar’s Balanced Ended output. The Single Ended output is not at the same level, neither in terms of output power nor – most of all – in terms of sound quality. Even on easy to drive loads GO Bar Single Ended Out is perceivably duller, much scarcer in microdynamics and more closed-in on space reconstruction.

Simply put, if you ask me GO Bar’s Single Ended output is to be disregarded, in favour of its Balanced Ended sibling.

Comparisons

Cayin RU6 ($250)

GO Bar is by far better than RU6 on pretty much every single count, although this is much more due to RU6 being an overall disappointing device to be honest – which makes the comparison meaningful only due to RU6’s ungrounded hype than anything else really.

Won’t spend more time on this for now, stay tuned if you wish for my piece about RU6, due Soon^tm.

E1DA 9038SG3 (€126)

The first big difference that pops to the eyes comparing GO Bar with 9038SG3 is the price of course: GO Bar is almost 3 times as much.

Another thing is power. GO Bar is more powerful on high and medium impedance loads, 9038SG3 wins big on loads from 20Ω down. In more practical words, GO Bar’s edge on mid/high impedance drivers proves useless (9038SG3’s power is enough for most drivers, and for those where it is not, GO Bar’s higher power is not enough either), while 9038SG3’s higher power on low and superlow impedance drivers allows translates in E1DA’s dongle being much more agile in driving certain “difficult” IEMs then GO Bar is.

Probably due to its performances on higher impedances, or to lesser efficiency, or both, GO Bar, unlike 9038SG3 or 9038D, is a power w**re (it absorbs up to 4W while working, which is 800mA – so it is not USB2 compliant and by far so). Oppositely, 9038SG3 is modest in terms of power needs vs its output power capabilities, and fully USB2 compliant.

GO Bar misses the harmonic compensation and masterclock customisation infrastructure available on 9038SG3, and that’s not small stuff, and offers only 4 different FIR filters to choose from instead of 7. On the flip side GO Bar literally covers the user with features one nicer and/or sexyer than the other, all of which are totally missing on 9038SG3: XBass and XSpace analog-domain effects, selectable low/high gain, integrated IEMatch, high quality integrated power filtering, and (for Tidal’s aficionados) MQA full decoding.

Sound quality wise 9038SG3 out of the box is definitely cleaner and comes across as more analythical and more energetic compared to GO Bar, which sounds more musical and more relaxing. Actionating upon its multiple tweaks 9038SG3 can be made “sweeter/smoother” though.

In the end GO Bar does give more than 9038SG3 especially in terms of overall features package in return for that much higher purchase price and much higher host power need. On the flip side 9038SG3 can power some IEM drivers which GO Bar can’t trigger well enough.

Apogee Groove ($220)

As extensively reported on my piece about it, Apogee Groove is an oddball. A badass of an oddball if you wish, but still an uncommon device, with the pros and cons one may after all expect from oddity.

Groove’s output stage is based on proprietary technology and does not support crossover filters or similar circuitry, and all too often it also powers Balanced Architecture drivers (even single-driver models) very quirkily. To cut it short, Groove is mainly if not solely intended for Dynamic Drivers, which is of course an apriori fact to seriously consider when looking instead for a “universal application” DAC/AMP dongle.

Groove swings 5V into 600Ω impedance cans which is a lot. It is indeed way short of GO Bar’s huge 7,2V although it’s worth noting that per se there’s little need for those extra 2V when driving high impedance dynamic divers.

On the opposite end Groove is less powerful than GO Bar onto 32Ω loads, but its current cap is a bit higher than GO Bar’s so it ends up delivering more power vs very low impedances like 14Ω or even 10Ω. As a consequence, GO Bar is (power wise) more agile than Groove when paired to the likes of Shure SRH1840, but the coin flips when considering Final E5000.

Groove is quite demanding in terms of host power (340mA, circa 1.5W) but with that it still stays well within USB2 compliance limits, unlike GO Bar which requires almost 3 times as much at full power levels. Beyond these differences, at the end of the day for both sticking an external powerbank onto one’s own preferred transport, and using a suitable single-leg-powered Y-USB cable is the right way to go.

Power profiles aside, Groove and GO Bar are quite different in terms of sound presentation.

Groove is way superior in terms of micro-dynamics and even more so in terms of spatial drawing: I hardly can name a single mobile DAC device better than Groove on this. GO Bar is less colored and may deliver some more subtlety in terms of sheer detail retrieval. Groove is no doubt “more musical”, GO Bar is “more neutral” (just in comparison to Groove – it’s not a “dry neutral” device taken per se).

Lastly, GO Bar’s additional features (selectable reconstruction filters, high gain option, MQA decoding, Xbass, Xspace) are totally alien to Groove.

Conclusions

In terms of their product line, GO Bar covers an evident lack in iFi Audio’s range which never offered a battery-less device before. Now they do and quite expectably their first attempt is definitely a hit.

GO Bar is a very good device. It’s in facts very powerful. While nitpickingly maybe not the single most powerful dongle around (yet still one of the top… three?) I can hardly name a direct competitor offering half of the extra features GO Bar makes readily available under the users’ fingeritps.

GO Bar is superbly design, solid, and – last in the list, but of course first for importance – sounds very well.

At the end of the day I guess its single relevant downside is the price – which is not low at all. A few other downsides are also there, but none of those seriously shadows its positives.

Our generic standard disclaimer.

The post iFi GO Bar Review (1) – Feature Packed appeared first on Music For The Masses.

9038SG3 is E1DA’s latest iteration of the 9038S model, which over time went through 3 generations – this being G3 in facts. The 9038S project, like PowerDAC, have always been designed around a balance-ended-only option. In conjunction with this third iteration of the 9038S project, howeverer, following quite a substantial flow of user requests E1DA decided to develop a single-ended (only) version, which is precisely what 9038D is.

9038SG3 can be purchased from E1DA’s AE shop, or directly from E1DA via paypal, for approx $105. 9038D has a regular price identical to 9038SG3 but it is currently not available: E1DA suspended production due to the excessive increase in chip costs – they rate that the higher price at which they would be forced to sell it would be unfair.

At-a-glance Card

A word on the manufacturer, and a few on this article

E1DA is a microscopical company. Indeed, a small family run business. The founder and key engineer in there is a Russia born guy now living in China called Ivan Khlyupin. He is an audio enthusiast, and an electrical engineer. Ivan is in charge of all hardware invention / designing, and his elder son takes care of relevant software development. The rest is chinese-cost and sadly chinese-quality contract manufacturing, which is why Ivan (literally) technically assesses and calibrates each sample one by one while or after assembly. Purest artisan’s pride DNA – which is probably why being Italian I feel a sort of natural empathy for the guy.

Their first project was the PowerDAC and it stemmed from a personal need: a low cost, powerful enough dac-amp to drive a pair of badass planar headphones. There wasn’t one on the market affordable and good enough at the same time, so Ivan DIY’d one. And then made it into a 50$ small-scale-industrialised marketable produt.

I’ve been following them for a while now and I own all their current “dongle” models (9038D, 9038SG3 and PowerDAC 2.1) which I of course purchased as a regular nameless customer.

As you will read here, and on a subsequent piece of mine dedicated to PowerDAC, E1DA’s products can easily be recognised as pure audio engineering competence concentrated into tiny, affordable little boxes. In a world overflooded by cheap and even not so cheap “meh-level” stuff, they feel relaxing like sea breeze.

Given their “vertical” technical nature, if there’s one thing E1DA lacks is vulgar-level communication. They do all they can to be super easily reachable via their Discord channel, and they are very responsive. Yet their documents, and their typical answers, are all very technical, with little to no concession to readability let alone accessibility for least competent laymen.

As a consequence high chances are their products get known, let alone “understood” and appreciated, only by already semi-skilled users at the very least. Take myself for one – I got onto them by pure chance and it took me quite some time to dig into some of the aspects of their stuff, and until now I even feel I only got part of that done.

So these are the main reasons why instead of a 4 – 5 pages small article this time I wrote an essay probably 10X longer. And I might even decide to update it in the future

Note: E1DA is to be pronounced “E one DAH”, following the sounds of the words “Ivan” and “Da” (“Yes” in Russian)

Features and description

Externals

The two devices are contained inside the same housing, a sandblasted anodized CNC-machined aluminium case, with marketing graphics (logo) and other data laser-marked in white on the black background.

Size is 48 x 22 x 9mm, weight is approx. 10-12g for either model. Simply put, either device is very small and lightweight which of course greatly facilitates it being used as a dac-amp external upgrade to any mobile device (phone, tablet, dap) which is capable of digital audio output through its USB port.

The housings’ sole apertures are the phone out on one of its small ends, and a USB-C port on the opposite end. Next to the USB-C port there’s also a tiny hole. No other elements, no buttons, no display.

Both 9038D and 9038SG3 are sold in a minimalistic small carton box, containing only the device itself (well protected). No cables / adapters are bundled. USB-C, Lightning and/or Y-USB-C cables can however be separately ordered from E1DA if needed. And they are needed. See more below in particular about appropriate Lightning cables.

Internals

9038SG3 and 9038D have very similar internal structures.

Both carry the same Comtrue USB bridge, featuring hw volume (actionable in .5dB increments) and furtherly tweaked with custom software.

Both devices’ heart is a ESS 9038Q2M DAC chip (spec sheet here) featuring current mode amplification, outstanding built-in jitter removal, and a host of user-configurable parameters including Master Clock frequency selection, FIR filter selection and customisable THD compensation coefficients (much more on this later).

Both devices have an amplification stage after the DAC but the two opamps are different on the two models. The internal power filtering also is somewhat different.

9038SG3 features an Analog Devices AD8397 (spec sheet here) opamp offering balanced end only output connectivity.

9038SG3’s internal power rail filtering structure is based on resistors and capacitors. Three different versions of 9038SG3 have been released over time. The earliest version adopted Yageo brand resistors + 2000µF capacitors. Such was a quite early version and according to E1DA shipped until a good 2 years ago.

Since then, 9038SG3 have been and still are equipped with Susumu brand (higher quality) resistors + either 2000µF or 3000µF capacitors. The higher the capacitance, the more efficient the power filtering.

9038SG3 SKU# did not change as internal equipment evolved over time. To tell which version a given unit is look at the product name engraved in white on 9038SG3’ housing: an underlined “3” (like this: “#9038SG3”) indicates a Susumu-equipped model. Looking at a Susumu 9038SG3’s housing, two or three white squares are engraved on the back side (opposed to the E1DA logo one): two squares mean 2000µF capacitance, three squares mean 3000µF.

For curiosity: the “squares” refer to the physical capacitors adopted inside. Two squares = 2 capacitors, three squares = 3 capacitors. By opening the enclosure (don’t! as you would have to reglue it later) they can easily be recognised as three lined-up orange “thingies” soldered on there.

9038D carries instead a TI OPA1622 opamp (spec sheet here) offering single ended only connectivity.

9038D also carries Susumu brand resistors, complemented with 4000µF total capacitance.

Input

Both 9038SG3 and 9038D only offer a single digital input, via a USB-C port (fully USB-2 protocol compatible).

Both carrying the same internal USB bridge and DAC chips, both support the very same digital input specs which are:

• PCM up to 32bit / 384Khz (requires ASIO drivers on Windows, otherwise limited to 24bit / 192KHz)
• DSD up to 256 (again requires ASIO drivers on Windows, otherwise no direct DSD support available)

No drivers needed for full features availability on all other major supported OS. I know of very few other “dongles” supporting DSD256 on Android.

Next to the USB-C connector, on both 9038D and 9038SG3 there’s a tiny LED. Its lighting behaviour has the following meanings (identical on both models):

Power draw

When connected to a USB host 9038D will absorb approx. 435mW (87mA) even when not playing back. Consumption while playing will be even higher of course, from approx. 500 to 585 mW (100-117mA) when receiving a PCM stream (from 44.1 to 192Khz resolution respectively), and from approx. 500 to 670mW (100-134mA) when receiving a DSD-64 to DSD-256 direct stream.

These figures are low, and even incredibly low when we consider the output this little kid is able to provide (see subsequent chapter).

Both models have an automatic Standby function, which can be enable/disabled by the user and is enabled by default out of the box. Thanks to such feature both 9038SG3 and 9038D will go into “Low Power Consumption” mode if they don’t receive any data from the host for 60 seconds – so when they are “on but doing nothing” so to say – and they will automatically wake back up when a new stream will start flowing again. Power absorption in such condition is circa 50% of the minimum required on quiesced playback status, so around 205-210mW.

I’ll provide more details much down below on how to enable/disable the Standby & Mute functions, and some caveats.

Considering the huge power especially 9038SG3 is able to deliver on mid, low and extralow (see below) the above figures are impressively low. Just a bit more than 600mW (only 120mA!) when decoding DSD256 and at high volume seems like a joke.

9038SG3 and 9038D are by far fully USB2 compliant, so no problem with any Android phone and if the phone has a relatively modern battery (3-4000mAh or more) 9038D / 9038SG3 will not meaningfully jeopardise battery duration.

They can be used on quite a few iPhone models too but choosing the Lightning cable that make it happen may be tricky. See much down below a dedicated chapter to this issue.

I thought about inserting a digression at this point regarding power demand on so-called “dongles” (like 9038SG3 and 9038D) and some considerations stemming from there, but my notes soon developed into something deserving a separate take. So please be patient, I’ll be issuing a standalone article on this ReallySoonNow.

Output

As mentioned above, 9038SG3 and 9038D carry a different amplifier module, which is the most significant difference between the two models.

9038D has one 3.5mm single-ended output port, which supports connection to earphone/headphones of any impedance and technology, and also connection to the single-ended input of an Amplifier device.

9038SG3 has one 2.5mm balanced-ended output port, which exclusively supports connection to earphone/headphones of any impedance provided they have a balanced-ended termination. Connecting Amplifier devices to 9038SG3 is not supported, not even via such Amplifiers’ balanced ended input ports – failure to comply to such exclusion will most surely physically damage the device.

9038SG3 features very significant output power:

• >600mW@10,3Ω
• >550mW@16Ω
• >340mW@32Ω
• And 3.3Vrms @0dB vs high impedance loads

9038D output power is also quite interesting albeit definitely lower:

• 120mW@16Ω
• 180mW@32Ω
• 200mW@40Ω
• And 2.75Vrms @0dB vs high impedance loads 

Both offer extremely low output impedance, around 0,1Ω.

All figures come from E1DA, and correspond to measurements conducted at room temperature, and at 1% THD+N

Sound and performance

I’m going to report about 9038SG3 first, then I will more easily cover 9038D in terms of differences from that.

In its stock calibration situation, so Out Of The Box as they say, 9038SG3 is easily one of the cleanest, most detailed and fast (short transients) dongle I ever auditioned. Indeed, from the sound structure standpoint it rivals much higher class devices.

Notes are exceptionally well separated and clean, while on the flip side they come accross somewhat lean, and this contributes to a general feeling of “scarce musicality” and “excess in detail” if that even makes sense of course.

Leveraging on its internal harmonics compensation generator via the Tweak9038 app, 9038SG3’s “presentation” can be altered to be made a bit “more musical”, “warmer”, even “tubey”. Sure, it takes some will on experimenting of course but possibilities are there. The effect however is not that of flipping the whole presentation into a dark/warm one. See much more on this below.

9038SG3 also has nice spatial rendering – “soundstage” as we call it is definitely OK. Not the level at which a Groove renders depth and height on highres (>96KHz) streams, but that’s related to Groove’s FR being uncommonly flat much beyond 20KHz more than anything else (see here for the full story). Barred that, 9038SG3 has nothing to envy to any other dongle I auditioned, at any price, on this respect.

9038SG3’s lack of sound coloration is obvious, but that’s possibly the second most outstanding feature I noticed right away – the first being the very high amount of power (current) this little box is able to deliver onto low impedance, low sensitivity drivers.

Give or take 9038SG3 delivers 550-600mW into 14 and even 10Ω loads: a sort of mini nuclear plant, perfectly capable to “move” deep insensitive drivers e.g. Final E5000 (14Ω 93dB), RHA CL2 (15Ω 89dB), Hifiman HE400S (22Ω 98dB) and pretty much any low impedance planar IEM you can think of, and with some plenty of room to spare.

As load impedance goes up 9038SG3 stays an uncommonly powerful thingie but starts to show its ropes of course (hey it’s an effing dongle…). For example SRH1840 (65Ω 96dB) are still kinda no problem, but Hifiman HE560 (45Ω 90dB) are no-no.

Lastly, 9038SG3 max voltage swing on high impedance drivers (3.3V) makes it more than decently fit to drive the likes of Sennheiser HD600 (300Ω 102dB/V = 96,7dB/mW) – on which the “tube emulation” is worth a spin, maybe even two… – see below how.

Once all the above is clear, describing 9038D is relatively simple: it’s virtually identical in tonality, timbre, cleanness and technicalities, but delivers way less current on the low loads end, and also more modest voltage swing vs high impedances.

On the former part I guess I can call this yet another example of how a “balanced” scheme is not a requirement to the purpose of outstanding quality on sound output. This consideration apart, 9038D like 9038SG3 sounds magnificently well, and it can be tweaked and changed exactly like its sibling so it’s up to each one to leave it “more analythical” as in its OOB tuning, or a bit “more musical”.

The latter part reflects into a quite different applicative span for 9038D compared to 9038SG3.

While 9038D can still properly drive the likes of Tanchjim Oxygen (32Ω 110dB/Vrms=95,5dB/mW) or Final A3000 (18Ω 98dB) or the recently hyped 7Hz Timeless (14.8Ω 104dB), other drivers like Final E5000, SRH1840 or other more seriously harder to drive planars are at various degrees not ideal, or not viable altogether pairs.

Similarly although less seriously on the higher impedance end: 2.75V are OK to make HD600 sing, but there won’t be much room to compensate in case of low-level recorded tracks and/or level-punishing EQ schemes.

Very succintly put: 9038SG3 delivers incredibly clean sound and very good technicalities and so much power that it can act as a one-stop-source for all IEMs on the market bar none, and most Headphones too, bar high demand planars only. 9038D offers the same sound qualities, can drive “most” IEMs and a few Headphones from of a single ended connection so without requiring cable swapping.

Before I forget: 9038D is virtually immune from hissing when paired to oversensitive loads (Campfire Andromeda, Penon Volt…). 9038SG3 does hiss a bit on the same drivers.

And lastly: as quickly mentioned above and explained in better detail down below, 9038D can be used as a pure DAC connected to a downstream amplifier. Given its outstanding sound profile and its ease of integration on pretty much any host OS, such application might be something to seriously look into, in spite of its external “superpocketable dongle” format.

Comparisons

Hidizs S9 Pro ($119,00)

An educational case insofar as we are talking about almost identical-priced devices, and based on the very same DAC chip (ESS 9038Q2M).

First of all, both 9038D and 9038SG3 sound simply obviously better than S9 Pro. Their presentation is much more linear, clean and detailed compared to S9 Pro’s balanced output. S9 Pro’s high mids very easily tend to “overdo”, and the treble end lacks some air in comparison. S9 Pro also lacks any form of tweakability. S9 Pro’s single ended output is almost unaudible to me quality wise.

On the power delivery standpoint, 9038D’s output is marginally more powerful than S9 Pro’s balanced ended out, and more than twice its single ended one. 9038SG3 is roughly 50% more powerful than S9 Pro at 32Ω, and even most importantly S9 Pro runs into a serious current shortage from right around 16Ω on down, while 9038SG3 still provides something like 600mW vs 10Ω loads. In practice: 9038SG3 easily drives E5000 and planar IEMs, S9 Pro can’t even start trying doing that, or doing that at a comparable level.

Simply and perhaps a bit unforgivingly put: S9 Pro is a toy compared to 9038SG3, and less desireable (although by a smaller margin) even compared to 9038D.

Cayin RU6 ($250)

As you’ll read on my separate take on RU6 (due Soon^tm), in less than a million words RU6 has in its unique timbre its main if not sole reason to be. Its internal R2R technology implementation delivers in facts an audibly different nuance to notes, and that is likely the reason for the ticket price for the curious modest-budgeted audiophile.

The rest is unimpressive at best, often underwhelming. The R2R timbre is audible on NOS mode only – which sadly requires high-res (>= 96KHz) digital tracks to be fed from the outside, as its noise, distortion and FR rolloff on Redbook material is nearly comical – which is even a worse pity if we consider that amongst all that noise one can hear above decent imaging and note body. That’s probably why many say RU6 should exclusively be used on OS mode where reconstruction of 44.1KHz becomes decent-ish, with an at least reasonable sense of space, and much less audible noise. Sadly, the OS circuitry is deltasigma based which defeats most if not all the purpose in this case.

Be as it may, RU6 never comes even close to 9038SG3 / 9038D in terms of clarity, cleanness and detail retrieval.

Power wise the situation is similar to S9 Pro: RU6 is a quite modest-powered device, delivering “just” 213mW@32Ω on BE (similar to 9038D on its single ended, and much less than 9038SG3), slightly more than half on SE, and most importantly dropping quickly below 16Ω so in this case, again: nothing special on E5000, and forget cheecky planar IEMs, etc – unlike what is fully allowed by 9038SG3.

Unlike S9 Pro, RU6’s relatively modest output power at least comes with modest host power requirements – that’s a quite important note. Together with the fact that all the above is attached to a 2X price tag.

IFi GO Bar ($329 / €329)

Also Soon^tm you’ll read my full article on GO Bar. In the meanwhile…

The first big difference with 9038SG3 / 9038D is in the price of course: almost 3 times as much. Better be something serious in there doesn’t it.

Another thing is power. GO Bar is powerful on high and medium impedance loads. It swings a whopping 7.2V into 600Ω (more than twice 9038SG3), and 475mW into 32Ω (40% more than 9038SG3).

Sadly, it hits against a wall of current limitation (circa 120mA) as impedance goes down. As a consequence GO Bar (balanced out) drives Final E5000 (14Ω 93dB) with good athleticism, although with less headroom compared to 9038SG3, but it won’t properly drive the likes of RHA CL2, which are instead perfectly managed by 9038SG3.

Probably due to its performances on higher impedances, or to lesser efficiency, or both, GO Bar, unlike 9038SG3 or 9038D, is a power w**re. It absorbs up to 4W while working, which is 800mA – so it is not USB2 compliant and by far so. Not all Android phones will drive it to its full power then: a laptop is required, or a battery in parallel with a phone. Oppositely, 9038SG3 and 9038D are very modest in terms of power needs vs their output power capabilities, and fully USB2 compliant.

One more thing is features. GO Bar misses the harmonic compensation and masterclock customisation infrastructure available on 9038SG3, and that’s not small stuff, and offers only 4 different FIR filters to choose from instead of 7. On the flip side GO Bar covers the user with features one nicer and/or sexyer than the other, all of which are totally missing on 9038D and 9038SG3: XBass and XSpace analog-domain effects, selectable low/high gain, integrated IEMatch, high quality integrated power filtering, and (for Tidal’s aficionados) MQA full decoding.

So in the end yes, GO Bar does give quite something more than 9038SG3 in return for that higher purchase price and a much higher host power need. I see 9038SG3 as a device delivering similar or better sound quality, and similar to much higher output power onto IEMs compared to GO Bar, in a nofrill package and for a fraction of GO Bar’s price.

Apogee Groove ($220)

As extensively reported on my piece about it, Apogee Groove is an oddball. A badass of an oddball if you wish, but still an uncommon device, with the pros and cons one may after all expect from oddity.

Groove’s output stage is based on proprietary technology and does not support crossover filters or similar circuitry, and all too often it also powers Balanced Architecture drivers (even single-driver models) very quirkily. To cut it short, Groove is mainly if not solely intended for Dynamic Drivers, which is of course an apriori fact to seriously consider when looking instead for a “universal application” DAC/AMP dongle.

That said, Groove swings 5V into 300Ω and 600Ω impedance cans, making it obviously more energetic compared to 9038SG3 and of course 9038D too, and to all other dongles on the market with the sole exception of iFi’s GO Bar.

On the opposite end Groove delivers less current than 9038SG3 which is why it can power Final E5000 (14Ω 93dB) well, but falls dramatically short when applied to RHA CL2, which 9038SG3 eats for breakfast instead. Always from the current delivery on mid/low impedance loads standpoint, Groove is OK-ish on SRH1840 (on the limit, let’s say), while 9038SG3 dances them around more “brilliantly”, and with much more headroom for sure.

Like 9038D, and unlike 9038SG3, Groove can be exploited as a DAC connected to a downstream amplifier.

Groove requires nearly 3 times the current 9038SG3 or 9038D do from their host, which is still USB2 compliant but a huge point to consider nevertheless.

Power profiles aside, Groove and 9038SG3 are very different in terms of sound presentation. Groove is way superior in terms of macro-dynamics (imaging) and even more so in terms of spatial drawing: I hardly can name a single mobile DAC device better than Groove on this.

On the other hand 9038SG3 is obviously less colored and has better subtlety on detail retrieval. Flipping the coin, Groove is gorgeously more “musical” than 9038SG3 and you won’t change the latter anywhere near the former via TCC tweakings.

In less than a million words: where applicable and therefore apriori comparable, Groove is more musical and sexyer, 9038SG3 is more technical, cleaner, sharper. Groove is more powerful on Sennheiser cans, 9038SG3 is way stronger on planar IEMs. I’m so happy I own both, and I would again buy both as these two together cover all possible needs south of a much higher end (and priced) DAP or battery-powered DAC-AMP.

Tweak9038

E1DA developed a companion Android app for their 9038SG3 and 9038D dongles. It’s called Tweak9038.

The purpose of the app is giving the user access to most if not all customisable parameters offered by the ESS 9038Q2M chip, and it indeed succeeds in doing it reliably and quite easily too. The down side is that those parameters are quite technical stuff, and customising them to “make sound better” requires knowing what one’s doing – no worries though, you can’t damage anything if you do it wrong.

There are some limitations to the app, including:

• It only works on Android OS, and they are not planning to port it anywhere else. It anyhow technically might never work on iOS due to Apple limitations.
• It only communicates with 9038SG3 / 9038D via USB.

So the device must be plugged into the Android device where the app is via a USB cable, and only then the dongle configuration can be accessed, seen and changed. The modified configuration can be saved into the dongle’s own non volatile memory, and it will stay there even when the device is plugged onto a totally different device, even if not carrying the Tweak9038 app, and/or if not even Android-based.

Some may consider the app cost (10$) also a limitation. They are wrong. This app is totally brilliant, and adds a lot to 9038SG3 and 9038D’s value and to their uniqueness, as I’ll explain in a bit.  It’s actually very cheap for all it offers and how accurately and reliably it works. If you have some strange problem with “expensive apps” (expensive? 10$? Well, ok…) just mentally add 10$ to the price you pay for the 9038SG3 or 9038D and you won’t fail noticing they will stay two incredibly inexpensive dongles in light of the over-amazing quality they offer.

So quit whyning already, and buy the bloody app to support its developers

As I mentioned above, Tweak9038 exclusively supports 9038D and 9038SG3. It does not support 9038SG2 or earlier. 9038SG3 is anyhow quite significantly better than 9038SG2, and still affordable enough that if I were a 9038SG2 owner I would not think twice about buying a 9038SG3 as an upgrade.

Enough foreword. In summary, Tweak9038 allows to:

• Customise the Minimum, Maximum and Default levels of the device’s hardware volume scale.
• Generate harmonic distortion (yes, you read well), and even do that diversely following playback volume.
• Customise the DAC’s clock frequency, and apply different values automatically based on the track sample rate.
• Select amongst different available reconstruction filters, and again choose different ones automatically depending on the track sample rate
• Enable/disable Standby and Mute options
• Save “sets” of all the above parameters under user-defined names, and recall + apply them to the device whenever liked.
• Lastly, scratch everything off just in case you need to return to the exact configuration and calibration that very device had when leaving the factory.

While some of these features may seem easy, others are quite obscure or at least they were to me. After some extensive use and a lot of curiosity applied, I must say this has been dual fun me: once because these tweaks resulted in amazing sound output, and twice because they gave me the occasion to study their rationales in deeper depth then I ever did in the past – which of course now helps me put other devices in a much more realistic and technically more correct perspective.

Here below I will go through most of the “stuff” I experimented and sometimes learnt. YMMV of course: if for you all this is already bread and butter well, just jump through .

A special mention is deserved by E1DA’s support team, which is easily reachable via their Discord channel, and always available to provide competent and precise input.

Volume scale calibration

Out of the box, 9038SG3 / 9038D hardware volume is set to go from -127.5dB to 0dB, with the default value set to 0dB.

Hardware volume values are to be intended as “attenuation” values. So 0dB means “no attenuation”, that is, “leave the amp output fully undampened”, aka “go ahead, kill my ears!”. Oppositely, -127.5dB means “drop the output volume down by 127.5dB” which is a huge dampening. It equates to “shut the F up!”.

The Default volume value is the value the device will set the volume at whenever it is turned on. Given it’s a battery-less device, it will turn off every time it is disconnected from a host, and on when it is reconnected back. Default volume set to 0dB means: whenever you turn the dongle up set the volume to “full unbridled sound out”. Sounds scary doesn’t it. And yes, it is scary.  

Until now I talked about the dongle’s “internal” (“hardware”) volume values, but we don’t normally “directly see” those values. What we most often see is a volume slider on our music player software, or even on the Operating System of the machine the dongle is connected to. Such slider is usually labelled as going from “zero” (meaning “zero volume”, 0%, or “silent”) to “100” (or “100%”, “full volume”, “full loudness”).

If our music player app’s volume slider is “linked” to 9038’s internal hardware volume (and it usually is – either automatically or by manually switching it on as you can do on most sw player’s settings) then out of the box the music player’s “zero” volume maps into 9038’s  -127.5dB attenuation, and on the opposite end the music player’s “100” volume maps into 0dB attenuation. And the “default volume position” at power-up will be “full volume” position, or, the music player’s latest used volume position (depends on the situation).

Twea9038 app allows for customising all 3 of these default hardware volume settings. But why should you?

Well first of all: the default volume level. It is much, much safer if we set it low, instead of high, let alone “full up” (i.e. “0dB”). This is simply because sooner or later we risk to forget to bring the volume down before hitting “play”, and doing this with the default volume level set too high blasts so loud sound into our ears that we can (seriously!) be permanently harmed.

Secondly: the lower end value. -127.5dB is such a huge attenuation value that unless we plug an extremely oversensitive driver in, starting from “volume full down” will require moving the volume slider a lot before getting some decent sound pressure (“loudness”) out.

First impression in such case might be that 9038S has very weak amplification (“hell… I need to bring it to 80% to get some loudness even on these simple IEMs…!”). But that’s not the case. Moving the volume slider further up the sound pressure will raise very strongly, up to deafening levels.

The real problem is that the “Min” (starting) value is way too low for practical purposes. Setting the minimum hw volume value to a more convenient setting “fixes” this. Which setting is exactly recommendable depends on the impedance and the sensitivity of the actual drivers we “usually” pair 9038 with. For my sets I found that a value of -80dB is OK.

Lastly, let’s consider the upper end value of the scale.

9038SG3, especially, is powerful. Quite seriously powerful I mean, as it can swing 3.3Vrms into a high impedance driver, or – on the flip side – push up to half watt (!) into a 16Ω load.

The downside of all such power is that (again) a wrong move with the music player’s volume control can deafen the user for good, especially if this happens while using IEMs or Headphones which are not “impossible” to drive for 9038SG3, which is like… 95% of the existing ones (and 100% of those in my possession).

Setting the upper end of the hw volume scale to a value lower than 0dB is a safeguard in such sense. Once set at (say) -10dB, this means that when (willingly or by mistake) music player’s volume is slammed to “full up” 9038SG3 won’t release all its possible power onto the drivers, bit quite a bit less.

Similarly to the bottom end value case, the “right” (“most practical”) value to choose for the upper volume end largely depends on which earphones / headphones are part of one’s rotation. If most drivers are very sensitive stuff like Campfire Andromeda, Penon Volt or the like, a pretty low value is recommended! Oppositely, if drivers at hand are hard to drive planars, or insensitive and/or high impedance DD’s, it may be best to leave the value near 0dB, or just below that.

Besides writing values into Tweak9038’s GUI, there’s also another “hidden way” to adjust the Maximum hardware volume boundary by “fiddling” in special ways with the host OS volume slider. This works on multiple different hosts (MacOS, Windows or Linux, and Android – in such last case UAPP is required).

Here’s the scoop:

Bring the host volume slider to 0% (so “all the way down”), and then quickly raise volume + bring it back down to zero% + raise volume again. The “gesture” is like “pulling the volume slider up from zero and quickly slamming it back down, then bouncing back a bit”. The “bounce up” should not exceed 50% of the totale volume slider run space. Do this “bounce” trick 3 times in a row and this will result in a -30dB Max Volume value being instantly set. A sort of quick way to impose a hard “volume limiter”, if you wish.

Bring the host volume slider to 100% (so “all the way up”), and then quickly lower volume + bring it back to 100% + lower  volume again. The gesture is specular to the previous one, it’s like “pulling the volume slider down from 100% and quickly slamming it back up, then bouncing back down a bit”. Same caveat as before: the “bounce down” should be less than 50% of the total volume slider run space. Do this 3 times, and Max Volume value will increase by 3dB. Do this another 3 times and it will increase another 3dB, so 6dB in total. And so on.

I don’t know if you agree but I find this so brilliant…

I have two minor negative points to mention, too.

One: even if a front end music player app directly controls 9038SG3 / 9038D’s hardware volume, it has however no way to know its absolute Min and Max values as they are set inside the dongles by the Tweak9038 app. So for example it did happen to me to spend some sweet time wondering (while swearing) why the heck my 9038SG3 could not make a certain IEM sound really loud even at “full volume” (on the player), only to remember much later that I had set the Max hw volume value to a low level myself.

Two: someone may feel more comfortable if the GUI mentioned volume values in Vrms units in addition to dB units. This would be easy to implement as there is a precise formula linking the two, and someone did already put this in the wishes box to E1DA, so I trust it will eventually happen. Until then, we can calculate them manually as follows : 

Output voltage [Vrms] = FullScale voltage [Vrms] * 10 ^ (Attenuation [dB] / 20)

So for example an attenuation of -10dB results in:

• (For 9038D)       2.75 Vrms * 10^ (-10dB/20) = 0,87 Vrms full scale voltage
• (For 9038SG3)   3.33 Vrms * 10^ (-10dB/20) = 1,04 Vrms full scale voltage

A “distortion generator” – to help reduce distortion

Sounds like a paradox doesn’t it.

As there ain’t such thing as a “perfect” device, of course there is no such thing as a “totally non-distorting” DAC/AMP. Alternatively said, our audio gear’s THD will always be >0.  

Audio equipment engineering and manufacturing of course includes keeping distortion as low as possible, for as low industrial cost as possible of course.

There are structural (I like to call them “static”) causes for harmonic distortion: the quality of the electronics, the cleanness of its implementation, etc. A badly engineered device based on crappy components will produce distortion under any operative conditions; under the very same operative conditions, a better engineered device based on higher quality components will produce lower distortion.

Then, there are those which I call the “dynamic” causes. Operative temperature, for example, is a factor: electronics do change their behaviour with temperature, and that makes a difference in their audio behaviour. EMI/RFI interference even more so. And load: depending on the impedance of the connected headphones the source device will “behave” a bit differently and will generate different distortion patterns – in very general terms, distortion goes up as impedance goes down. Even volume: the same amp will distort less when operated at mid-volume, more when pushed at the top of its capabilities. Etc.

So not only the most skilled manufacturer in the world will be unable to deliver a zero-THD device, but even their best device will have an always somewhat variable THD, as some of that THD depends on how that device is being “actually used”.

Is there a way to cope with such distortion depending on usage conditions? Well… in part, yes.

If we know which parasite harmonics is the device generating under certain usage conditions (e.g., when a given headphone is connected), then we can create some harmonics ourselves which are “equal-but-opposite” compared to the unwanted ones, thus effectively “cancel them out”.

The ESS 9038Q2M chip does have a sort of built-in harmonics (compensation) generator, and indeed that’s what E1DA exploits to first of all calibrate each and every 9038D or 9038SG3 unit prior to shipping.

By design 9038SG3 and 9038D aim at a THD of -125dB as a target value. A value of THD = -125dB is considered ideal, -124dB is considered “standard”, and -123dB is the threshold below which that very unit will be sold as b-stock.

Given what we just noted, we wonder: under which effective usage conditions are such THD values verified?

E1DA reports: they calibrate and measure their 9038SG3/9038D devices with a 32Ω load, and at an operative temperature of 25°C. Then they observe the (unwanted) harmonics during playback of a reference signal while going from min to max volume, and they set compensation values into the the ESS 9038Q2M chip harmonics generator to cancel them out.

Such compensation values are finally burnt into the device’s firmware before shipping. No matter how hard we subsequently mess with the harmonics generator for experimenting etc (see below), we can always go back to E1DA’s original “factory values” by tapping on Restore Factory Settings on the Tweak9038 app.

During real life use we will of course plug all sorts of different impedance drivers into our 9038SG3 or 9038D dongles. When their impedance will be significantly different from 32ohm those factory-pre-set compensation values will be less effective to the purpose. What can we do then?

One: experiment “by ear”.

Simply reach out for the Tweak9038 app and modify the values on the “THD panel”. Doing this while playing music, the result is hearable in real time. So anyone can judge by oneself if the change is adding or removing distortion, and by how much.

As distortion often also depends on the volume at which the device is being made to work, the Tweak9038 app allows to define 3 sub-ranges of the entire volume range. Range 1 goes from -127.5dB to Low Threshold (blue-green), Range 2 from Low Threshold to High Threshold (orange), and Range 3 from High Threshold to 0dB.

Threshold values can be freely modified. To set them either drag the bullets atop their vertical bars, or tap their values (the blue-green and orange figures atop the bar) and directly key the new number in.

Once Ranges are defined, tap on each of the 3 “THD Edit” buttons, at the top, and enter harmonic correction value for each of those Ranges.

It is possible to generate even (2^nd) and odd (3^rd) harmonic values. Each value must be entered in dB, and by ticking the Invert flag we flip the harmonic’s phase.

Similarly to how it works for the Thresholds, harmonic values can be input either by dragging un/down the 2 orange bullet atop the animated “graph spikes” at the centre of the screen, or by tapping on the number values within the frame on the upper-right.

Suppose we don’t want to take volume variations too much into account: how do we define a “flat” correction, all equal for the entire volume range?

There’s 2 ways to do that: either input the exact same values into “THD Edit” for all three ranges, or define Threshold values such that… only Range 3 is effectively ever active (i.e.: set both Low and High Threshold to -127.5dB), and input correction values only under Range 3’s THD Edit space.

Two: go the engineeristic way

As I very briefly mentioned above, THD is typically inversely proportional to load impedance. Which means that E1DA’s factory calibration, centered upon a 32Ω load, will deliver ideal results for 32Ω but results will still be much more than decent at higher impedances; viceversa it will “need some help” – so to say – when pairing 9038SG3 or 9038D with sub-16Ω IEMs.

To find out as accurately as possible which new values optimise a 9038SG3 (or 9038D) when paired with a specific earphone / headphone, some equipment, and following a similar procedure, to what E1DA uses and does in-house to pre-calibrate 9038S3G and 9038D will be required.

And guess what: E1DA develops and sells such equipment. It’s called Cosmos, and does exactly that (and much more). Here is the link to the description – I will not go in more detail here, this article is already long enough isn’t it.

A “distortion generator” – to actually add some distortion

We do all we can to get distortion-free DACs. We even calibrate them in respects to our headphones, one by one, to compensate for load-dependent distortion… why the hell would we want to “add” distortion???

As I mentioned above, 9038D and 9038SG3 come with whopping -125dB (or so) THD which is a monumentally good value for such a device and especially price class. 

And in facts they do sound… clean. Holy cow they really do! 

Tell you what: maybe a tad too much ?

E1DA 9038SG3 and 9038D’s harmonics generator can (incredibly in a sense, but really) help also for this case. By fiddling with the THD compensation values we can “add some colour” to sound, making it “more musical” in a sense.

There’s even a way to simulate the voicing of a tube amp – that’s mainly about playing with the 2^nd harmonic. The Tweak9038 app even offers “tube emulation” presets, those are labelled “SE” – there’s one for 9038D one for 9038SG3 – all it takes is loading them, and they can be furtherly tweaked of course.

When on your quest for more colored sound you may also want to remember that

• the lower the MCLK, the lower the DR (more on this below)
• linear filters often tend to make notes less sharp (more on this below, too)

Playing with 9038SG3 / 9038D’s sound tweaking gauges is just amazing This video (by E1DA) shows a live demo of the game.

Setting a custom clock frequency

Tweak9038 allows to set the ESS 9038Q2M chip’s Master Clock frequency at 3 different values: 12.5Mhz, 25MHz or 50MHz.

A higher clock frequency lowers the noise floor, produces better note definition, sharper attack, better space reconstruction, but generates more high order harmonics (higher distortion).

Lower clock frequency is the opposite: less distortion, a bit higher noise floor, softer note contours, more “intimate” stage.

To give an idea, 12Mhz has lower THD vs 25MHz but is 1-1.5dB(A) worse in terms of SNR/DR.

The effect is indeed quite apparent especially if you have a trained audio ear already. If you are not particularly ahead in your critical listening experience, try “extremising” the values: slam all THD compensation to zero, set clock to 50MHz, and chose a minimum phase filter – you should hear all notes definitely more “sculpted”.

By the way, the fact that Tweak9038’s “Tube emulation preset” profiles are only defined at 12Mhz frequency is indeed consistent with the above: music comes across softer (not fuzzier of course, definition is still there), less “carved in stone”. Like tubes do.

Similarly to the other areas of intervention, for MCLK selection too Tweak9038 app allows to pre-set which clock value to use depending on the track’s sampling rate, and save the full association table under a custom named preset file, which one can load and apply at leisure.

For my taste, lower clock speeds are a better compromise on lower sample rates – and higher clock speeds “fit” (my tastes) better on higher sample rates.

Reconstruction filters: why we need them, and why Tweak9038 is cool

Takes as it should be, deeply understanding reconstruction filters would require a treaty on signal processing. If you are technically inclined an elementary starting point might actually be this Wikipedia page.

The ESS 9038Q2M chip adopted inside 9038SG3 and 9038D offer 7 different reconstruction filters (Linear Phase Slow and Fast, Minimum Phase Slow and Fast, Apodizing, Brick Wall and Corrected Minimum Phase) + 1 “ESS-Reserved” filter.

Assuming you read the above, or you know from before and even better than me, Linear Phase Slow and Fast, and Minimum Phase Slow and Fast filters don’t need much presentation I guess.

Brick Wall and Apodizing are variations of a Fast Linear filter. Corrected Minimum Phase is a not-very-slow Min Phase filter. The R (ESS-reserved) is similar to the Apodizing, but with less ripple.

Linear Phase Fast Filter

Linear Phase Slow Filter

Minimum Phase Fast Filter

Minimum Phase Slow Filter

Apodizing Fast Filter

Hybrid Fast Filter

Brick Wall (fast) Filter

9038SG3 and 9038D allow the user to freely select their preferred filter – which happens via the Tweak9038 app of course. Not so many other dongles offer the same possibilities (at any price, by the way).

Veeery widely said, I personally tend to apply Min Phase Slow to >=88.2KHz tracks, and Min Phase Fast to Redbook tracks.

What I find absolutely brilliant, and unique, here is that the Tweak9038 app makes it possible to map which filter is to be used based on the sampling rate of the incoming digital file.

So I can set things up such that (e.g.) on all tracks < 88.2Khz a Min Phase Fast is automatically used, while the DAC automatically switches onto a Min Phase Slow filter when resolution goes above 96Khz. Or whatever other pairing you might instead find best for your ears

Standby and Mute

9038SG3 and 9038D have built-in automatic Standby and Mute functions.

Standby will set the device in Power Saving mode if it does not receive data from the host for 1 minute. This is of course very nice to reduce power consumption when you are not actively listening to music. Power Saving mode reduces consumption by 50%.

Mute will also turn DAC output off while the device is in Power Saving mode.

There is a single drawback in leaving Standby and Mute working their automatic way: when 9038 starts back receiving again data from the host while in Power Saving mode it may induce annoying and sometimes quite loud “pops” on the drivers. The more annoying and lower the higher the drivers’ sensitivity.

Standby and Mute can be turned “off” by accessing Tweak9038 Settings panel, and just tap to remove the flags on the two options.

Settings persistence and “Presets” management

This part may seem a bit confusing at first, at least it was for me. Let me try to make it simple and straight.

The Tweak9038 app makes all of the above illustrated 9038SG3 and 9038D parameters visible to the user, and allows the user to change them.

As soon as you plug either 9038D or 9038SG3 into the Android device running Tweak9038, its currently active parameters are read-in, and shown by the Tweak9038 app.

Whenever any change is made to one of the parameters shown by the Tweak9038 app while the dongle is connected, such new value is instantly saved onto the connected 9038SG3 / 9038D non-volatile internal memory.

Due to such non-volatility, all values will persist even after unplugging the dongle from the Android device running Tweak9038, and after plugging it onto a totally different device, regardless of such device’s OS.

So again: Tweak9038 shows the values which are written “inside the dongle”, and allows the user to “prepare”, “tune” his 9038SG3 / 9038D how he prefers, save the values back into the dongle, and use it, so tuned, wherever he wants, without ever needing the Tweak9038 app once again.

Clear till now? Good.

In addition to the above, Tweak9038 allows for saving “full sets” of such parameters. Such sets are however saved onto the Android device hosting Tweak9038.  You do that by tapping on the Save frame, on the main app screen, and then giving the preset a name.

Again: Presets are not saved onto 9038SG3 / 9038D. They are kept local to the Tweak9038 app.

Existing Presets can be accessed by tapping on the Preset frame, on the main app screen. Once there, Presets will be found under 2 different categories – accessible by tapping on the first 2 buttons atop: User are those previously saved by yourself, and Official are those supplied by E1DA.

After tapping on a User Preset, it’s possible to SET, EXPORT or DELETE it.

By tapping on “SET”, the Preset’s values are written all together and all at once onto the dongle’s non-volatile memory, and saved. Exactly the same as if they were input one by one by hand.

User Presets can be Exported. Which is meaningful as Presets can also be later Imported – for instance in case a friend wants to pass us one of his sets, or to acquire a special Preset developed by E1DA which is not included inside the standard Tweak9038 app distribution set.

Official Presets cannot be Exported (no need to) nor Deleted (so you can’t mess up). Therefore, when tapping on an Official Preset the system just asks for confirmation before applying its values, and that’s it.

One last note: whenever a Preset is saved into Tweak9038’s workspace, all of the configuration values of the currently connected dongle are automatically saved onto the named file, including Min, Current and Max Volume figures. But, when a Preset is recalled from storage, and “SET” (applied) onto the currently connected dongle, the system asks wether Volume parameters need to be also Set, or those need to be left at their current “live” values.

Other stuff

Using 9038D with an external Amp

It is possible to exploit 9038D’s 3.5mm single ended phone out as a preamp out, and connect it to a downstream Amplifier. It’s quite logical to assume that, very likely if not always, 9038D will be connected to a desktop transport of sort for this application – like a Windows or Linux machine.

To get best results it is recommended to use the Tweak9038 app (see above) to:

• Apply appropriate THD compensation values
• Disable Mute and Standby
• Set Max Volume to -3dB (this way Max Vout will be 2 Vrms – this is only required if this is the max allowed input value for our Amp)

It is optionally possible to save a Preset for this, especially in case one plans to dedicate 9038D to this application only occasionally, and needs therefore an easy way to switch back and forth between these settings and others more appropriate for mobile use paired with headphone or IEMs.

All easy.

Sole doubt is: how do I devine the “appropriate THD compensation values” to apply when 9038D’s load is represented by the amp? No worries.

The 9038D has a nice matte-finished black metal housing. I would call the front side the one with the E1DA white logo engraved on, of course. Let’s flip it to the back side. Near the end corresponding to the USB-C plug there are some other minuscule-font-size engravings. On my unit I read:

Calibrated unit:
DR 124.7 / 125.4
TCC  2 / -70

I already mentioned far above that E1DA calibrates each and every unit upon manufacturing. These figures actually regard my own unit (so will in general be different from anybody else’s unit).

DR refers to my unit’s Dynamic Range, and the two figures refer to the left and right channel respectively.and mean:

Dynamic Range : 124.7 dB LEFT / 125.4 RIGHT

THD Compensation Coefficients: 2  /  -70

TCC stands for THD Compensation Coefficients, and that’s what we are looking for now, as those figures are what’s needed on my very unit to minimise distortion on “No Load Condition”.

As I explained above, THD changes based on various dynamic situations, one of which is the impedance of what gets connected to 9038D’s output (the “load”).

So here E1DA is telling me what’s the compensation to apply when I connect my 9038D to… nothing ?!? Well let’s dig better into this technical wordage.

Connecting “nothing” to the output can be said in a more electrical-engineering-friendly way as “connecting to an open circuit”.

When voltage is applied to an “open circuit”, no current will pass through. “Of course… there is not even the effing wire!!”. Well… (again) an electrical engineer would rather say that there is a “wire”… with an infinite resistance (!).  

And finally, amplifiers have very high Input Impedances. Not “infinitely” high, but “very” high nonetheless.

Now let’s connect the dots: those TCC values reported on the back of 9038D’s enclosure are the settings needed to minimise distortion when nothing is connected to 9038D’s output, i.e., when something with infinite resistance is connected there. So, they are a good approximation of the figures needed to minimise distortion when something with a very high – albeit finite – resistance is connected. Like an Amplifier, for example

How do I do that? Of course via the Tweak9038 app. Reach for the THD panel. Set both Low and High Threshold to -127.5dB. This way only Range 3 will ever be “active”. Open Range 3 panel and (in my specific case) set 2^nd harmonic to 2dB, and 3^rd harmonic to 70dB, while also ticking (only) 3^rd harmonics Invert checkbox to reflect the “-” sign.

Should I plan to use 9038D as a “fixed volume” DAC, and only regulate volume on the downstream connected Amp, I would also want to set its Min Volume = Current Volume = Max Volume to -3dB, this way effectively forcing the device to always output 2Vrms flat.

MasterClock and Filter selection panels have nothing to do with the output connection, so I will leave my usual settings map in there.

And finally, I will save the whole thing under a User Preset called e.g. “9038D Dac”.

Heck! This way I overwrote my 9038D’s factory-imposed THD settings, those offering the least distortion when using a 32Ω IEM. And only now I realise I did not save the previous configuration into a User Preset before modifying it  

No worries. Original factory-recorded THD compensation data are hard coded into the firmware. By accessing Settings / Restore Factory Settings on Tweak9038 app I can swing back to those values in no time.

Using 9038SG3 with an external Amp

TL;DR: don’t.

Some Amplifiers offer balanced input, and their owners would prefer exploiting that route, especially when the Amp also offers a full balanced internal structure.

Too bad that, simply put, such connectivity is not supported by 9038SG3. I found E1DA’s tech support explanation to why is this the case so efficient that I can’t find a better way then quoting that directly here:

DAC and Amp need to be grounded to each other to ensure safe operation. The headphone output of the 9038S has four pins: Hot/Cold Left and Hot/Cold Right. There’s no GND pin. Therefore, it is not possible to ground it to your Amp. People have ignored our warnings before, connected their 9038S to external Amps via a 2.5mm to 2x XLR adapter, and have fried their 9038S as a result. The output really is for headphones only. If you want something that you can use both with headphones/IEMs and with external DACs, get a 9038D.

https://discord.com/channels/483873307251310592/608625162115612693/1001215598203768883

Case closed.

If you really don’t want to use the Tweak9038 app

If you are really really really unwilling to pay the 10 bucks for the app, then – even if you shouldn’t deserve it – there’s a B-plan for you: carefully flashing some pre-cooked firmware made available by E1DA themselves.

The firmware flashing package exclusively exists for Windows OS. Additionally, it requires Comtrue ASIO drivers to be pre-installed to operate correctly.

E1DA makes all the required sw packages available as free downloads of course, and they come with a collection of various alternative firmware versions ready to be flashed on either 9038D or 9038SG3.

It’s of course needed to select the files inside the folder called like the device model which is supposed to be updated of course.

Inside each “model” folder there’s a single file named “Tweak […]”. That is the firmware required for the Tweak9038 app to work. So basically, it’s the firmware that comes preinstalled from factory. Once one of the other firmware files gets flashed in, the Tweak9038 app will not be able to work on that device anymore, and to restore its functionality it’s required to re-flash the Tweak[…] firmware first.

The various files contain non-app-tweakable firmware configurations, quite clearly described by their names.

It’s worth nothing that:

• Firmware names containing “noSTBY” disable the automatic Standby feature
• Firmware names containing “SE” simulate a Tube amp sound signature (set 2^nd harmonic to -50dB)

Special notes about iPhones (with a final Android hint)

Don’t be misguided [too much] by the above-mentioned power drains imposed by 9030D or 9038SG3 on the host, going well beyond 100mA which is known as a hard apriori limitation on Apple’s design.

In spite of that, both 9038SG3 and 9038D can be connected to iPhones via a Lightning connector cable but… whether it will work or not depends on the cable and on the iPhone specific model (!). Some models won’t ever work, other models won’t work with some cables (no power to turn 9038SG3 or 9038D up at all), and finally shall work with some other cables.

E1DA maintains an incredibly helpful if rustic worksheet – available here – collecting internal and users’ experiences: which cable works (or not) with which iPhone model. Check both the Yes/No and Voltage tabs to have the full information you want. The sheet also includes leads on where those cables can be purchased.

Once said it is possible, of course it stays legitimate to wonder whether it’s convenient to indeed connect 9038SG3 / 9038D alone to an iPhone, thereby heavily shortening its daily battery durability.

I myself use 9038SG3 (and Groove for that matter) on the go with “transport pack” made of a tiny DAP + a small lightweight powerbank, kept together by some Bluetack and connected to the dongles by means of a special Y-USB cable.

But that’s another story, and applies to quite a few Android phones too. This specific topic will be covered by a separate article of mine which will be published, well, you know already when.

Considerations & conclusions

You really got through all this article reading it all till here? Heck! I owe you a coffee at the very least. You deserved it.

As I hope I suceeded in saying and explaining why, E1DA 9038SG3 is a very good battery-less DAC-AMP, and an even more brilliant product overall. Unbelievably powerful, drives all nastiest IEMs including no matter which planar IEM, and most HPs as nothing – surrenders only against the most demanding planar overears. This, while offering outstanding sound clarity and superb detail retrieval.

9038D is also extremely good, indeed as good as 9038SG3 sound wise with the sole caveat of output power capabilities which are lower than 9038SG3 but still higher than 90% of the other mobile sources around. Can be the right choice for fixed-cable IEMs and anyhow most other IEMs around bar (some) planars only.

The Tweak9038 app further allows to “play” with the device timbre.

For how much ya’ll know I love my two Grooves, if I were to recommend “one” dongle only for true audiophile use at the lowest possible cost I would name 9038SG3 without the shadow of a doubt. Excellent, indeed.

As already mentioned at the beginning, the 9038SG3 and 9038D I am talking about have been personally purchased.

Very much lastly: I’ve had as always loads of fun going through all this with my audiophile friend Simone Fil, also an enthusiast E1DA user, and much deeper than me in technical audio competence – which of course I ruthlessly exploit. Quite some of the above content is the direct descent from our late evening chats and common findings.

Our generic standard disclaimer.

P.S. – last minute news

On their Discord channel Ivan just recently announced he found a good successor for 9038D’s opamp. Same power, even a bit better THD and DR. It’s code-named 9038D6K as it will also have 6000µF power filtering capacitance, up from 4000 on og 9038D.

So it seems 9038D will be available again… Soon^tm.

The post E1DA 9038SG3 and 9038D Review – They Know What They’re Doing appeared first on Music For The Masses.

It took me so long to make myself a decently clear idea about this topic that I eventually took a breath and put it down in form of a single (hopefully) organised story. Enjoy!

The problem we have, and how does it matter

DACs’ business is receiving digital files and re-convert them into analog electrical signals. Such signals are then amplified and sent to drivers (big loudspeakers or small IEMs, it doesn’t matter now) to become audible music.

Sadly while doing its job a DAC, any DAC, fatally produces “spurious copies” of each note. .

Such unwanted “note replicas” are higher frequency copies of their relavant intended note, and such “higher” frequencies are beyond the humanly audible range. That’s why we call such artefact notes “Ultrasonic images“.

If they are not audible why the heck do we care about them? For two reasons.

First reason: the frequencies at which these notes are generated are not audible by our ears, but are significant for our physical drivers (the tweeters in our speakers) – which overheat and overwear and can even break if they receive them or they receive them “too loud”. Indeed, on a wider horizon, the DAC might also produce or anyhow transmit various other forms on unwanted high or very high frequency “notes”.

For this reason alone we should find a way to avoid or kill such images, like any other ultra high frequency stuff.

The second reason is a bit more complicated.

On certain conditions when two notes close to one another are played, a sort of “reciprocal disturbance” happens between such two notes. That’s called “Inter-Modulation Distortion” (IMD).

Let’s get this picture as a conceptual example:

Suppose our “real” notes are the two big peaks in the center. Sometimes, they will “autogenerate” those shorter and leaner ones you see at their sides, one on the left (lower frequency), one on the right (higher frequency). The picture shows a case with only 2 spurious notes generated, but they could be more.

Now let’s suppose that the left real note in our example actually is one of those unwanted Ultrasonic Images produced by our DAC. As we said above, it will be on a too high frequency to be heard. Let’s suppose it’s at 21Khz. So of course it won’t be audible but big chances are that it’s “left side baby copy” might well be “distant enough” from “mummy” as to fall below 20KHz, so within the hearable frequency range.

That’s why this is called “fold-back effect”, and that’s the second important reason why we need to help our DAC getting rid of high frequency images generated during the reconstruction process in the first place.

Summary: DACs produce spurious ultrasonic images. Those need to be taken care of to prevent damage to drivers, and audible sound alterations.

The solution, and its drawbacks

How do we get rid of the ultrasonic images? With a filter.

That is, with some circuitry that (simplistically put) makes sure that once the DAC has done its job reconstructing our wanted music + those extra unwanted notes at higher-then-audible frequencies, a cut down is imposed on all frequencies above the maximum humanly audible level – effectively “filtering” them off.

No ultrasonic images, of course, so also no “fold-back” IMD-generated audible spurious notes.

Was it so easy? Well no, not so easy.

This filter we need is an electrical circuit and as such not “perfectly ideal” as we would dream it to be. If by my description above you imagined a sort of guillotine blade cutting the crap out, with no side effects, well… no.

What we would ideally want it a filter that leaves audible music totally unoutched, and kills all and only those ultrasonic images. Instead, real world filters will either let some of those ultrasonic images pass through, or apply some change to audible music, or both.

Depending on how it’s realised, the filter will have inaccuracies and side effects that we can’t entirely avoid. Such imperfections can first of all be more or less important depending on the original music’s sampling rate, and besides that the unwanted audible sonic changes might be more or less unwelcome by each particular user.

So that’s why it’s in a sense nice if a DAC device offers a choice amongst different filters: we will end up choosing the best – or least bad, if you wish – one depending on our tastes.

So let me explain about these filters imperfections and how they impact on our music. Let’s start with the inaccuracies.

Fast and slow filters

To describe how these filters work we use certain methods and graphical representations. Here is one.

This graph shows the behaviour of 3 different filters as they are implemented on a certain DAC. Forget about the specific DAC model now, this is just to describe in principle how these filters work and inhowmuch can they be different from one another.

This graph plots the attenuation (the reduction of “loudness”) that each filter imposes to the analog signal (our music) that’s coming out of the DAC.

On the X axis we have the frequencies. Remember human audible music gets as high as 20Khz.

On the Y axis we have the output “power” reduction which that filter applies, expressed in dB. Zero dB = zero impact. -110dB = shut up!

The signal being processed (the “music”) in this specific case is white noise, so a sort of “artificial music” composed of notes of all frequencies, all at the same power level. The white noise sampling rate in this example is 44.1Khz.

If there weren’t any filter, the graph would be an horizontal line accross the entire graph space, stuck on 0dB.

Let’s look at the graph lines, starting by following them from their upper left point, where they hit against the Y axis, and going towards the right.

All lines are equally flat up there, one on top of the other, and they all indicate zero dB attenuation (marked on the Y axis).  That means: all those filters apply no attenuation at all on all frequencies until… (follow the plots going right by at the same time reading down on the X  axis) …at least until 18KHz.

If we now keep “going right” from 18KHz, we see the various plot lines spreading apart.

The green and the purple lines are the first which start to “go down” right after 18Khz. The red line stays instead flat horizontal, stuck on 0dB attenuation, until above 21KHz.

The green line “drops” faster than the purple one. At 20KHz the green line is already at approx -12dB, while the purple line is still around -5dB. The green line “drops dead” (-110dB attenuation or so) just above 22Khz. The purple line reaches the same “full attenuation” level not before 28KHz.

So what does this mean?

We know that ultrasonic images can only happen above 50% of the input sampling frequency, in this case 22.05KHz.

The graphs tell us that the “green filter” (i.e. the filter represented by the green line) gives full attenuation to all signals lower than just above 22KHz so our green guy will kill all our ultrasonic images, we won’t get any.

Viceversa the purple filter will take its sweeter time before dropping down, and will let quite some ultrasonic images pass through. All those until approx 28KHz will pass, although progressively more and more attenuated as their frequency goes up.

Both the green and the purple filter also have a visibile drawback: they both start their job from 18KHz, which is still within the audible range.

While “dropping faster” to be able to cut everything above 22Khz, the green filter applies a stronger cut on the higher part of the audible range (-12dB or so at 20KHz, which is a lot). The purple filter is more permissive with images, but also less punitive on trebles (only -5dB or so at 20Khz).

Let’s look at the red line now.

Its vertical part is visually roughly parallel to the green one, it’s just offset towards the right by 2KHz.

So how does the red filter work? Test if you got the drill from above.

Summarising: all these filters have some sort of inaccuracy. These of graphs tell us that which sound frequencies are cut down by our filters, and how much power do our filters take off from them (zero, full, or something in between).

One filter makes fewer frequencies pass, the other lets more of them through. And this, already, prompts us to wonder which option is best for us.

But before that, let’s investigate about filters’ side effects.

Ringing, everywhere

Let’s talk about the frequencies (the musical notes) which our filter lets through.

Shall those notes pass through really unmodified ?  Or will the filter introduce any audible modification to them?

To study how these filters behave in terms of transparency we use a different test procedure, and graph. While before we submitted “all frequencies at once” (white noise) to the filter, now we’ll oppositely submit just one note to the filter – an as “neat”, quick, sharp and clean note as possible. And we’ll plot what our device (the filter) outputs as a result.

We call such single clean input note “impulse”. And the output is an “impulse response”.

We can imagine to create a pure impulse as a digital file where all samples are set at zero value (representing pitch black silence of course), and just one sample has the digital value of a single, loud note.

Assuming that note plays at a frequency which the filter is not attenuating, an ideal filter will of course reconstruct precisely that note in analogue form, and nothing else. Will a real world filter behave differently?

This is a good first example of impulse response graph.

(For the illustrations I am helping myself with the pictures published within this article, which is very well written by the way, and might also be interesting for you to take a look at).

On the X axis there’s time. On the Y (vertical) axis there is Voltage, so ultimately output power.

The plot of an ideal filter applied to the DAC’s reconstruction of an ideal impulse would be a dead flat horizontal line at 0 Volts (total silence), until a precise time where voltage pops up for a very brief moment (the Impulse), and then comes immediately back down to zero, to continue dead flat onto zero forever thereafter.

The picture above is instead the plot of a real world situation.

The peak in the middle is what we expected: the impulse note reconstructed based on the single <>0 digital sample in the input file. So far so good.

The unexpected part is those ripples to the left (so, in time, before the note) and to the right (so right after the note).

If the note we are talking about is audible, those ripples will be audible in terms of (faint) “fringes” to the notes. That note will not come across perfectly neat as it should. Ideally we would not want those ripples then, but let’s not commit suicide yet. The ripples before the note are called “pre-ringing” by the way, those after are called “post-ringing”.

The graph above reports the impulse response of a fast filter.

If you remember, a fast filter is what in general seems most desireable as it cuts ultrasonic images quickly. Now we find a quite important drawback though: a fast filter adds unwanted note alterations both before and after each note.

Let’s see how a slow filter behaves:

The central part is very similar of course, but the ripples (the “ringing”) are minimal in comparison to the fast filter case.

So we know from the previous chapter that the slow filter is less effective in terms of ultrasonic images cutting, but it turns out to be much less impactful in terms of changes on the musical notes themselves.

Ringing, single-sidedly

The fast and slow filters we saw in the previous chapter produce more or less ringing both before and after their impulse response, are sometimes called “linear phase” filters.

In tech talk, the “phase” of filter refers to a delay. Too complicated. Here, let’s concentrate on the effects more than on the technical causes.

On linear phase filters the impulse energy is in its maximum part concentrated at the exact time when the note needs to be played, with some fringes of energy “escaping” some time before and some time after the pulse. The two filters we saw before obey to this decription: the high peak is in the center of their graph, and the ripples are at both its sides.

Other filters can be created which instead concentrate the maximum part of the incoming energy at the beginning of the impulse response, and overflow only onto the time past that. Like this:

A filter like this is called a “minimum phase” filter.

The impulsive part (the main note) is totally similar to the fast, linear phase sibling. The ripples, however, are only after the note.

Open brackets. As you’ll notice this graph has “a lot” of ripples – although only on the right side – so this will be a “fast, minimum phase” filter. Close brackets.

The amplitude of the ripples after the note is bigger (ripples are taller) compared to that of the ripples before or after the note observed in the “fast, linear” case. Indeed, it’s double. The fact is that the total energy is the same: in the linear case it gets distributed “a bit before and a bit after” the “main event”, here instead the ex-flow it’s all “delayed to after” the peak note.

Soooo……. there is no pre-ringing!

Lastly, let’s consider a “slow, minimum phase” filter. How will it be?

As expected: similarly to the “slow, linear phase” case the ripples are minimal compared to the fast alternative, and similarly to the “slow, minimum phase” case the ripples are only after the impulse. No pre-ringing.

There are quite a few other ways to build reconstruction filters but it was never my intention to write an encyclopedia. These 4 cases are quite enough for us to get where the main trick stays.

Ringing ?

I mentioned before that we would ideally want ringing not to be there at all.  Do we really ?  Less than it seems, actually. That, for two reasons.

First reason.

It’s crucial to note that our internal auditory organs… do ring too.

When our internal ear receives an “impulse” (a relatively strong, sudden sound) our organs will vibrate in conjunction with that note… and for a short time after it. This is so physiological that our brain… knows!

Our perception of a “neat” (not reverberating, not “persisting”) sound is the result of an originally neat note hitting our ears. There, some human post-ringing is added, then it all goes to the brain which… subtracts (tares off, if you wish) the ringing and tells us that the note was “neat”.

On the flip side, our ear organs do not pre-ring when they receive a sound. So when our brain gets some “thing” coming before a sound it will “remark it immediately” as “odd”.

Given this, a filter-inducted note post-ringing will actually be much less important that it seems on paper. As long as it is modest in amplitude and/or length it will be masked in full or in part by our physiological system.

The filter’s pre-ringing instead, that will most matter. As our brain does not know what pre-ringing even means (!) a note affected by even modest pre-ringing will first of all be decoded as “strange”, “not totally right”.

Second reason.

Ringing will not happen on each single note played by the DAC. Oppositely, it will happen only on somewhat “unlegit” notes: e.g. those connected with clipping, or with pre-existant ringing.

As it looks, then, we should never actually have ringing as long as we listen to good quality digital tracks. Do we?

Yes, precisely. Too bad that “good quality digital tracks” are surprisingly rare. Ever heard of Loudness War for example? Good audiophile-quality masters are, indeed, extremely rare to find. The last place where to look for some in my experience is an online streaming service (any online streaming service). But that’s another story.

Connecting some dots

Let’s summarise what we learnt until now:

• Bloody DACs create unwanted artefact notes
• Such artefacts only appear at frequencies higher than 50% of the original digital file sampling rate (so e.g. for a 44.1KHz file they start happening above 22.05KHz)
• Those artefacts need to be removed as they create problems both audible (affecting the music) and non audible (affecting audio devices)
• To remove them we use filters, which are engineered in different possible ways
• Filters of different types have different scope/efficiency spans, and come with different side-effects.

About filters we learnt that:

• Fast filters start killing notes very few KHz higher in frequency from the filter’s inception frequency.
• Slow filters oppositely let notes of much higher frequencies vs their inception frequency pass through.
• Linear filters (be they fast or slow) add some note imperfection called ringing both before and after the notes
• Minimum phase filters (be they fast or slow) add stronger ringing after the notes, but none before them.

Finally, we noted that the human auditory system has some modest physiological post-ringing, but no pre-ringing.

So in summary: filters do kill unwanted ultrasonic frequencies but they introduce some issues while doing that, so there’s a tradeoff to be found between how strict is the filter and how much of we can afford bear, or even we appreciate, filter’s drawbacks.

To make an example of such tradeoff let’s go back to the graph we originally used to learn how these filters macroscopically work. Here it is, do you remember it?

The specific case is an ordinary CD-quality 44.1KHz digital track.

Which will be the pros and cons of those 3 filters ? Try yourself !

Size Sampling frequency matters

It’s not entirely accurate to take 20KHz as the top frequency that matters to our purpose of an optimal music reproduction.

While it is in facts true that no human hear can perceive sounds higher than 20Khz frequency (and only being young, too!), it is also true that music is not made of sounds only, but also of timing.

Information about silences (their length), echoes (their timings) etc, can involve frequencies much beyond 20Khz. Sileces, echoes etc are what adds spatial accents to the sound.

Of course not all digital tracks will be sophisticated enough to even contain such high(er) frequency space-related information in the first place. A bad recording is a bad recording, and do we want to talk about bad mastering? Sheesh.

Also, it does take a not too ordinary DAC to appropriately make use of such >20KHz (see my article about Apogee Groove for an example, and some hint on the reasons for this).

When talking in general terms as we are doing today, however, we must appreciate that frequencies above 20Khz are not easily expendable as one may think at first.

Which of course makes our filters story even more dramatic. If you read back the conclusions of the previous chapter – those commenting on the pros and cons of the Green / Red / Purple filters we used for training – this is the reason why I underlined that all three filters would harm soundstage a bit.

Now on to some good news, for a change.

Suppose we have our digital tracks sampled at sample rate much higher than ordinary CD-quality. Let’s say they are at 96Khz.

The DAC will start producing its ultrasonic artifacts starting from (remember?) one half of that, so in this case artifacts will not have frequencies lower than 96/2 = 48Khz.

That’s a lot above our human hearing upper limit, and also up quite enough to let most if not all those high(er) frequency samples partaking to spatial cues that we mentioned just before.

Connecting more dots

If we can count on digital tracks sampled at rates much higher than 44.1KHz, then the situation changes when it comes to the pros and cons of fast and slow filters.

Let’s recycle our training case: the green / red / purple filters. Let’s assume the track is sampled at 96Khz instead of 44.1KHz this time. Artefacts will start to appear from 48KHz on up.

The Green filter would in this case drop dead at 48KHz, killing all artefacts. Perfect like before. It would start from 4Khz lower, so from 44KHz. Unlike the previous case, this would not harm treble air, nor likely spatial-related frequencies at all. Perfect (thanks to the higher incoming sampling rate!). As a fast, minimum phase filter there would be some just modest post ringing. Good!

The Red filter would drop dead at 50KHz so it would let 2KHz open for Ultrasonic images. Not totally welcome yet surely unaudible. Fold-back images would be there, but too far off from audibility threshold for some to pass it. Nice-ish. Inception would be at 45KHz so still zero harm to treble tones nor to treble air, and hardly any chance of chopping on soundstange. Good. Still a fast linear filter so its pre-ringing smoothing notes down would be there. So-so (tastes dependent).

The Purple filter would drop dead at 54KHz so even more (unaudible but potentially harmful) ultrasonic artefacts and little to none of their IMDs would reach down enough to fold-back into audibility. Not so good. Inception at 44KHz so again zero harm to treble notes, treble air and likely to soundstage too. Good. It’s a linear filter so smoothing notes down due to pre-ringing. So-so (tastes dependent).

Choices

So in the end which reconstruction filter would I choose – if given the option of course ?

Like in many other aspects of audio (and of life)… it depends.

First and foremost, it depends on the the digital music’s sample rate. Sarting from higher up, fold-back images generated by the slow filter will hardly if ever fall into audible range. Which makes a slow filter’s main benefit – its significantly more modest ringing – a much more “affordable” option of course.

This is so much true that whenever at all possible and doable at a decent technical level, up sampling lower rate (44.1/48KHz) tracks onto at least twice the rate is a no brainer to me!

Another discriminating element to consider is one’s preference in terms of note transients. I do tend to prefer short transients, cleaner timbres. Therefore I sharply lean towards minimum phase filters, which unlike linear filters are immune from pre-ringing.

When my tracks are desperately 44.1KHz, and my transport does not give me a [viable] up sampling option, then I’m kinda forced to opt for a fast filter of course. Which will generate much stronger ringing compared to the slow alternative, and will make minimum phase even more desireable vs linear!

Lastly, all of the above is to be taken “in principle”, and not necessarily “in practice”, as the path going from principles to practice passes through actuality, in our case it pass through the particular DAC devices we actually have available. Our DAC may not offer a choice of different filters. Or, may offer only a very restricted choice. Or even may offer various filters of which some well implemented and others badly implemented.

Very long story short: the above is how it should work. How it does work can only be told by our ears – with good peace of all die-hard objectivists.

The post Reconstruction Filters For Dummies appeared first on Music For The Masses.

Logitech Media Server is a piece of software, and it’s well described here: https://en.wikipedia.org/wiki/Logitech_Media_Server

The highlights:

• It gets music files or streams from a plethora of diverse origins (files on local storage, files from private or public cloud storage, streams from other private streaming platforms eg another LMS or from public services eg Spotify, Tidal, Qobuz…), transcodes formats if need be, and streams/sends the songs towards compatible “renderers”, i.e. music players which in their turn feed the actual audio hw (DAC -> AMP -> Transducer)
• It’s available for Windows, Mac and of course Linux, including a few specialised Linux distributions
• It therefore runs on “usual” X386/64 hw, Apple hw, and – what matters most – on a huge array of low cost and especially low power consuming SBCs (Single Board Computers)
• Considering today’s available hw performance level, its system (CPU/RAM etc) requirements for an even fancy home setup are unbelievably low
• It’s free (GNU)

LMS does not “play music”, it just collects music, and manages its stock and access, and distributes them to the actual music players (the “renderes”).

As a “renderer” you can use either a preconfigured hardware device e.g. a Chromecast, a Squeezebox, etc which can be reached via various channels like wired ethernet, wifi ethernet, BT, AirpPlay and protocols like DLNA etc, or you can install a compatible receiver software on a general purpose system e.g. your pc, your Mac, your xbox, etc, or finally you can build a “hardware rendering device” from scracth, which is indeed my case and the good news is that it is way less complicated than it seems.

The physical system acting as LMS server may also have a Renderer inside, to “manage files, and play them out” from the same machine. Even in such case though that machine will keep being able to stream audio to other external Renderers.

While streaming audio to Renders, LMS can also manage keeping them “in sync”, resulting in simultaneous music playout in different rooms, for example.

So summarising:

• LMS is “the server”, the manager of the whole system. It cllects and indexes music files, makes them browsable, and sends (“streams”) them to companion devices called “Renderers”.
• The Renderers are the devices which get digital music data streams from LMS and push them into a locally connected DAC>AMP>Speaker/HP/IEM stack.

How I deployed it

No I won’t write a full book on the infinite ways to deploy an LMS infrastructure. I’ll just describe how my own infrastructure has been organised, for you to take inspiration

My LMS is running on an SBC-class computer.

In my specific case we’re talking about a BananaPi M2+ (recently upgraded from a NanoPi NEO2 , which is now dedicated to other tasks) but it could easily be “any” RaspberryPi, or dozens of similar alternatives.

My Banana-LMS server is wired-connected to my main home network switch.

Another SBC-class server is acting as a general file server for my home needs, that’s where my digital music files are deposited, and my Banana-LMS accesses them via NFS. In a simpler setup, I could plug a USB drive right onto Banana-LMS of course.

Once installed, the LMS server publishes an HTML interface. Which means that from any one of my PCs, or Laptops, or wifi devices (phones, tablets, daps…) I can access it as long as I can browse onto its address.

LMS creates an index of all music files on the storage, much like any “media manager” application does (including those inside DAPs).

Let’s now suspend the LMS description for a sec, and pass on to the Renderers.

Renderers

My first Renderer was is – guess what – a RaspberryPi Zero W.

As you read above, a Renderer is a device which takes the digital music data from the LMS server and sends them to the actual DAC. To do so, some sort of “music player” application is required. My choice on that is PiCorePlayer which I like as it offers two great features at the same time: it’s super-easy to install, and it sounds wonderfully well.

PiCorePlayer on Linux platform is distributed complete with a bare-bones Linux distribution, ready to work and do its job – and its job only – at the best of the hosting hardware ability. The maximally stripped-down, highly-optimised nature of PiCorePlayer’s underlying Linux distro is crucial to its performance as a low noise music player.

It’s good to note that PiCorePlayer also optionally carries LMS built in. That means that in an even simpler situation I could have avoided keeping a standalone Banana-LMS device acting as a server, and I could have elected one or my Renderers to the role of Renderer and Server for itself, and for all others.

Once at least one Renderer (the PiCorePlayer) is installed and running, I can go back onto LMS’s webpage – called from a phone, while sitting on the sofa – and I’ll see a Renderer available in my network. At that point I can browse and choose a song from LMS’s visual index, a Renderer to send it to, and click PLAY.

I have a total of 3 RPi-base Renderers active right now.

Allo

My first Renderer is the aforementioned RaspberryPi ZeroW, and it’s called Allo, at it hosts an Allo MiniBoss I2C DAC card.

Such mini-network-DAC box is subsequently connected to an Allo Volt+ amp box, giving juice to a pair of Roth Audio OLIRA1 bookshelvers. Depending on my seasonal feelings, the Allo renderer and its downstream line is either installed in a sitting corner in my livingroom, or takes some place on my desk and around it as a nearfield setup, for some non-overpretentious-quality audio output.

Groovy

My second PiCorePlayer-based Renderer is a Raspberry model 3B+, which is sitting on my desk, next to my PC.

To this Renderer one of my Groove units is normally plugged in, and it’s the resource I tap onto when I want to enjoy some specific drivers directly paired to the Groove. Hence the name “Groovy”

Indeed, Groovy is also what I typically use as a realiable, reasonably-clean USB host to audition other USB-input DACs or DAC/AMPs that I happen to receive from time to time.

Fun

The third PiCorePlayer Renderer is named “Fun”, and it’s based on a more recent RaspberryPi model 4B.

This is the support device for my “main desktop stack” for headphones at the moment, ending into my Burson Fun headphone amp – hence of course the name given to the PCP device.

Cutting the laptop out

Until some time ago I used to have a 4th Rendering point represented by my Windows Laptop itself. You do that by installing a windows app called SqueezeLite-X, which takes care of talking to the backend LMS server – much like a PiCorePlayer does. I used to, as I said, then more recently I quit using my laptop as a host for musical playing for good.

Long story short: the level of perturbance generated on a multipurpose, multimedia, gaming-level laptop like mine is significant. While a filter like the iFi Nano iUSB 3.0 undoubtedly helps reducing much of that, it’s nevertheless quite evident that cutting the problem at the source instead of fixing it later is a smarter option, when available! So I quit employing a noisy platform like a laptop in the first place, and now excluisively adopt less-noisy-to-begin-with ones for my musical pleasures.

More about LMS

So LMS allows me to browse my local digital music collection, and “play out” my preferred tracks on any of my connected Renderers.

I can reach that and browse through it via a normal web browser, or a nice number of supporting apps – either fully dedicated ones (e.g. OrangeSqueeze or others, available on Google Play) or multi purpose ones (e.g. UAPP, Neutron, HiBy Music, or any other app featuring DLNA-Controller capabilities)

If music tracks are decently tagged LMS also does some nice job in terms of music collection presentation. You can also have it acquire and cache album art, album and artist info, and even lyrics from various online resources.

If you access it via a browser you can choose the GUI “skin” you prefer, or customise your own if you are skilled enough. The UI is not remotely as phantasmagorical as on higher rank systems like Roon, but still quite pleasing nonetheless, with the non-secondary side-benefit of being… free!

And there’s more: a host of additional features can be activated / removed in forms of plugins.  Some examples:

Format conversion. LMS can convert to/from countless digital formats “on the fly”, i.e. while actually sending the file to the Renderer (and the DAC attached to it). So for example it can convert (e.g.) a DSF 128 track into a 24 bit / 176.4KHz PCM FLAC file while sending it to an endpoint which won’t natively be able to decode the DSF itself. Big caveat: this does require quite some muscle! My BananaPi-LMS does not have enough for that, for example. So for all DSD-level tracks I have, I took care of creating their relevant PCM (FLAC) version, and stored it as an alternative version of the same album on my NAS, and let LMS access them too.

Tidal, Spotify, Qobuz integration. Adding account credentials to LMS, it will connect to those services and make them available for browsing from within its GUI, and for reforwarding to the Renderers – just like it happens for any local-resident digital track.

UPnP / DLNA integration. I partially already covered this above. Any DLNA-capable mobile device (phone, tablet, dap, etc) can home interact with LMS. If the device only has DLNA-client support, you can only use it as a sort of Renderer – i.e., you need another device to browse LMS and push music from LMS into the DLNA-client device. If the device has full DLNA-controller support, instead, then it will be able to browse LMS in full authonomy, and call tracks to play onto itself. This – of course – can happen from “inside home”, and from “outside home”, provided you made your LMS accessible from the outside of course, and that your outgoing internet bandwidth is at least decent.

Airplay integration, Webradio integration, etc etc etc

Summary and conclusions

So, summarising: Logitech Media Server can be the heart and the brain of an entire domestic audio infrastructure.

What it ultimately offered me is:

• A centralised visual database of all my local digital audio material
• Some nice integration with extra artist / track information
• Access from within home, and from outside (via VPN).
• An “easy” way to keep digital audio transport off from general purpose computer hardware and OS (higher audio quality)

All this at an extremely low cost profile: LMS and its various Rendering companion sw packages are free of licenses, they can run on ARM-based hardware which is both inexpensive to buy (compared to an X86/X64 class alternative) and to electrically power up.

LMS served me well as my main audio infrastructure until a few months ago, when I switched over to Roon. I’ll write another piece on that… soon(tm).

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