Oh dear. This video is so troublesome to me, because it seems like such a let down compared to your usual fund of knowledge. Worse, its spirit breaks the usual ethic I admire you for-namely eschewing the manipulative marketing strategies of certain manufacturers. Starting in many ways with the brands you champion in this video, I’ll leave that there. Can we take a bit of a broader, pragmatic view of what one truly needs to address the digital to analog conversion process in our audio setups? Because if we started with another, less costly DAC in the first place, we’d have to revisit wholesale what the true utility of a DDC is. Let’s start by staying in the digital realm, because that’s where we are. We have a signal path to attend to. And despite all of the complexities you addressed in this lecture, the only practical purpose a DDC fundamentally serves is to solve compatibility issues when connecting digital audio sources and DACs. If we start with a great measuring, SOTA modern DAC (which for the last five years at least would be a delta sigma DAC, period, full stop), they would have a USB input, as would the vast majority of signal sources in the past decade. If we don’t have either of those options, we may need a DDC-but either way, adding one to their signal path would have NO impact on “sound quality” at all. I believe you know this already for sure, but as we’ve chosen to stay in the digital realm, we should recall that digital audio is not sound, and “bit perfect” is a silly marketing term, not an audio science term. Also “timing” to input receivers with modern DACs has been handily resolved by buffers-and especially with asynchronous USB connections (and delta sigma DACs), the DACs clock is the master (as you pointed out). That’s why USB has been adopted as the reference, most practical standard for high end, super high resolution digital audio streams, and IIS has remained a superfluous expense we tolerate due to the marketing manipulations of PS Audio (their reasoning is based on another epic scam involving circumventing Sony’s SACD DRM rights, but that’s a topic for another time and place). That is also why informed, sensible people who care about the money they work hard for don’t bother anymore with the antiquated R2R ladder DACs you discuss in this video. They are technologically inferior, obsolete, yet paradoxically more expensive-and unless they include a killer integrated amplifier, they’re literally stuffing cash in the furnace. Absolutely, the DACs you reference here sound great-but they don’t sound any more or less great than a $300 SMSL option with an ES9038pro or a $400 refurbished Topping d90 with an AK4499. Going back to the digital domain, the only “sound” that emerges from the back end of a DAC is the sound of the original master, varying only by the degree of noise, distortion and dynamic range that comes with. And the newest (meaning, in computational technology terms, the most advanced) inexpensive DACs out there right now all vary way beyond the capacity of the human ear. Which means that buying all those resistors is the equivalent of buying a hand-built computer tower from 10 years ago at exponentially higher prices (in the case of the eight year old DAVE, 20-30 times higher), just because the one sheet uses a lot of super cool sounding, atomic age terms that promise to lift those veils once and for all. Seriously-the idea of spending extra cash for a DDC to fuss with IIS, only to solve the problems with expensive Denafrips or Holo May rip offs, instead of just buying a far superior, far cheaper and better-measuring DS DAC with a USB input is to me absurd. The DAC in 2022 should be the least expensive component in an audiophile setup. See, although they sound really wicked cool and esoteric, we don’t need fancy phase lock loop systems in this modern era because the processes they perform don’t change the outcome, if the goal is listening to music, instead of measuring computational ephemera. Progress, in the form of AKM and ESS, has solved all these issues for us. The “jitter” you’re wringing your hands over is indeed inaudible-even for “expert ears” such as yours, it’s true. Your thesis here sounds erudite enough, but to me it amounts to an orgy of misapplying analog audio terms to ones and zeros. I truly admire your efforts. But I feel an obligation to advocate for the dignity of others’ hard-earned cash. I’m sorry for using this tired cliche, but this vid is simply peddling a bevy of unnecessary snake oil. And please, do take note of the contradiction in your headline for this vid-one could argue that it’s a bit disingenuous to bait clicks by signaling better sound quality to be had, and then disclaiming three minutes in that the discussion is not about what is “audible”. Please go back to sticking to what’s audible. The wallets of all your followers will be so much better served if you do. After all, the money they’ll save from not spending four to five figures on a futzy DAC and a DDC to solve its connector problems can be set aside for their headphones-where the quality of construction and design truly has been demonstrably proven to make a difference. Peace…
-120db noise from jitter is NOT EXTEREMELY audible at all!!, in fact your amp background hiss more audible more snake oil audiophoolery, would like you see your "golden ears" detect jitter in a blind test where you really have no clue which is which
Just a question - "...the Schiit EITR is a very affordable device..." I assume you are aware that this product was discontinued about a year ago and is only available second hand? (where it commands a huge premium over the original MSRP). FWIW, the EITR was based on Schiit's older USB implementation (called Gen5) and the EITR was discontinued when Unison was introduced. Also, I agree that the Pi2AES is outstanding... I have 3 of them.
Eitrs sometimes pop up in Schitt's b-stock page for $99 plus shipping. Schiit discontinued Wyrd as well, so I guess you have a point unless someone wants to use spdif coax connection for older DACs like me.
I thought by now any reasonable DAC would have it's own internal elastic buffer and just internally re-clock using it's own precision clocks. Paul Lesso wrote an AES paper about S/PDIF design 15 years ago called "A High Performance S/PDIF Receiver". A small buffer chip should be super cheap these days, and should be able to wipe out any jitter from a PC optical port. I get the impression audiophiles like to make simple problems hard.
@@sanjacobs6261 Delay is bad if you are watching a movie or trying to mix tunes as a DJ, cuz the audio will arrive too late. But if you only listen, it doesnt matter if the audio comes out a second later then it got send.
@@gayusschwulius8490 For reclocking you usually use atleast 2 flipflops. This means that it's atleast 2 cycles of a clock worth of delay. The clock in this case has the frequency of the bitclock which is usually 44.1k*16bit*2 = 1411200bit/s. Which corresponds to 0.7microseconds of delay PER flipflop. I'd say delay typically becomes audible in live content at about 15-25ms. You could legit use a thousand flipflops and still not hear the delay.
For us noobs, might I suggest you spare a few moments to explain you graphs better and how one is better versus the other? I kind of think I understood so thanks for the great effort and video!
Soooo I'm somewhat new to audio. My understanding of those graphs is the more lines you see randomly sticking out from a pack, the worse it is. At 13:50 you can see a line sticking up on the left graph towards the left of that graph, then just before the big spike, theres a bunch of bumpy lines sticking up from most of the other lines. Neither of those are in the right graph. Those things are the noise/jitter. (I COULD BE WRONG.)
I'm not into audio at all. Just came here from my recommendations. But I have done some electrical engineering. The plot has frequency on the x axis and loudness on the y axis. He's sending a 12kHz sounds wave across the line which causes the giant spike at 12k. Since it's supposed to be a pure 12k signal, everywhere else on the plot should technically be at negative infinity (no sound at all). A lot of the reason it isn't at negative infinity is because of random electrical noise or jitter. Jitter causes other frequencies to spike slightly because changing the timing between points in the wave changes the slope of the sound wave and thus the apparent frequency of that wave.
@@wabbasMEpern I agree,...unfortunately. I have been in a rock band and have listened to live performances of all kinds of music from every genre! That's why I listen to "primarily" Redbook CD's (Yup MP3 duhhhhh) through an OPPO BDP105D (which have highly regarded Sabre DAC chips{even by today's standard}) connected by balanced cables to a Violectric V281 HP, which is connected by Periapt balanced cable to CHiFiMan Arya V2 headphones! It's all about the chain folks,....the fewer and shorter the links, the less are the kinks! With no insult to the most highly underrated band extant! Ahem!
I think these problems should have been solved in the DAC. The whole reason I am considering buying an external DAC is to get rid of electrical noise. Cue the "YOU HAD ONE JOB!" meme.
Considering that analog LPs are still considered to be the ultimate experience by many high-end enthusiasts, a 60db noise floor is not a problem and a 110db one even less.
@@dingdong2103 you’re misunderstanding the problem. The issue isn’t noise floor: it’s electronic noise and jitter which are a *completely* different thing.
@@GodfreyMann What is 'electronic noise' if not the thing that rises the noise floor? And jitter is not a practical problem in reality, it's just marketing jargon: hifigo.com/blogs/guide/what-is-digital-jitter-and-how-to-avoid-jitter-dac-101-part-3
@@dingdong2103 you’re correct that ‘electronic noise’ on the analogue side rises the noise floor, but the discussion is about ‘electronic noise’ on the digital side which gives rise to jitter. It’s not a marketing jargon - the clocks are voltage sensitive components and ‘electronic noise’ can cause fluctuations in voltages in the circuit. Although tiny this can affect the quality of the square wave signal and therefore how the bits are interpreted....it can change the bits and timing.
I thought the MQA video was good then I watched this one! Bloody brilliant, science based explanation of how electrical interference has nothing to do with snake oil. As a direct result of this vid bought a usb galvanic isolator (the intona 7054), 2 decent usb cables (also sorted power supply end out too) and my god my setup has been improved to a new level! I will for sure sort patreon support! Great work
Audiophiles: Measurements don't matter, all that matters is what you hear!! Also audiophiles: This device has a noise floor at -144 dB vs that other one that has it at -120 dB and costs 1/10, we know that it is a proven fact no human being can perceive the difference but just look at these GRAPHS!!!
In fact, any unwanted noise at -144db can still affect the sound quality when such frequencies are played, its still a delta to the "perfect" soundwave.
My thoughts exactly. There's no way in hell a human can hear a signal that is 120 dB lower than any other simultaneous sound (that is a millionth of a millionth of the energy in the main signal), even in the ideal condition when the main signal is a pure tone that does not overlap in frequency with the jitter-generated harmonics. Maybe a super low noise no-jitter-at-all DAC is useful in lab environments, but as a music consumer...
@6:35 an oven controlled VCO is exactly what it says - it is temperature controlled to optimise clock waveform & jitter performance - it is NOT an “temperature controlled” crystal, that is voltage is still used to set the crystal into its designed resonant frequency, but temperature controlled conditions, stabilise & optimise the oscillator performance😀🤓
Dear Goldensound, In this world where pure gold and sneak oil (EDIT : « snake ») are sometimes so difficult to tell apart, your reviews are really in a league on their own, thanks to your deep understanding, objective measurements, and explanation skills. Very helpful. Kudos ! And thank you so much.
6:40 for what it's worth an OCXO is still voltage-tuned (EFC) with a typical pulling range of something like 1-2e-7. The ovenized part is for short-term stability, not frequency adjustment. Essentially, quartz crystals have a characteristic curve of frequency change vs temperature change over temperature, which is very different for different "cuts" (the literal way the crystal is cut and polished). The standard is the AT cut, which is pretty much all quartz crystals you see everywhere. The AT cut crosses 0 Delta-Hz/Delta-K at around 25 °C. Which is just about perfect for normal ambient room temperatures. But the slope around that point is steep (relatively speaking, so let's say you're at 20 °C, you might be getting 5 ppm/K or so). Now the clever trick about OCXOs is that they use different cuts, often the SC cut. The SC cut has a completely different response to temperature changes; it crosses 0 at around 90 °C or so, and has a very gentle slope around this area. So OCXOs keep their crystal heated to a very _stable_ 85-95 °C, using a control loop. The attenuation of ambient temperature changes by the control loop colludes with the shallow slope of the dHz/dT to give extremely low tempcos. The _expense_ of that is: SC cut crystals are much more expensive (tiny tiny tiny fraction of the market), OCXOs are a lot of hassle to build and even fast-stabilizing types will exhibit a pretty wild swing from turning on to becoming mildly stable (at least 15+ minutes), they have very high power consumption compared to normal oscillators (couple Watts steady-state vs. a few mW for a normal oscillator and maybe 50-100 mW for something fancy. (And some OCXOs tend to fail under repeated power cycling, like you'd do in a residential setting -- they're _absolutely not_ designed for that and _will not_ hold their specs when treated like that). Most (all?) OCXOs have their own reference voltage which is typically ovenized as well. Sort of like bootstrapping, the reference has a tempco (selected for ~0 tempco at the elevated operating temperature, another reason these are expensive), so it would impact the temperature controller, so it's ovenized itself. It's of course stable, the tempcos of precision references are tiny to begin with. Now the reason people went to this trouble is that for, mostly physical experiments, extremely accurate and stable clocks are a useful tool. A good OCXO achieves short-term stability better than 1e-12, that's one part per billion, or for the typical 10 MHz reference, that's less than 0.00001 Hz variation over a few seconds. (Jitter and stability are not the same thing). These are mechanically vibrating devices: Vibration creates low-level spurs. Most physical phenomena influence oscillator output, temperature is just a big one. Gravity is another; angling or rotating a quartz changes the frequency (much more than 1e-12). So for optimal performance you'd want to dampen vibrations of all frequencies. Sounds like something you could sell to audiophiles. Big fat granite rock on a bunch of thick steel rope springs. Obviously, as you allude to at the end, the oscillator can't actually be used as a reference for anything (what's 10 MHz / 44.1 or 48 kHz?), so you're looking at a fractional-N PLL anyway, and at that point the quality of the output clock is liable to be swamped by the PLL's qualities (or lack thereof). To be frank, a normal quartz is probably just dandy for audio, if the rest of the system design holds up. Audiophility seems to be especially liable to buzzword engineering instead of system-level engineering.
Well, I'll find out soon. I'm waiting for my Pi2AES Pro Shield to arrive. I'll tell you this much, the Topping DX7 Pro benefits from an op-amp upgrade. I installed a full set of SparkOS units, takes the unit up a few notches.
And the verdict is... We have a winner! I use my DX7Pro as a DAC-Pre straight into my amp. So my take is the Pi2AES on Rpi4 running Ropieee via I2S elevates the sound quality by a decent margin. Overall the sound is even more coherent, soundstage width and depth is wider and deeper, the sound has a bit more body to it if that makes any sense, bass is more detailed, everything seems a bit more detailed, you hear more into the finer details of the music, transients and such, better decay and layering of it in more complex passages. I'd be saying these things with the Pi2AES regarding the previous upgrade, but this brings out even more. I wasn't expecting that at all because I thought I was hitting limit with what I already had. Not bad for $350 (I bought a FLiRC dongle as well).
Some DDCs will add delay yes. This isn't something I have tested for this video, but some DDCs will introduce a notable delay (particularly network streamers), and some will have none. Worth checking reviews/feedback on the DDC you're looking at. Some dacs also have this inherently. The denafrips dacs for example use very large buffers and this gives them a notable delay even without an external DDC
I use a GAIA DDC into my Chord Qutest. With 48k/24bit audio from my AppleTV, I've got no lip sync issues at all as opposed to running straight optical out from my TV. Latency will increase with sample rate, btw.
I do not often see videos which are pure and utter nonsense. But this one is. The spdif and the usb interfaces have error detection integrated. They will not deliver an invalid data stream to the DAC. Noise on a data line can cause problems but usually you recognize that by drops in the audio presentation or by connectivity rejection (no sound - full stop) and not by noise on the analogue side. Since the cable length and quality is good these days, you usually have however not problems or drops.
It sounds like some dacs would benefit a lot from a Pi2AES, where as others (the more expensive one's presumably) would benefit less if at all. I'd love to hear your recommendations for a good bang for buck combo with the Pi2AES if you end up doing a dedicated video on that device.
I do really appreciate what you do, especially the MQA stuff :) But. All this might make a measurable difference yes, but you do exaggerate the actual difference A LOT. "This is just awful levels of jitter" and you show a graph with spikes at -125dB.. Sure you say that we should make up our own minds etc, but when you show a graph with quite low levels of jitter than no normal person would EVER hear than you really shouldn't say that it's awful. Awful from a measurement equipments perspective maaybe, but not from a human perspective. This not a professional of explaining things even in a objective way. And you might be right that there needs more studies on jitter, but personally I've been into audio all my life and have done a few jitter tests (Archimagos among others) and the threshold that I hear jitter is very much higher than what even quite bad stuff nowadays measures. The general consensus from people that know audio (which excludes people like Darko and Paul McGowan) is that jitter is NOT a problem. So saying that something that measures this "bad" is awful only keeps that jitter myth alive, so please tone it down :) But yeah sure, I would still gladly appreciate more proper studies on this!
I'd like to, but unfortunately MQA has quite deliberately made it VERY difficult to obtain any equivalent files to test. And no MQA full-unfold device is allowed to have digital output. The only actual comparison i've seen where there was an MQA and non-MQA file that were the same was done by stereophile, and it showed all sorts of problems.
@Fernando Martinez It obviously depends on the type of signal being played. However, ignoring most psychoacoustic research, any metric exceeding 120dB (-120 dBFs depending on the measurement being made) is provably inaudible and that there isn't much room for argument.
Really interested to hear your thoughts on the May Holo. I ended up going with an Yggy GS for only like a 1/3 or so the price, along with a Pi2AES. I still wonder how it'd compare to the Holo. I guess one perk of the Yggy is that it's upgradable, and Schiit is already working on a new analog board for it. Looking forward to your review!
When you nerd out on stuff like this and the very technical reason who one thing is better than another, you ever... AB test your assumptions? My assumption that 24/192 FLACs being better than 320 mp3s seemed obvious, and even perceivable, until I did AB testing, and couldn't actually tell which is which. I'm genuinely concerned all this shit is borderline $800 silver cable territory.
a phase-locked loop is literally necessary to keep multiple signals in sync, I don't think I understand how something is supposed to properly interpret a digital signal it's receiving without one, the clocks would keep going in and out of sync.
Has anyone found how to stream all of PC audio to one of these? I heard Zeos in his review mention it was possible but I havent seen how people do it. TIA
I've been able to try internal Bluetooth with my Pi4, the quality and delay are quite terrible however. I also tried streaming with "Stream What You Hear" over the network, and while this uncompressed the audio up to a cap of CD Quality, it's still very badly delayed. Neither option was acceptable for videos of any kind. There may be other Linux-focused answers that highly technical users may be able to sort out, like using the Pi as a USB device or setting up the pi as a different kind of Linux audio server, but for a normal install of Moode I found none of the evident options worked very well. I'm halfway interested in coding my own solution to this problem, because the absurd delay doesn't make much sense to me. I get less delay sending packets to another continent than I get streaming the sound mixer's PCM output to a local Pi4 streamer with "Stream What You Hear", and that really shouldn't be the case. I'm just concerned I might be duplicating existent or better work since there hasn't been a ton of fruitful discussion on this and the Linux community tends to have answers to everything prefabbed and ready.
At first I also thought usb wouldn't matter, but I noticed an audible noise coming from the apple dongle on my laptop, specially when the fan was running louder and a cheap usb isolator from alliexpress was all it took to fix it in my case.
You should check out the Cherry 130dB+ DAC DAC 3, which will be introduced this summer! The coax/optical SPDIF input is internally reclocked for jitter immunity. Several levels of power supply reconditioning. True balanced, DC coupled outputs, too (:
I appreciate that you've presented actual measurements, but this seems like a solution in search of a problem. These DDCs seem to be all about second guessing the internal design of the DAC you paid good money for. A high quality DAC should already have its own isolated power supply and isolate any electrical noise coming in from the USB. It should also already have a high quality clock which is the only place you need to worry about jitter. If you just use the USB input, there is no jitter from external sources so no need to spend more money to try and minimize it. Just buy a good DAC in the first place.
While it's true that you are essentially "secnd guessing the internal design of the DAC" and that the DAC already SHOULD have this implemented, it doesn't mean that it actually has. A good clock oscillator requires a good crystal (which is just expensive to manufacture/buy). Attenuating input jitter can only be done IF you have a proper local oscillator. You are making a compromise between how much the external clock matters and how much the local clock matters. For USB DAC's this is irrelevant because USB is asynchronous. For power supplies I completely agree though. You won't make the power supply much better than the way it's implementend internally.
@@hidjedewitje with asynchronous audio dacs you also have the option of putting in slave or master mode when using an external master clock - despite all the advantages of async usb, the search for cutting edge performance continues... Unless you're Bruno Putzeys, design a good one 10 yeara ago & its still cutting edge....
@@BogdanWeiss Bruno's stuff is indeed very fascinating. His work at Grimm is also quite fascinating! The AD1 (now discontinued) is still the best AD converter when it comes to linearity and jitter performance 20 years later! The question is, do we really need such performance?
@@hidjedewitje Performance just like intelligence is a double edged sword - once seen or heard, it's difficult to un-hear it - I didn't realiae that the grimm adc has been unchallenged for this long - truly impressive
@@BogdanWeiss While true, there's definitely diminishing returns. The difference between a laptop's on board audio to a standalone DAC with THD+n of -115dB is pretty audible. There are also DAC's with THD+n under -120dB. I doubt anyone can hear those differences, despite the performance being objectively better. There's also people who stare themselves blind on numbers regardless of looking at the weakest link in the chain. They buy DAC's of $1000, but leave the room untreated (resulting in dips of 30dB in the frequency response!!!). There sure is a place for state of the art products (like the Grimm's or the Mola Mola's), but they should also be placed in a state of the art environment to be actually usefull.
I appreciate the actual graphs that prove what you're saying. I'm new to high-end audio, and for the most part I assume that the things that people tell me, like "this is better than this because this one costs 10x as much and it makes me feel good" is mostly subjective, but you've supplied receipts which actually puts weight behind your words.
Also: NONE of these people commenting here, nor the uploader has ever heard jitter in their lives. They do not conduct proper blind testing but just conclude things. Measurements do not lie but most people don't understand the decibel scale and the number in it. If it is below -100dB: do not bother, you will never be able to detect that in any real life scenario, yo can only catch it with measurements. In most cases we should not care what happens below -70dB but it is better to be safe and aim for a bit better than Redbook, 96dB dynamic range.
Hi @GoldenSound! Great reviews, as always. But unfortunately looks like the Pi2AES project is no longer available and they have a closed and more expensive version. So, I was wondering, with the release of the new Topping M50 which is kind of a weak streamer, can it be used as a cheap USB to IIS converter? If so, could you measure that to see how clean the signal is? Thanks!!!
If at all possible, you should beg, borrow or steal an EtherREGEN along with a quality external master clock, such as an AfterDark Giesemann or a Mutec. I've added both to my Pi2AES along with quality LPSs. My listening experience has been transformed for the better. I've never heard such 3-dimensionality, fluidity, realistic dynamics, sweet resonant decays, etc.
Just a quick question... Would one be able to use a DDC with an audio interface? As far as I'm aware audio interfaces send and receive sound instead of just receiving it. Recordings are clean as anything but I get an awful lot of noise (especially CPU noise) at audible volume which ruins a nice album at the best of times. Curious if a DDC could fix that. Either way, excellent video : )
Like many of us i use my PC directly to my receiver amp ( Yamaha RX-A3070 with ESS pro chips ) via an HDMI cable connect to my graphic card.... HOW IS THE JITTER with this connection ?? Thank for the answer ( i hope )🙏
Great Video, I agree with using I2S, I was wondering if you have heard of IAN Canada. I found these HATS for the PI and I was thinking about using these two devices with a PI 4. 1. IAN CANADA FIFOPI Q3 ULTIMATE FIFO Reclocker Module PCM 32bit 768kHz DSD1024 DoP 2. IAN CANADA HDMIpi MKII Transmitter I2S / DSD / DoP to HDMI
Outstanding review. Man, I wish you had a Denafrips DAC using the external clock function with the DDC for this round of review. I just ordered the Terminator-Plus DAC with the Gaia DDC and Kinki Studio EX-P7 with EX-B7's.
This is such a misleading video. I like charts and data and all but commentary is just awful. Show -130db on the left, show -140db on the right - MUCH BETTER. Who are you kidding?
Since you are basically the only one to do this kind of things on RU-vid (and thanks a lot to do so!), I think you should take more time to explain what you present. This is some really really interesting and in-depth information, it deserves longer videos, or more videos. Something I don't understand: why should we need a device that clean the signal from the source to send it to the DAC if you can have that in the DAC itself? Shouldn't it be a requirement for any DAC to basically reclock what's coming from the USB and isolate from the voltage coming from the USB?
Wouldn't mind seeing a video talking about power input with regards to audio equipment: the effect clean power has on equipment, usefulness of power conditioners and UPS's, and what to actually do if mains power isn't reliable with frequent brown-outs and general inconsistencies.
Just some "personal" perspective from my experience as a software developer for real time embedded systems and deep interest in electronics. (not professional in electronic tho) I'll only talk about those three topic you start with, over usb, some applies to other protocol, but I don't know them in detail and I'm not going to read pages of specifications for a youtube comment. The TLDR is just to say, normally, you shouldn't have to care about those three issue in a "just enough" designed modern usb device, even more one who cares about those well known issues and tells customer they're some kind of high end devices, dac, or anything else. Data integrity Being digital doesn't mean you can't have data corruption, for instance if you communicate over serial without any specific protocol, or a protocol that doesn't integrate some form of data integrity mechanism, you can have data corruption and no fix for it. That being said, the usb protocol specs specify a mechanism for that. A CRC is sent to ensure the integrity of the received data and if the check failed, the data are re-transmited. (to make it shortà That means, for any device compliant with the usb specifications, it's not possible, on the transport, to corrupt data and not notice it and fix the situation, it's by design. So no need for extra middle man. So, being "bit perfect" and "not altering data", I don't understand the take. A usb hub does it, nothing special. But I don't understand what the point is, so I just mention, to anyone worried about this, usb, by itself, protect against data corruption on the line. If you worry about that, use usb, and you'll be fine, by design. Electrical noise You mention using optical link as solution to decouple two electronic circuit. That's way more on point than it seems and is in fact quite... "original', that any "high end" dac doesn't do it. The solution is called optocouplers, it's used everywhere, even in low end electronic, it's cheap, and it does just that perfectly. The fact you have to introduce middle man to solve that issue for expensive units is a shame. Also, optocouplers main interest, outside of the noise, is mostly security concerns, and it should be used in scenario with interconnection to any other unknown circuit. For instance, you certainly don't want to blow your expensive device by linking it to a faulty one that might send huge spikes to yours. If you design something like a dac, being proactive about those situations shouldn't even be questioned, even more for expensive equipment where you should do anything possible to protect it from the outside world at the benefit of the customer and your brand image. At the same time, solving noise propagation issue through data or ground lines. Imho, any dac, even low end, should do it. And to be honest, it's a solution found in a lot of studio equipment, and you don't have to go high end for it. So blame the dac manufacturer for not doing their job. Jitter It was a real issue for instance with spdif and the reason was... mostly a "bad" design we don't rely on anymore. Also there's different type of jitter with different sources and different solutions. The way you represent what jitter introduce as a problem doesn't match the general design of todays dac, dac aren't clocked by the input protocol clock anymore (and if they are, it's on their own and they decided to go the wrong way, maybe buy something else... who easily solve the problem) But, usb has a well known workaround. The main solution to solve almost all jitter issues is to use asynchronous mode. Jitter will be small to the point it's humanly not noticeable anymore and, if you really want to solve that picosecond problem, just introducing a buffer or related, audiophile dacs, imho, can do it, I wouldn't mind it a all as a client, but might be wrong. The fact this problem is still present really makes me wonder: are they using 20 years old design that just "get better over time" but don't do re-design that would solve the issues ? Once again, it's the dac manufacturer fault, it's a design issue, it's a known problem, with known or even integrated solutions, if problem persist, maybe go back to your R&D and fix it before requesting any substantial amount of money from your clients. Paying a middle man device to fix those three issues, at least over usb, until you are like using an old dac you really love, shouldn't even be a thing. Shouldn't even be mentioned on modern hardware, not even as a selling point. It's something people sometimes say about dac to say every dac is the same, I don't have opinion I want to share on that topic, but for those three issue, if you have a modern dac that features them, I would consider them "by design broken dac", any mention of those issue on something I plan to buy would make me totally never ever go toward that manufacturer again. To me, those issue, on new hardware, shows a total lack of dedications and I really hope I get this video wrong and it's about fixing old gear issues. At least for usb, don't know for the rest, but wouldn't be surprised there's solutions too and modern hardware shouldn't rely on middle man.
I think that this is for someone not using a steamer. Instead of using a Hermes at a $1k us, I would have used the Iris at $499 US. You can find them under $400 US. It will only do USB to I2S and does a great job of doing it. For the other digital connections, I have an external 10m clock that works wonders with my DAC.
I'm considering buying a sub $1000 DDC and have 4 candidates, but am open to other suggestions. Candidates are 1) Gustard U18 2) Singxer SU-6 3) Denafrips Iris 4) Topping U90 In stock form, which puts out the cleanest IIS signal?
All of them sound exactly the same, but I went with a Topping d90se. I use its USB input, because it is the most convenient option and handles any sampling rate. There’s no benefit to IIS, despite the marketing mistruths perpetuated by PS Audio’s Paul McGowan-who promotes that tech because it was required under his clandestine agreement with Sony to circumvent their DRM rights for outputting an unadulterated digital signal from SACD players to his own proprietary DACs. Don’t fall for this ruse, it’s an unjustified expense that has no impact on “sound quality”. Digital audio is not sound, and no matter what transport you use, the same signal is passed to the DAC and the resultant outcome is indistinguishable to the ear.
Golden sound I'm sure you have explained more times than you care to remember so please from an American fan, what are those card or pictures on the wall I've been wanting to ask for a year or so. Thanks for the videos!
Wouldn’t ferrite beads make DDC’s redundant in the case that someone is only using USB outputs from their original source? Edit - Assuming that the source of EMI isn’t some massively noisy fucker that can’t be handled without special boxes idk
Interesting video but a nightmare when viewed on a large screen. After a few minuets I felt quite travel sick! All that camera swinging back and forth. Please get a tripod.
I use a Topping D10 as DDC for converting USB from a Mac to optical. Then I go via optical into the MScaler and then TT2 via dual BNC. My impression is that this sounds much better than going directly via USB from the Mac into the Mscaler. Now, you said optical from a PC is the worst one can do. Am I wrong about what I'm hearing or what is happening here?
Would SATA & PCIE have more or less jitter noise than USB? I'm currently running 4x ASUS STXII's through a PCIE 4x riser kit, i'm also using a USB CD ROM but I'm considering a SATA one and shutting down the USB Ports...
Wow, really enjoying this channel and the clear technical explanations. I'm glad to find out that my DAC has PLL for all SPDIF inputs, I was about to go researching a Pi2 AES!
Waste of time and money, use async USB and that's it... And unless you have a seriously broken DAC, the DAC will perform up to specs and jitter is just noise... well below human hearing noise floor...
This entire industry is one giant marketing scam. Your $2000 DAC doesn't have a good enough clock! Gotta spend another grand for about $20 worth of off the shelf electronics for that chum...p
Am I correct in thinking the Pi2AES is only worth having if one has an I2S input on ones DAC? Or can you also perceive a difference over TOSLink? I’m wondering whether or not it might worth using over TOSLink with my ADI-2 Pro FS (instead of a direct USB 2.0 connection from my iMac)?
I've got a Pi2AES on the way, out of curiosity I was going to try the I2S to my Gustard X26 Pro, I was wondering if you could please tell me what cable you were using? when I googled a lot of the recommended cables were more expensive than my actual gear.
Can't the DAC store a few samples of information and convert to analogue by itself whenever required, without asking the source/pc for every sample? So that even if there is Jitter in collecting the data from pc, the DAC won't experience any Jitter because samples are readily available in Buffer. I want to know your opinion or answer for my query.
How did you connect your Hermes to your May via I2S? I have a Gaia and a May linked up by AES/EBU because none of the I2S pinouts on the May seem to match Denafrips' implementation.
Why should the DAC ask the source for each and every sample? Can't the source send some number of samples and let the DAC store them for sometime before converting to analog waveform.
Conclusion: more streamers should have I2S output. And more DACs I2S input. But it seems the interface for I2S is NOT standardized, which arises new problems.
Great video. But the giant question it leaves unanswered is whether the little no-ISB Pi device that beats the Hermes is uniquely good at destroying jitter, or whether *any* transport without USB (such as my MiniDSP SHD Studio) would yield similar benefits, simply from avoiding USB.
Considering he already did the number crunching, I should hope they wouldn't have much to screech about - though I know that's probably wishful thinking. But so long as it just means more data, and not just ridiculing, then all is well and good.
Hi I have purchased the Pi2aes , but with 24v switching power supply the sound is awful connected to my Denafrips Pontus 2 I2s port with high end hdmi cable using roon raat to Pi2aes., also tried BNC out with my £800 Chord Signature Super Aray digital bnc\bnc. Compared to my mac mini M1 with a £300 curious evolved usb lead connected to Pontus 2, it was not worth the purchase of Pi2aes as it is. You say you will be recording a follow-up video to build the Pi2aes and install your 5v power supply. What power supply did you use to power the Pi2aes via the 5v pins? I have managed to find out how you connected to the internal 5v pins via another source. Regards
I had the exact same experience but I am using the Allo USBridge Signature instead of an RPi4. I was thinking of getting a linear power supply like the Shanti but was warned about making modifications to the Pi2AES and powering it via a linear power supply. Anyone have any pointers as to how one can power both the PI2AES and USBridge using a Linear supply like the Shanti?
