Hello Hans, as always a great contribution from you! I myself have been dealing with the topic "Ethernet sound" for a long time and am also in contact with Alex from Uptone and have also exchanged with Jaap about his first measurement. I have my own blog on which I have shared various measurements. All this is not because I want to discredit audiophiles, but I want to understand what is really going on. My findings in a nut shell: Most of the sound change is caused by common mode noise in the Ethernet signal. This includes the 50Hz leakage currents John talks about in his paper. But also RF noise that enters the ground system in the receivers and then becomes noticeable as in Jaap's measurements, or even takes effect during the conversion itself. The muon filter inserted in Jaap's measurements, I believe, is nothing more than strong common mode chokes reducing common mode noise.
Why would the network matter if the whole song could be transferred and buffered on the player within a second of starting playing. Then there isn’t any additional traffic and the song is playing from internal memory. As far as Ethernet, the player could just optically isolate the port so there is no electrical interference to filter out.
@@TheHansBeekhuyzenChannel yes, they do, otherwise you wouldn't hear less clean audio, you'd be getting an awfully gltiched mess that's basically unplayable. I'm wondering why nobody that claims switches make a difference will provide a simply test of packets integrity with Wireshark
Because the packets do arrive perfectly. I have explained in several videos it’s the noise and distortion of the analog signal that is used to transport the digital signal that cause minute timing errors on the clock input of the digital to analog converter.
But the ones and zeroes that are playing from the separately clocked buffer memory don't care if the transmission to it is riddled with timing error. 10010110 transmitted with timing error is the same as 10010110 transmitted without timing errors because the buffer memory uses its own clock.
We have a LOT of measurement training but essentially ZERO listening training. If we want the world of audio to improve, we need to learn to LISTEN. That will not happen by DIY.
Just to note, there are a couple of standards that can improve jitter and/or packet loss on ethernet. They are AVB ("audio video bridging") and DCB ("data center bridging"). In both cases, they require hardware that supports the standard, and is not something you can do on a generic ethernet switch.
There are so many companies who knowingly make products which are stupidly expensive that make no difference. Less technically inclined people will not know, buy it, and because of confirmation bias think it works. That money could've been spent on better speaker/amps (that make a huge difference) or a nice dinner with the family.
@@TheHansBeekhuyzenChanneljitter doesn't matter, on the receiving end it gets compensated. Was the listening done blindly? I often did a blind test (or even lied to have some confirmation bias) between Spotify over Bluetooth (yes) and a wired Qobuz stream. Either blind, or told people they would now hear Qobuz whilst really it was Spotify. And every single time confirmation bias won and they'd describe it as more natural (favorite term) or more open. Needs to be blind, if you can still tell the difference multiple times I'd agree.
I will most likely never own a Muon, as hearing the difference is dependent on the variables you mention, some of which I cannot control. I have just spent $800, mostly on bass traps and some diffusion. I instantly heard a difference when I quickly positioned the just arrived traps in my room, I am now in the longer process of dialing in placement of the traps and calibration of my Genelec speaker system, but it's clear to me that it was money well spent. Although you did say that better upstream clocking is a component of achieving the "natural" sound we pursue, the distortion off the the analogue signal that carries the digital information you also discussed had a larger resonance in my mind. I also read somewhere in the Uptone literature that using a more accurate clock than the one already in the EtherRegen had not much effect. I managed to buy a used ER and a quality used ethernet cable for ~$650 earlier in creating the endgame set up I have now, which includes a Wattson Madison, largely because of your reviews. Thank you for all you do for us Hans!
Thank you for another fantastic and informative video, Hans. It never ceases to amaze me how many naysayers and detractors we have out there who choose to embrace their (limiting) instruments that spit out data that’s quintessentially useless, RATHER than choose to rely on their ears. But the concepts you explain here in detail explain so much, so thanks again for your dedication to our education. And BTW, I’m a huge fan of Alex and EtherREGEN. 😊
Hans you still are not understanding how digital audio transport over a network works : as you said, there's buffering on both ends of the IP transport part, so IP/Ethernet jitter measurement is totally irrevelant. Those bits are not directly transfered to the DAC chip, but debuffered by the DAC itself at its won pace (thats is, with its own internal clock)
I am no network specialist. But when it comes to audio, I know a thing or two. My guess is, more than you. I am learning about the influence bit transport has on digital audio. But regardless whether all bits arrive in tact - and they do -- there are influences on the sound quality and what I described in this and other videos were base on the best explanations of digital audio experts. Denying what's audible doesn't bring us further.
How much do you see cable length or number of hops affecting the outcome? Does more ever equal better sound? What's makes these devices so expensive - is it R&D?
Thank you for taking the time and effort to investigate this matter and then make it clear to us. Really well done! Yet, the MUON filter better measurements are not exactly spectacular. Does this mean there may be even more to be achieved with signal filtering? How much even lower can we go with jitter and phase noise?
The measurement was relative. The difference in sound quality was drastically better. Measuring phase noise depends on many factors, like power supply, cables used, EMF and so on.
Good day Hans, I have always loved your videos. The information and moreover the background work you put into same is to be admired. Being an engineer I have struggled with many aspects of Audiophile accessory sales, I will add at this point that my ears are not perfect and do not pick up everything and that my experience with electronics is not in the field of audio. However having worked with very complex analogue and digital distributed control systems I struggle with many claims. For you to release this video is admirable, thank you for restoring my belief in my own training and work history. Keep up the great work you do, I will always be watching. Stay well.
If data integrity is maintained by every switch and streamer interprets data, how it could affect the sound? Does road quality determines quality of the goods transported on the track? No, until integrity of the goods is not affected. Is there any filter or conditioner for the wifi? Can you hear the difference between files played from different hard disk drives or memory cards?
@@TheHansBeekhuyzenChannel sorry, but a digital signal can't be "noisy". It's 0 and 1 (not quantum bits...) and if they are corrupted, it can be detected (and sometimes corrected) by error correction codes.
@@eya83fr I think that's the disconnect, although I might not understand. Even light takes time to move. I think he is looking at the tiny nearly immeasurably differences in the signal and believes they affect affect the sound. Like maybe a 1 or 0 has a slightly higher or lower level, shape, or length. I think it's similar to why it's difficult for hardware midi to sync than expected. But I would think here it would either break the signal or add a tiny bit of latency to the whole stream. I can't imagine a halfway decent filter or converter having an audible difference from this though, but I'm not entirely surprised if you measure it with enough detail there would be a difference in the transmitted signal itself
Thank you very much for sharing your insights gained on this issue. I especially like to hear a man say, “Well, I was wrong.” I know there is good stuff coming. I too have been around networking since 1985 or so, but I still don’t understand it all that much. Just getting into its use in audio. Your video was most interesting, I’m now in the market for a passive filter. :-)
Good Hans, so I’m using WiFi when streaming with my dmp-a6me with lps, I’m running a usb from dmp-a6 into denafrips iris DDC and toslink from iris into sonnet morpheus Dac. Will I improve my sound by running a Ethernet with a network switch?
Hi, I have a technical question that may be very basic. In a digital signal, each bit has a value of either zero or one, which is represented by a specific voltage in the signal. Does this mean that a 16 bit recording must measure voltage 16 times in order to have a complete representation of the binary number that represents the sample?. How does this relate to sample rate? A sample rate of 192khz, for example, does it measure voltage 192000 times? and every "X" bits represent a full sample? or does it measure 192000 samples, each one with several voltage measures to build the full sample
The signal is measured 192 thousand times and the measured voltage, 1 volt, is then converted into a binary code using 16 bits. The binary code is just another way of writing that 1 volt.
@@TheHansBeekhuyzenChannel thanks for your response! But then the representation that you make in your videos is not fully accurate? In this case the signal may have as many possible voltages as 2^bit depth (so that the DAC translates a very specific voltage to a binary number)
@@dfrancoanzola You appear to have it the wrong way round, The ADC (analogue to digital converter) measures the signal and creates a binary number according to the input amplitude (and polarity, positive or negative) of the incoming signal at the instant the signal is sampled. That's done in the recording processes. At playback time the DAC takes that binary number and creates an analogue voltage according to the binary value it receives as input.
Hi, Hans. I am using an Ediscreation Silent Switch OCXO and an Ediscreation Fiber Box II. The change both brought to my system is incredible. Both have OCXO clocks for the ethernet digital signal and the Fiber Box II internally converts from wired to optical and back again, thus galvanically isolating the signal from the modem supplied by the internet service company to the switch. In addition, I have a filter from PS Audio that is placed at the entrance of the cable that comes from the street into the aforementioned modem.
There seems to me to be one big question here: how is a passive device able to improve phase noise? Unless I completely misunderstand the definition of phase noise, it's essentially a measurement of the deviation from perfection of a periodic signal. How can a passive device, such as the Muon do this? This video article's title seems to me a little unfair and misleading. Fine, it's a fair and balanced review of the Muon, with an added technical assessment. But I see no equivalent review or analysis of any network switch. Will we be seeing an equivalent suite of measurements of the EtherREGEN? Myself, I am the grateful owner of an EtherREGEN, and I would state that it had as big an impact on my system as the Innuos PhoenixUSB did. Why the digital signal needs regenerating before AND after my Zenith streamer has long been a puzzle to me, but one I happily accept. Returning to the topic of phase noise, I also added an AfterDark OCXO clock to the EtherREGEN, plus an optical stage to precede it, plus decent LPSs to both clock and EtherREGEN. All, to some extent, have offered benefit.
Thank you Hans for all your insights on hifi. I was wondering if the Roon RAAT protocol, which they claim is 'bit perfect' makes a difference in streaming music vs for example steaming from Tidal direct?
It is easily tested using the Audio Precision behind me. And, indeed, it is bit perfect. But there is more to a digital audio signal than bit perfect. Watch ruclips.net/video/ZCFvIzzMqfk/видео.html
To my understanding audio (and video) streaming utilizes UDP. UDP packets beneifites from low latency, but there's no possibility of retransmits or rearranding of packets. They are sent in an ordered stream. So the quality of networking components could potentially have an impact on the stream, eg. packet loss. Please correct me if I'm wrong. BTW: I'm not a network engineer either.
@@TheHansBeekhuyzenChannel No, I have not. I do not experience packet loss in my simple setup at home, so I believe my equipment is of adequate standard. I only have a broadband router with WiFi and four Ethernet ports, One of the ports is connected to my NAS (wired short distance). Streaming endpoints are connected via WiFi (Marantz SACD 30n, Mobile phones, tablets), A WiFi extender helps were the signal otherwise would be weak. I just wanted to point out that your statement in the video regarding retransmission and reordering of packets does not apply to UDP connections/packets - which is the protocol used for media streaming (if I'm correct). The UDP protocol only has a checksum. If the checksum doesn't match the data (stream) at the endpoint, then the packet is discarded. This will be audible and/or visible of course.
It’s not the wifi technology as such but the way it is implemented, including the locally available bandwidth. In an apartment building that will be more difficult than on a ranch.
Not for the first time, I wonder whether having Golden Ears is a gift or a curse! Does Hans actually like music - he never mentions the music, only the technicalities.
There’s no such thing as golden ears (great title for a next James Bond movie), only an accurate hearing which needs to be trained. And you train it by listening to music 🎵
@@jmtennapel you listen and enjoy the music or you end up in the world of Audiophile and end up buying as an example SR vibrotrons, Audiophile fuses and sticky bits of metal you stick all over your speakers convincing yourself it has made a difference. Spend what you like but this hobby is about enjoying music. If you understand networks and data transmission you know how this works.
Last week I bought a Grimm MU1 this combining with the MM Tambaqui. Does it still make sense using the Etheregen which I already owned or can I remove this from my system?
Again, great info Hans! I was using an etherregen close to my SoTm sms 200 Ultra Neo streamer in my set up and wanted to improve the cleaning of my LAN (my router and NAS are downstairs- roughly 40 feet away). So I added a Ifi LAN ISilencer ($89 USD) at the output of my router (going to the EtherRegen). An instant big improvement… the biggest being with bass notes; the pitch is so much more perceivable, with more punch and body. The timbers are improved in such a more credible, natural way. So yes, clean up your stream and everything is going to sound closer to the “real thing”.
Hans, this is certainly a great video and I appreciate your work and competence a lot. But ultimately I didn’t really understand it - please forgive my ignorance as a formally trained electrical engineer, decades ago. Somehow rather prosaic questions aren’t answered. I’m sure you could do that with ease: a) what does this filter actually do? b) where and how would I need to place it in my stereo chain (between which gear exactly)? C) I have a setup with a relatively good DAC and active monitors in a decent room, BUT my digital signals comes from a NAS, via a Switch, via a WLAN rooter, via relatively weak signal (our house is rather large), through a WiiM pro streamer to the DAC ; isn’t in such poor setting a “filter” doing anything good? In short, do you have some practical guidance for practical settings? Cheers.
What this filter actually do? We son't know. Apart from reducing phase noise, of course. It is placed in between the streamer and the network, closest to the streamer. It doesn't seem a sound investment given the streamer you use.
Great work Hans but I can share that my high end audio system with monoblock amps benefited greatly with a Wiim Pro streamer. With a Verizon fiber router, a Cisco switch and fiber, the improvements are nothing less than excellent. I’ve heard the benefits are obtainable with Juniper and other commercial switches for far less than audiophile versions. Agreed the quality of the system will play a major role. Thanks for your strong contribution to music quality and understanding.
Picoseconds... I didn't see that one coming. Glorious work, Hans and Jaap! Visitors wonder (and smirk) at the laboratorium - like aproach audio seems to require at my place and probably yours, fellow audiophiles. But Hans is so right: It only takes secondes to hear. And there is usually no way back once you heard It. Oh well, still great times we live in with the equipment available. And the reviewers that help explain what we hear.
I got the iPower PSU to improve my Netgear switch (recommended by Jaap) but that seemed to make the sound very sharp and think sounding. Any idea why that could be the case? I had to switch back to the factory supplied PSU...
My guess would be that the iPower psu makes the switch more precise. That will make the DAC have more resolution leading to a sharper sound due to other problems in your setup. Like interconnects that use poor silver. But of course I can be totally wrong. The iFi psu really is good for that money and most likely better than the one that came with the switch
@@TheHansBeekhuyzenChannel Thanks Hans. I was wondering if the PSU having more Amperage (1.5A) vs the standard PSU (0.5A) could have anything to do with the observation. Then there's the central house router that feeds the switch via LAN... Wondering if just having one switch in the chain improve the pSU enough... thanks
I have tried the ifi PSU too (the X version), I believe on the same switch, and while I found it quite accurate, it sounded quite a bit too much of a good thing, so similar to your observation. And I would think it might have to do with the PSU’s bleeding of HF hash into the power line, not its DC output. You might try that: if you insert the psu and power some device that is not linked to your hifi gear you might notice a shift in SQ. Or not ;) - anyway I tried tried to eliminate as many switch mode power supplies as i can, the remaining ones get ferrite between socket and psu, to reduce HF hash. And it did improve things in terms of treble smoothness. now I use a linear Psu for the netgear switch, an sbooster, with the ultra device and - an ifi dc ipurifier at its output (yes that is crazy) but its better ;) possibly overkill, but I found it an important improvement wrt prat and dynamics. Not sure what is going on there. Audiophonics too have an ok lin psu for
The Muon sounds like a good product but I would prefer a more cost effective filter. The Muon is about $3,000 here, something almost as effective for half that price would find a place in my system.
@@TheHansBeekhuyzenChannel I guess cdp introduces problems of their own. Apart from their inherent limitations of music choice. You have made me interested to try out the Muon filter with the Pachanko server I run. Thanks.
Well if Network acoustics invented and patented this improving device they should be doing very well. If they have not protected it there is no stopping others from getting one and breaking it opening and seeing what is in there and if they choose include it on their streamer ethernet port. Maybe expensive streamers like Grimm already has it?
Hans my point has always been that with local rendering with the music source @ device.( as a baseline for operation) All those readings can also be influenced by .02 volts I've built Linear Power supplies with 8 isolated outputs including my network switch. ( ( preventing device to device noise is critical)) in the power chain
You can clean SMPS to "HIFI GRADE" with over the counter parts then add ferrite to each cord to reduce EMI. Preventing noise propagation is the actual goal
Hello, of course these test executed by people owning commercial businesses are very easy considered a marketing tool. The bigger the number of devices they can sell as being " audiophile grade " the more money they will make. It seems that the Grimm devices are kind of immune to the garbage that comes along with the signal delivered by your internet provider. EtherRegen with a audiophile quality power supply, some nice feet, aftermarket power cable, ethernet cable and you are getting close to 2000 € additional cost to a device that is not exactly cheap. Sometimes this audio hobby is getting boring. I already decided that after receiving the Grimm i will stop investing in digital gear. Greetings,Eddy
@@TheHansBeekhuyzenChannel Hello Hans, i already have the Etherregen with an improved power supply but probably it won't improve anything so it is good to know! In your mu2 thread you already mentioned that very probably i won't be needed. Which makes things easier. Reducing the number of power cables and " interconnecting cables" will save me some money and should bring some improvements. But on some " marketing site" USA based i already see people asking about upgrading the fuse. I already have a bamboo chopping block so i will try that one. There is info on the internet about diy vibration killing feet using copper sponge as the key element!! Greetings,Eddy
The internet and associated online services work perfectly well due to Transmission Control Protocol and Internet Protocols.; Ethernet itself is differentially transmitted, etc. Concentrate on your internal system clocks.
1- as an IT hardware professional, i know for sure that switches leak data. Most likely due to not switching fast enough. I'm assuming the opposite should be true here as well, ie. they block out bits that should have been let through. Now computers are designed to handle this without breaking a sweat. But do you think this negatively affects audio? 2- Is this device acting as a buffer (a storage buffer type) between the switch and the dac input?
This is the first time I hear this from an IT professional. That will impact isochonous connections. But should not happen with asynchronous connections I would say.
For a home network this should be extremely rare unless you have very very bad setup. This would show up as dropped packets and can be tracked. If you are getting dropped packets at all on a wired connection most likely you have some defective equipment.
@@a_macaulaygood point. It's a bug but was it a result of hardware going faulty or just the way it does it's switching/processing or a combination of factors, I don't know. This was me doing some testing on my home network. Specifically, the ISP provided switch (no not a router and connected before the modem). What I found was it was randomly letting through 5-8 bits when it should be blocking that data (due to a different ip configuration, very complicated...). This is what led me to wonder if the reverse is true as well, ie. blocking data that should be let through, say for example the first 5-8 bits of a stream. Now, when one is referring to computers or servers etc. the TCP/IP protocol is the first layer that is designed to easily rectify such lost bits. So it doesn't always amount to packet loss. In fact, the errors have to be seriously bad for the TCP checksum correction to fail and result in a packet loss. Also, the TCP protocol (or any other protocol ftm) can ask for retransmission of the lost packet in case of uncorrectable errors. This again, the packet is not fully lost, ie. it's recovered. Packet loss would only result if the data can not be retransmitted, due to things like being flushed out of memory or not being in memory in the first place. That's as far as computers are concerned. Music is a whole different topic. Here, bits are not (just) bits. It is time sensitive, and extremely so. Hans should be aware of various DAC configs in this regard and resulting effects. Also, audio gear is nowhere nearly capable of possessing as much processing power. Also, regarding the retransmission request of lost data packets - the USB 3 and above standards can also request retransmission in case of failed checksum for data integrity, just like TCP/IP. I have no idea which audio gears actually utilise this. Retransmission request and receive times on a home network, or a USB 3 connect between a host and recipient, definitely would be in picoseconds, even if at the higher end of the ps scale in the worst case scenario. So request and receipt completed even before the entire data transfer completes. All this raises a further question - the time factor. I had gone down this rabbit hole long ago, and it had looked very interesting to find what was there, but I guess it was too far down ...
@@TheHansBeekhuyzenChannelah yes. I remember the asynchronous & synchronous part now. Also, please do refer my last comment here. Just leaving it here for you, in the hopes that you are the best person to be able to use this, if it fits, and maybe dig a little deeper for us. 👍
No proof needed as long as someone hears it or thinks they hear it and whose to know, as anyone who doesn't obviously has crap audio equipment or ears. Thank god theirs always a fix, even if it changes week to week etc, by some expensive box or five and magical cable. Its a wonder the rest of humanity can bare to listen to music at all.
keep up the good work! people need to know this info as digital glare and time smearing can totally ruin the music. Thank you for your clear and considered commentary. a great digital front end allows me to listen to a wider variety of music that previously was quite tiring or simply un-engaging.
If I undrstand correctly non of the network players work correctly except just one, because digital signal shouldn't influence the analog output, end of the conversation. Basically the problem same as in digital TV if there is distorted or no signal I got a green frame or block on the screen, unlike analog TV where I get fuzzy or distorted picture. Live TV Sport broadcast do this battle for years they rather bradcast 720p 60 FPS lower quality but faster information, than 1080p 30 FPS higher quality lower speed, because there is no enough bandwith for 1080p 60 FPS. If everything what you mention in the beginning of the video about digital audio is in order there souldn't be any problem. Digital data comes in, put it in a buffer and DAC converts the digital data into analog signal. If the buffer is big enough and the delay is long enough there will be no problem at all. For me this is clearly a digital processing error. What a car or airplane chip or my PC handles with ease a way overpriced audio gear couldn't handle. I bet this problem doesn't exist on low end gear, for example if you play 128kb/s 44.1kHz mp3 on a 16bit 44.1kHz capable device. As a technical person I highly agree with other technical people, if you can hear it you can measure it, simple as that. Record the audio put two plots on eachother there is your difference, we have the answer for the "What" question, and for the "Why" the equipment is faulty/false advertisement, doesn't know what the manufacturer states. A well designed device should never accept a jittered signal in the first place, so just like in digital TV you have a perfect image or non at all, if you have a bad cable you should hear nothing not distorted sound.
Perhaps it’s good to remind you that in the ‘80s we heard the same comments about those nasty sounding CD players. They measured great but sounded poorly. We now know what to measure. But it took at least five years before the first measurement devices offered very limited jitter measurements. So we heard correctly but didn’t know what to measure.
May I suggest you do an episode on phase correctness? In source, amplification, cables, filtration, actual loudspeaker, loudspeaker placement, and listener positioning. I.m.h.o. an important factor that is often neglected.
Could be, but you barely mention it. And you don't discuss the utter importance of it. In this domain I was totally flabbergasted when I compared my KEF 107 Reference to the Wilson Audio Cub 2 and other d'Appolito setups. A world of difference. They even presented spatial bass, which I thought was theoretically not possible. But when harmonics of bass instruments reach your ears with correct timing, it most certainly is.@@TheHansBeekhuyzenChannel
Thank you for the nice co-operation, Hans! It was very educational!
tx to both of you for sharing your insights
Glad you enjoyed it!
I appreciate your drive to find the truth
🙏
Hello Hans, as always a great contribution from you!
I myself have been dealing with the topic "Ethernet sound" for a long time and am also in contact with Alex from Uptone and have also exchanged with Jaap about his first measurement.
I have my own blog on which I have shared various measurements. All this is not because I want to discredit audiophiles, but I want to understand what is really going on.
My findings in a nut shell:
Most of the sound change is caused by common mode noise in the Ethernet signal. This includes the 50Hz leakage currents John talks about in his paper. But also RF noise that enters the ground system in the receivers and then becomes noticeable as in Jaap's measurements, or even takes effect during the conversion itself. The muon filter inserted in Jaap's measurements, I believe, is nothing more than strong common mode chokes reducing common mode noise.
Thanks for sharing
TL;DW "I measured something and I assume that's why I hear a difference."
Quite the leap in logic.
😁
Why would the network matter if the whole song could be transferred and buffered on the player within a second of starting playing. Then there isn’t any additional traffic and the song is playing from internal memory. As far as Ethernet, the player could just optically isolate the port so there is no electrical interference to filter out.
Then you assume that buffers do what they supposed to do perfect.....
@@TheHansBeekhuyzenChannel yes, they do, otherwise you wouldn't hear less clean audio, you'd be getting an awfully gltiched mess that's basically unplayable.
I'm wondering why nobody that claims switches make a difference will provide a simply test of packets integrity with Wireshark
Because the packets do arrive perfectly. I have explained in several videos it’s the noise and distortion of the analog signal that is used to transport the digital signal that cause minute timing errors on the clock input of the digital to analog converter.
But the ones and zeroes that are playing from the separately clocked buffer memory don't care if the transmission to it is riddled with timing error. 10010110 transmitted with timing error is the same as 10010110 transmitted without timing errors because the buffer memory uses its own clock.
Admitting you were wrong is a noble act. Kudos!
For me that's standard practice.
If weren't so stubborn, it would be a full time job for me. @@TheHansBeekhuyzenChannel
Excellent video, Hans. It is in your top 5 in your ever-growing video archive. Great curiosity and follow-through. Nicely done! Thank you..! 😊
Many thanks!
Seems like the old “noise on AC mains line affecting analog signals in your amplifier” argument.
Kind of..
We have a LOT of measurement training but essentially ZERO listening training. If we want the world of audio to improve, we need to learn to LISTEN. That will not happen by DIY.
So true
Just to note, there are a couple of standards that can improve jitter and/or packet loss on ethernet. They are AVB ("audio video bridging") and DCB ("data center bridging"). In both cases, they require hardware that supports the standard, and is not something you can do on a generic ethernet switch.
Can you name a streamer which supports this?
It’s NOT about jitter in the Ethernet signal
Greatly appreciate the work here. Fascinating results.
as always great info.
thank you for your work and information.
look forward to continuing the audio adventure
Thank you! Cheers!
Good video. Important for audiophiles to once again consider that digital is digital. If it arrives uncorrupted, it's fine.
There are so many companies who knowingly make products which are stupidly expensive that make no difference. Less technically inclined people will not know, buy it, and because of confirmation bias think it works. That money could've been spent on better speaker/amps (that make a huge difference) or a nice dinner with the family.
@@techtt6213Yes. But say you have 200000$ system (and it happens a lot) - would you not spend say 2000$ to make it better?
@techtt6213 I don't think you got the message in the video completely right...
@@No_Limits_411nope, I even strongly advised against it
@@TheHansBeekhuyzenChanneljitter doesn't matter, on the receiving end it gets compensated. Was the listening done blindly? I often did a blind test (or even lied to have some confirmation bias) between Spotify over Bluetooth (yes) and a wired Qobuz stream. Either blind, or told people they would now hear Qobuz whilst really it was Spotify. And every single time confirmation bias won and they'd describe it as more natural (favorite term) or more open. Needs to be blind, if you can still tell the difference multiple times I'd agree.
We love your attention to the details and your pursuit of the truth.
👍🏼
I will most likely never own a Muon, as hearing the difference is dependent on the variables you mention, some of which I cannot control. I have just spent $800, mostly on bass traps and some diffusion. I instantly heard a difference when I quickly positioned the just arrived traps in my room, I am now in the longer process of dialing in placement of the traps and calibration of my Genelec speaker system, but it's clear to me that it was money well spent. Although you did say that better upstream clocking is a component of achieving the "natural" sound we pursue, the distortion off the the analogue signal that carries the digital information you also discussed had a larger resonance in my mind. I also read somewhere in the Uptone literature that using a more accurate clock than the one already in the EtherRegen had not much effect. I managed to buy a used ER and a quality used ethernet cable for ~$650 earlier in creating the endgame set up I have now, which includes a Wattson Madison, largely because of your reviews. Thank you for all you do for us Hans!
Money spent spent on room treatment (traps, diffusers etc) is money well spent. I’m in the same position and like you heard differences immediately
It's the complete chain that counts, including acoustics and filter.
Thank you for another fantastic and informative video, Hans. It never ceases to amaze me how many naysayers and detractors we have out there who choose to embrace their (limiting) instruments that spit out data that’s quintessentially useless, RATHER than choose to rely on their ears. But the concepts you explain here in detail explain so much, so thanks again for your dedication to our education. And BTW, I’m a huge fan of Alex and EtherREGEN. 😊
Many thanks!
Hans you still are not understanding how digital audio transport over a network works : as you said, there's buffering on both ends of the IP transport part, so IP/Ethernet jitter measurement is totally irrevelant. Those bits are not directly transfered to the DAC chip, but debuffered by the DAC itself at its won pace (thats is, with its own internal clock)
I am no network specialist. But when it comes to audio, I know a thing or two. My guess is, more than you. I am learning about the influence bit transport has on digital audio. But regardless whether all bits arrive in tact - and they do -- there are influences on the sound quality and what I described in this and other videos were base on the best explanations of digital audio experts. Denying what's audible doesn't bring us further.
How much do you see cable length or number of hops affecting the outcome? Does more ever equal better sound? What's makes these devices so expensive - is it R&D?
If you use an Audiofile switch, keep the distance between it and the network player short. Prices are because of R&D, components and small series.
Okay I watched this video I upgraded my switch and as you say the improvement was very good. Now do I buy audio Ethernet cables?
Buy a good one for between the audiophile switch and your streamer
Great cooperation between my to dutch favourite reviewers!!
Glad you enjoyed it!
Thank you for taking the time and effort to investigate this matter and then make it clear to us. Really well done! Yet, the MUON filter better measurements are not exactly spectacular. Does this mean there may be even more to be achieved with signal filtering? How much even lower can we go with jitter and phase noise?
The measurement was relative. The difference in sound quality was drastically better. Measuring phase noise depends on many factors, like power supply, cables used, EMF and so on.
Good day Hans, I have always loved your videos. The information and moreover the background work you put into same is to be admired. Being an engineer I have struggled with many aspects of Audiophile accessory sales, I will add at this point that my ears are not perfect and do not pick up everything and that my experience with electronics is not in the field of audio. However having worked with very complex analogue and digital distributed control systems I struggle with many claims.
For you to release this video is admirable, thank you for restoring my belief in my own training and work history. Keep up the great work you do, I will always be watching. Stay well.
🙏🏻
Thank you very much for your time and commitment. There is obviously a lot to explore and verify so that the „bits are bits“ people can follow too.
My pleasure
@@TheHansBeekhuyzenChannel Have you also seen the comment by UpTone Audio? They say there was no reply from you.
If data integrity is maintained by every switch and streamer interprets data, how it could affect the sound? Does road quality determines quality of the goods transported on the track? No, until integrity of the goods is not affected. Is there any filter or conditioner for the wifi? Can you hear the difference between files played from different hard disk drives or memory cards?
If a drive is in a case with a noisy psu, I might. It's not the signal itself, it is the carries of the digital signal that is influenced.
@@TheHansBeekhuyzenChannel sorry, but a digital signal can't be "noisy". It's 0 and 1 (not quantum bits...) and if they are corrupted, it can be detected (and sometimes corrected) by error correction codes.
@@eya83fr I think that's the disconnect, although I might not understand. Even light takes time to move. I think he is looking at the tiny nearly immeasurably differences in the signal and believes they affect affect the sound. Like maybe a 1 or 0 has a slightly higher or lower level, shape, or length. I think it's similar to why it's difficult for hardware midi to sync than expected. But I would think here it would either break the signal or add a tiny bit of latency to the whole stream. I can't imagine a halfway decent filter or converter having an audible difference from this though, but I'm not entirely surprised if you measure it with enough detail there would be a difference in the transmitted signal itself
Thank you very much for sharing your insights gained on this issue. I especially like to hear a man say, “Well, I was wrong.” I know there is good stuff coming. I too have been around networking since 1985 or so, but I still don’t understand it all that much. Just getting into its use in audio. Your video was most interesting, I’m now in the market for a passive filter. :-)
Thanks man
Good Hans, so I’m using WiFi when streaming with my dmp-a6me with lps, I’m running a usb from dmp-a6 into denafrips iris DDC and toslink from iris into sonnet morpheus Dac. Will I improve my sound by running a Ethernet with a network switch?
If you choose the right one. See my reviews:
ruclips.net/p/PLMbsmejHnP8HxdJ89h4s30gXO2u42-xmR&si=jKvzA7tMr4qg_Sqi
No you will not as long you have a good signal, save your money, don't fall in this rabbit hole.
Brilliant as usual. Thank you Mr. Hans 🙏
My pleasure
Hi, I have a technical question that may be very basic. In a digital signal, each bit has a value of either zero or one, which is represented by a specific voltage in the signal. Does this mean that a 16 bit recording must measure voltage 16 times in order to have a complete representation of the binary number that represents the sample?. How does this relate to sample rate? A sample rate of 192khz, for example, does it measure voltage 192000 times? and every "X" bits represent a full sample? or does it measure 192000 samples, each one with several voltage measures to build the full sample
The signal is measured 192 thousand times and the measured voltage, 1 volt, is then converted into a binary code using 16 bits. The binary code is just another way of writing that 1 volt.
@@TheHansBeekhuyzenChannel thanks for your response! But then the representation that you make in your videos is not fully accurate? In this case the signal may have as many possible voltages as 2^bit depth (so that the DAC translates a very specific voltage to a binary number)
@@dfrancoanzola
You appear to have it the wrong way round, The ADC (analogue to digital converter) measures the signal and creates a binary number according to the input amplitude (and polarity, positive or negative) of the incoming signal at the instant the signal is sampled. That's done in the recording processes. At playback time the DAC takes that binary number and creates an analogue voltage according to the binary value it receives as input.
Hi, Hans. I am using an Ediscreation Silent Switch OCXO and an Ediscreation Fiber Box II. The change both brought to my system is incredible. Both have OCXO clocks for the ethernet digital signal and the Fiber Box II internally converts from wired to optical and back again, thus galvanically isolating the signal from the modem supplied by the internet service company to the switch. In addition, I have a filter from PS Audio that is placed at the entrance of the cable that comes from the street into the aforementioned modem.
Thanks for sharing
There seems to me to be one big question here: how is a passive device able to improve phase noise? Unless I completely misunderstand the definition of phase noise, it's essentially a measurement of the deviation from perfection of a periodic signal. How can a passive device, such as the Muon do this?
This video article's title seems to me a little unfair and misleading. Fine, it's a fair and balanced review of the Muon, with an added technical assessment. But I see no equivalent review or analysis of any network switch. Will we be seeing an equivalent suite of measurements of the EtherREGEN?
Myself, I am the grateful owner of an EtherREGEN, and I would state that it had as big an impact on my system as the Innuos PhoenixUSB did. Why the digital signal needs regenerating before AND after my Zenith streamer has long been a puzzle to me, but one I happily accept. Returning to the topic of phase noise, I also added an AfterDark OCXO clock to the EtherREGEN, plus an optical stage to precede it, plus decent LPSs to both clock and EtherREGEN. All, to some extent, have offered benefit.
Perhaps you might want to watch this video: ruclips.net/video/B-StTplQZys/видео.htmlsi=UUmwV4JJ7iSV6xIW
Thank you Hans for all your insights on hifi. I was wondering if the Roon RAAT protocol, which they claim is 'bit perfect' makes a difference in streaming music vs for example steaming from Tidal direct?
It is easily tested using the Audio Precision behind me. And, indeed, it is bit perfect. But there is more to a digital audio signal than bit perfect. Watch ruclips.net/video/ZCFvIzzMqfk/видео.html
@@TheHansBeekhuyzenChannel Thanks a lot. I'm new to this and I'm trying to grasp it. I'll rewatch the video soon.
To my understanding audio (and video) streaming utilizes UDP. UDP packets beneifites from low latency, but there's no possibility of retransmits or rearranding of packets. They are sent in an ordered stream. So the quality of networking components could potentially have an impact on the stream, eg. packet loss. Please correct me if I'm wrong. BTW: I'm not a network engineer either.
Have you tried an audiophile switch?
@@TheHansBeekhuyzenChannel No, I have not. I do not experience packet loss in my simple setup at home, so I believe my equipment is of adequate standard. I only have a broadband router with WiFi and four Ethernet ports, One of the ports is connected to my NAS (wired short distance). Streaming endpoints are connected via WiFi (Marantz SACD 30n, Mobile phones, tablets), A WiFi extender helps were the signal otherwise would be weak.
I just wanted to point out that your statement in the video regarding retransmission and reordering of packets does not apply to UDP connections/packets - which is the protocol used for media streaming (if I'm correct). The UDP protocol only has a checksum. If the checksum doesn't match the data (stream) at the endpoint, then the packet is discarded. This will be audible and/or visible of course.
Used to be the case but now most use TCPIP an error correcting protocol.
Doesn't wireless transmission in the form of wifi help at all? Pretty impressive speeds we get these days over wifi.
It’s not the wifi technology as such but the way it is implemented, including the locally available bandwidth. In an apartment building that will be more difficult than on a ranch.
Not for the first time, I wonder whether having Golden Ears is a gift or a curse! Does Hans actually like music - he never mentions the music, only the technicalities.
There’s no such thing as golden ears (great title for a next James Bond movie), only an accurate hearing which needs to be trained. And you train it by listening to music 🎵
Amen
Hans does mention the music, he has ten videos about the music he likes: ruclips.net/p/PLMbsmejHnP8HvTcBKS2pA_YnWPvAczgJT&si=ly3J-0QpIHlJpocN
@@jmtennapel you listen and enjoy the music or you end up in the world of Audiophile and end up buying as an example SR vibrotrons, Audiophile fuses and sticky bits of metal you stick all over your speakers convincing yourself it has made a difference. Spend what you like but this hobby is about enjoying music. If you understand networks and data transmission you know how this works.
@@a0r0a7 ?
Like it has been said: what is heard, it cannot be unheard 🙂🙂🙂
😆
Last week I bought a Grimm MU1 this combining with the MM Tambaqui. Does it still make sense using the Etheregen which I already owned or can I remove this from my system?
Did you listen to it?
I had the Etheregen between my switch and the MU1 and removed it, I can't notice any difference now, in my old situation it did differ
Not surprised as your streamer is uniquely exceptional. Already built in.
You made me happy by ending with a bombshell again. I missed them for some time.
I'll keep that in mind
I am grateful for you and your great contributions
My pleasure
Again, great info Hans!
I was using an etherregen close to my SoTm sms 200 Ultra Neo streamer in my set up and wanted to improve the cleaning of my LAN (my router and NAS are downstairs- roughly 40 feet away). So I added a Ifi LAN ISilencer ($89 USD) at the output of my router (going to the EtherRegen). An instant big improvement… the biggest being with bass notes; the pitch is so much more perceivable, with more punch and body. The timbers are improved in such a more credible, natural way.
So yes, clean up your stream and everything is going to sound closer to the “real thing”.
Thanks for sharing
Would you like to plan a test comparing measurements of the Muon with for instance the more affordable SOtM passive Lan filter (that i use) 😮
I try to elect reviews that seem to make sense. This one doesn't .
Hans, this is certainly a great video and I appreciate your work and competence a lot. But ultimately I didn’t really understand it - please forgive my ignorance as a formally trained electrical engineer, decades ago. Somehow rather prosaic questions aren’t answered. I’m sure you could do that with ease: a) what does this filter actually do? b) where and how would I need to place it in my stereo chain (between which gear exactly)? C) I have a setup with a relatively good DAC and active monitors in a decent room, BUT my digital signals comes from a NAS, via a Switch, via a WLAN rooter, via relatively weak signal (our house is rather large), through a WiiM pro streamer to the DAC
; isn’t in such poor setting a “filter” doing anything good?
In short, do you have some practical guidance for practical settings? Cheers.
What this filter actually do? We son't know. Apart from reducing phase noise, of course. It is placed in between the streamer and the network, closest to the streamer. It doesn't seem a sound investment given the streamer you use.
@@TheHansBeekhuyzenChannel thanks, that's what I thought.👍
Great work Hans but I can share that my high end audio system with monoblock amps benefited greatly with a Wiim Pro streamer. With a Verizon fiber router, a Cisco switch and fiber, the improvements are nothing less than excellent. I’ve heard the benefits are obtainable with Juniper and other commercial switches for far less than audiophile versions. Agreed the quality of the system will play a major role. Thanks for your strong contribution to music quality and understanding.
Thank you Hans, I appreciate your work & efforts. Take care.
Thanks, you too!
Picoseconds... I didn't see that one coming. Glorious work, Hans and Jaap! Visitors wonder (and smirk) at the laboratorium - like aproach audio seems to require at my place and probably yours, fellow audiophiles. But Hans is so right: It only takes secondes to hear. And there is usually no way back once you heard It.
Oh well, still great times we live in with the equipment available. And the reviewers that help explain what we hear.
Ears and hearing can distinguish down to about 10 picoseconds, this is where MQA makes a difference.
👍🏼
Many thank to you Hans!! ❤
👍🏼
I learn more here and quicker than anywhere
Thank you very much.
I got the iPower PSU to improve my Netgear switch (recommended by Jaap) but that seemed to make the sound very sharp and think sounding. Any idea why that could be the case? I had to switch back to the factory supplied PSU...
My guess would be that the iPower psu makes the switch more precise. That will make the DAC have more resolution leading to a sharper sound due to other problems in your setup. Like interconnects that use poor silver. But of course I can be totally wrong. The iFi psu really is good for that money and most likely better than the one that came with the switch
@@TheHansBeekhuyzenChannel Thanks Hans. I was wondering if the PSU having more Amperage (1.5A) vs the standard PSU (0.5A) could have anything to do with the observation. Then there's the central house router that feeds the switch via LAN... Wondering if just having one switch in the chain improve the pSU enough... thanks
I have tried the ifi PSU too (the X version), I believe on the same switch, and while I found it quite accurate, it sounded quite a bit too much of a good thing, so similar to your observation.
And I would think it might have to do with the PSU’s bleeding of HF hash into the power line, not its DC output. You might try that: if you insert the psu and power some device that is not linked to your hifi gear you might notice a shift in SQ. Or not ;) - anyway I tried tried to eliminate as many switch mode power supplies as i can, the remaining ones get ferrite between socket and psu, to reduce HF hash. And it did improve things in terms of treble smoothness. now I use a linear Psu for the netgear switch, an sbooster, with the ultra device and - an ifi dc ipurifier at its output (yes that is crazy) but its better ;) possibly overkill, but I found it an important improvement wrt prat and dynamics. Not sure what is going on there.
Audiophonics too have an ok lin psu for
The Muon sounds like a good product but I would prefer a more cost effective filter. The Muon is about $3,000 here, something almost as effective for half that price would find a place in my system.
An Eno 2 review in in the pipeline
Don’t all these excellent reviews of streaming devices make a good case for the cd player?
I haven't owned a CD player for donkey's years. And don't want to go back to them.
@@TheHansBeekhuyzenChannel I guess cdp introduces problems of their own. Apart from their inherent limitations of music choice.
You have made me interested to try out the Muon filter with the Pachanko server I run. Thanks.
Well if Network acoustics invented and patented this improving device they should be doing very well. If they have not protected it there is no stopping others from getting one and breaking it opening and seeing what is in there and if they choose include it on their streamer ethernet port. Maybe expensive streamers like Grimm already has it?
I love your work
Me too😃
Hans my point has always been that with local rendering with the music source @ device.( as a baseline for operation)
All those readings can also be influenced by .02 volts I've built Linear Power supplies with 8 isolated outputs including my network switch.
( ( preventing device to device noise is critical)) in the power chain
I would rather use a good, audio grade SMPS for the switch.
You can clean SMPS to "HIFI GRADE" with over the counter parts then add ferrite to each cord to reduce EMI. Preventing noise propagation is the actual goal
Great video. Research. Thank you Hans
My pleasure.
I always respect the truth 👍
But, as always, the question remains what is the truth? The truth of today or that of tomorrow🥴
@@TheHansBeekhuyzenChannel this is the million dollar question 🙋♂️
Hello, of course these test executed by people owning commercial businesses are very easy considered a marketing tool. The bigger the number of devices they can sell as being " audiophile grade " the more money they will make. It seems that the Grimm devices are kind of immune to the garbage that comes along with the signal delivered by your internet provider. EtherRegen with a audiophile quality power supply, some nice feet, aftermarket power cable, ethernet cable and you are getting close to 2000 € additional cost to a device that is not exactly cheap. Sometimes this audio hobby is getting boring. I already decided that after receiving the Grimm i will stop investing in digital gear. Greetings,Eddy
The Grimm player doesn't need these peripherals.
@@TheHansBeekhuyzenChannel Hello Hans, i already have the Etherregen with an improved power supply but probably it won't improve anything so it is good to know! In your mu2 thread you already mentioned that very probably i won't be needed. Which makes things easier. Reducing the number of power cables and " interconnecting cables" will save me some money and should bring some improvements. But on some " marketing site" USA based i already see people asking about upgrading the fuse. I already have a bamboo chopping block so i will try that one. There is info on the internet about diy vibration killing feet using copper sponge as the key element!! Greetings,Eddy
The internet and associated online services work perfectly well due to Transmission Control Protocol and Internet Protocols.; Ethernet itself is differentially transmitted, etc. Concentrate on your internal system clocks.
Did you watch the video?
1- as an IT hardware professional, i know for sure that switches leak data. Most likely due to not switching fast enough. I'm assuming the opposite should be true here as well, ie. they block out bits that should have been let through. Now computers are designed to handle this without breaking a sweat. But do you think this negatively affects audio?
2- Is this device acting as a buffer (a storage buffer type) between the switch and the dac input?
This is the first time I hear this from an IT professional. That will impact isochonous connections. But should not happen with asynchronous connections I would say.
For a home network this should be extremely rare unless you have very very bad setup. This would show up as dropped packets and can be tracked. If you are getting dropped packets at all on a wired connection most likely you have some defective equipment.
@@a_macaulaygood point. It's a bug but was it a result of hardware going faulty or just the way it does it's switching/processing or a combination of factors, I don't know. This was me doing some testing on my home network. Specifically, the ISP provided switch (no not a router and connected before the modem). What I found was it was randomly letting through 5-8 bits when it should be blocking that data (due to a different ip configuration, very complicated...). This is what led me to wonder if the reverse is true as well, ie. blocking data that should be let through, say for example the first 5-8 bits of a stream.
Now, when one is referring to computers or servers etc. the TCP/IP protocol is the first layer that is designed to easily rectify such lost bits. So it doesn't always amount to packet loss. In fact, the errors have to be seriously bad for the TCP checksum correction to fail and result in a packet loss.
Also, the TCP protocol (or any other protocol ftm) can ask for retransmission of the lost packet in case of uncorrectable errors. This again, the packet is not fully lost, ie. it's recovered. Packet loss would only result if the data can not be retransmitted, due to things like being flushed out of memory or not being in memory in the first place. That's as far as computers are concerned. Music is a whole different topic. Here, bits are not (just) bits. It is time sensitive, and extremely so. Hans should be aware of various DAC configs in this regard and resulting effects. Also, audio gear is nowhere nearly capable of possessing as much processing power.
Also, regarding the retransmission request of lost data packets - the USB 3 and above standards can also request retransmission in case of failed checksum for data integrity, just like TCP/IP. I have no idea which audio gears actually utilise this.
Retransmission request and receive times on a home network, or a USB 3 connect between a host and recipient, definitely would be in picoseconds, even if at the higher end of the ps scale in the worst case scenario. So request and receipt completed even before the entire data transfer completes.
All this raises a further question - the time factor. I had gone down this rabbit hole long ago, and it had looked very interesting to find what was there, but I guess it was too far down ...
@@TheHansBeekhuyzenChannelah yes. I remember the asynchronous & synchronous part now. Also, please do refer my last comment here.
Just leaving it here for you, in the hopes that you are the best person to be able to use this, if it fits, and maybe dig a little deeper for us. 👍
Again, thank you Hans.
👍🏼
Merveilleux ❤ merci beaucoup 🙏
Avec plaisir!
I use WIFI instead and have never heard any difference, less cables to the people.
It depends on how busy the WiFi band locally is.
Thx Hans for the info.
🙏🏻
How can we proof that the measurements are legit?
If we only knew....
No proof needed as long as someone hears it or thinks they hear it and whose to know, as anyone who doesn't obviously has crap audio equipment or ears. Thank god theirs always a fix, even if it changes week to week etc, by some expensive box or five and magical cable. Its a wonder the rest of humanity can bare to listen to music at all.
Enjoy the music
Thanks!
Anytime
keep up the good work! people need to know this info as digital glare and time smearing can totally ruin the music. Thank you for your clear and considered commentary. a great digital front end allows me to listen to a wider variety of music that previously was quite tiring or simply un-engaging.
Enjoy the music 👍🏼
Thats the commitment and drive that is uplifting 👍👏👏
🙏🏻
If I undrstand correctly non of the network players work correctly except just one, because digital signal shouldn't influence the analog output, end of the conversation.
Basically the problem same as in digital TV if there is distorted or no signal I got a green frame or block on the screen, unlike analog TV where I get fuzzy or distorted picture. Live TV Sport broadcast do this battle for years they rather bradcast 720p 60 FPS lower quality but faster information, than 1080p 30 FPS higher quality lower speed, because there is no enough bandwith for 1080p 60 FPS.
If everything what you mention in the beginning of the video about digital audio is in order there souldn't be any problem. Digital data comes in, put it in a buffer and DAC converts the digital data into analog signal. If the buffer is big enough and the delay is long enough there will be no problem at all. For me this is clearly a digital processing error. What a car or airplane chip or my PC handles with ease a way overpriced audio gear couldn't handle. I bet this problem doesn't exist on low end gear, for example if you play 128kb/s 44.1kHz mp3 on a 16bit 44.1kHz capable device.
As a technical person I highly agree with other technical people, if you can hear it you can measure it, simple as that. Record the audio put two plots on eachother there is your difference, we have the answer for the "What" question, and for the "Why" the equipment is faulty/false advertisement, doesn't know what the manufacturer states.
A well designed device should never accept a jittered signal in the first place, so just like in digital TV you have a perfect image or non at all, if you have a bad cable you should hear nothing not distorted sound.
Perhaps it’s good to remind you that in the ‘80s we heard the same comments about those nasty sounding CD players. They measured great but sounded poorly. We now know what to measure. But it took at least five years before the first measurement devices offered very limited jitter measurements. So we heard correctly but didn’t know what to measure.
May I suggest you do an episode on phase correctness? In source, amplification, cables, filtration, actual loudspeaker, loudspeaker placement, and listener positioning. I.m.h.o. an important factor that is often neglected.
Every device I review is measured, including a phase measurement.
Could be, but you barely mention it. And you don't discuss the utter importance of it. In this domain I was totally flabbergasted when I compared my KEF 107 Reference to the Wilson Audio Cub 2 and other d'Appolito setups. A world of difference. They even presented spatial bass, which I thought was theoretically not possible. But when harmonics of bass instruments reach your ears with correct timing, it most certainly is.@@TheHansBeekhuyzenChannel