Melco S100 Ethernet Switch Measurements

[firedog]

33 minutes ago, ASRMichael said:

“and make no claims about them other than how they sound to me”

 

So I understand you correctly, you are saying if they sound good you’d be happy to have it in your system? That makes sense to me. Have you tried the EtherRegen or Melco switch in your system? 

That's not what I said. I will sometimes post about how something sounds to me. I try to be very careful not to imply or say that that means anything about how it will sound to others or what it does to a system/SQ in general.

 

In my present setup with the Kii Control and Kiis I don't think add-ons do much, if anything. In general over the years I've come to doubt such devices do much in general, as almost all the "evidence" is based on sighted anecdotes, and in my own testing I can't find consistent improvement if testing is unsighted. But, of course, there may be specific setups where they improve/change SQ.

 

So if you say it sounds better to you, I won't comment. If you make a claim that such a device generally improves SQ with various setups, then I'd like proof. That's what manufacturers do, and they have a monetary incentive. Hence I'd like the claims backed up.

If such devices improve the output at the DAC enough that humans (especially those over 40 years old) can consistently hear it, it should be measurable in some form at the DAC output: jitter reduction, S/N improvement, audio difference comparator, etc. Note that in a previous post I linked to measurements done for such a device that showed it can make an improvement which should be audible - at least with some devices.

 

As far as the two devices you asked about - they are quite expensive and where I live I'd pay quite a substantial customs/tax bill to import them. Since I'm skeptical about them providing consistent SQ improvements, and even if I return them for a refund after a trial I'm out quite a bit of money - I probably wouldn't give them a try. If you want to bring me one for a trial, I'd give it a whirl.

[plissken]

7 hours ago, R1200CL said:

Yes, i watched the video. I can unplug here as well. It will play quite short. 
I suppose the point with that video is only to show that there is a buffer present ?
 

So that’s why I’m asking where is the buffer ?
I also have two Cisco here. One is SMG 300, and the other is 8 port 2960G which I haven’t been able to log into yet. (But I think I know why). 
 

Can either of those two switches be set to extend the buffers ?

My point, and it should be obvious, is that when a switch delivers data ahead of play back and you can literally pull the plug, according to the leakage current argument that a difference should be audible.

 

The second argument about phase noise is superdad either trying to put one over on people that don't know any better or it is a staggering lack of technical understanding. 

 

Phase noise can only happen during data tx.

[Superdad]

24 minutes ago, plissken said:

The second argument about phase noise is superdad either trying to put one over on people that don't know any better or it is a staggering lack of technical understanding. 

 

Says the network jockey playing with data analyzer bits to the chip engineer (John Swenson) of 31 years who actually designed the power networks Ethernet switch chips and PHYs and understands and measures what happens at the lowest levels.

 

Ground-plane noise is real, caused by both leakage currents and clock-threshold jitter. Impact of such transfers through the PHYs and all other chips, ultimately affecting what goes on downstream--buffers or not.

[plissken]

29 minutes ago, Superdad said:

 

Says the network jockey playing with data analyzer bits to the chip engineer (John Swenson) of 31 years who actually designed the power networks Ethernet switch chips and PHYs and understands and measures what happens at the lowest levels.

 

Ground-plane noise is real, caused by both leakage currents and clock-threshold jitter. Impact of such transfers through the PHYs and all other chips, ultimately affecting what goes on downstream--buffers or not.

 

Clock Threshhold jitter is only happening on actively sending interfaces. The quicker the interface the smaller the window. There is occasional management frames on unused lines.

 

So your jitter argument is borderline criminal.

 

Name me the computer based setup that I can hear the difference with the cable plugged and unplugged during play back please.

[R1200CL]

1 hour ago, plissken said:

My point, and it should be obvious, is that when a switch delivers data ahead of play back and you can literally pull the plug, according to the leakage current argument that a difference should be audible.

Again, where is the buffer (that will remove phase noise, “threshold jitter,”) located ?

 

[plissken]

30 minutes ago, R1200CL said:

Again, where is the buffer (that will remove phase noise, “threshold jitter,”) located ?

 

 

Same place it was last time I told you. It hasn't moved. It's also in the video I posted.

 

Does phase noise/jitter exist on a DSD or PCM file that is saved to your hard drive?

[R1200CL]

On 9/16/2020 at 7:23 PM, plissken said:

I concatenated a ~45 minute album into one .wav file and flac'd it to 445MB.  I start playback and in the time it takes from clicking play to unplugging my network cable the entire song is in buffer.

Where is the buffer ?

Video only telling your plugs in and out cables. And music still plays. We dont know actually the setup. We don’t know if you’re pinging the server, or something else. And for sure we don’t know where that buffer is located. 
 

There is no need to be arrogant. 
 

I’m asking so I understand if I also can find a buffer somewhere. 
 

Also will this buffer magic work for streaming services ?

And if not, why ?

[plissken]

33 minutes ago, R1200CL said:

Where is the buffer ?

Video only telling your plugs in and out cables. And music still plays. We dont know actually the setup. We don’t know if you’re pinging the server, or something else. And for sure we don’t know where that buffer is located. 
 

There is no need to be arrogant. 
 

I’m asking so I understand if I also can find a buffer somewhere. 
 

Also will this buffer magic work for streaming services ?

And if not, why ?

 

In the video the playback software is JRiver. It has a setting for up to 1GB of buffer. Tidal will buffer entire tracks. I don't know about other streaming services.

 

Sorry I thought it was readily apparent that JRiver was the application that was playing back the music without a break even with the cabling unplugged (the continuous ping).

[R1200CL]

2 minutes ago, plissken said:

 

In the video the playback software is JRiver. It has a setting for up to 1GB of buffer. Tidal will buffer entire tracks. I don't know about other streaming services.

 

Sorry I thought it was readily apparent that JRiver what the application that was playing back the music without a break even with the cabling unplugged (the continuous ping).

Then we must understand that buffers only applies to those that plays directly from PC to DAC via USB, with SW allowing you to set a buffer. 
 

I must be wrong?

[plissken]

4 minutes ago, R1200CL said:

Then we must understand that buffers only applies to those that plays directly from PC to DAC via USB, with SW allowing you to set a buffer. 
 

I must be wrong?

Not correct. High end streamers like Lumin can have several minutes. Low end units may have just a few seconds.

 

Since PCM has been around awhile I would hope and expect purpose built streamers to increase in buffer size since Moore's law is in effect: That is computing transistor count doubles every 18 months. While PCM data rates have stayed the same over the past ~30 years.

[Blackmorec]

To follow up on my previous post, consider a high resolution system, using a server with large buffer, internal storage and an operating system that shuts down disc access while playing music from the internal RAM......

The SQ of the system playing music from the internal storage and RAM improves when either; the network is improved or the network is unplugged. So the improvement is nothing to do with the data coming from the network and more to do with whatever the network is injecting into the system. Its why implementing a fibre optic link improves SQ, why increasing the quality of network power supplies improves SQ and probably why even cables make a difference. Perhaps its also to do with network traffic and how non-audio related packages impacts the system.

All I know is that improving the network improves SQ, buffers notwithstanding. 

Regarding the video....kind of Plissken to go to the trouble to illustrate his argument but there’s no way that what’s left of the SQ after cameras, microphones, compression algorithms and crappy replay devices is sufficient to hear even major differences or musical attributes. If there’s absolutely no recording venue ambience, imagery, sound stage, performer detail or timbral information present to begin with, there’s no way you’re going to hear improvements in those regards

 

 

 

 

 

 

 

[ASRMichael]

The world is flat.....We never went to the moon.....We have two suns.....

[R1200CL]

10 hours ago, plissken said:

Not correct. High end streamers like Lumin can have several minutes. Low end units may have just a few seconds.

This is interesting, cause let’s say you have a 5 min buffer, you would also like to be able to at lest operate stop/play from that buffer. 
This will in most cases require a network connection. 
So if you’re pulling the plug, you can’t do it in that way. 

 

With fiber this shouldn’t be an issue in any case. 
 

I notice there is a clock buffer in the opticalRendu, but I think we’re talking about another buffer here. 
Pulling the fiber cable at my etherRegen fiber side to my opticalRendu indicates no buffer. Or not more than a second. 
Pulling input etherRegen gives same results, expect establish a connection takes longer time. 
 

So maybe there still could be some improvements done by John Swenson design when it comes to buffers. I don’t know if a huge buffer is an issue working with RAAT, but since Lumin is Roon Ready, I suppose not. 
 

Then one can probably argue that buffers should be located in the DAC.

 

Understanding fully the digital chain with all sorts of interfaces and requirements for clocks and buffers isn’t easy. Maybe a tread or white paper on that would be nice. 
 

 

[plissken]

7 hours ago, Blackmorec said:

So the improvement is nothing to do with the data coming from the network and more to do with whatever the network is injecting into the system.

 

And this where I don't think you can cash that check...

[plissken]

2 minutes ago, R1200CL said:

This is interesting, cause let’s say you have a 5 min buffer, you would also like to be able to at lest operate stop/play from that buffer. 
This will in most cases require a network connection. 
So if you’re pulling the plug, you can’t do it in that way. 

 

You sure can with Tidal and JRiver...

[R1200CL]

7 hours ago, Blackmorec said:

Perhaps its also to do with network traffic and how non-audio related packages impacts the system.

I really don’t hope that’s the case. I can’t be. It’s impossible. Then we would need a dedicated audio internet from streaming services 😀

However bad home network is another story. But still non-audio related packages shouldn’t be an issue. 
I suppose @plissken can educate us if QOS and similar technology in managed switches will help.

There must be a reason why ip-phones gets a priority, so why not audio ?

Personally I think good terminated cables and well known brands in use with router and switches will help. Probably plan for 10GB fiber. But that maybe is an overkill. 

[R1200CL]

21 minutes ago, plissken said:

You sure can with Tidal and JRiver...

I think you missed the point. (Or did I miss yours 😀). 
 

How do you communicate with your PC. Yes, of cause with mouse or keyboard, so you can do if your PC is USB to DAC.
But unless a pc isn’t your endpoint and player in one package, you can’t control the buffer (stop/start) remotely. 
 

You test has only a value in some very specific environment. And as you know there is in general accepted that a PC is the worst endpoint. With expensive exceptions. 
Maybe why you couldn’t hear the benefits of the etherRegen?

[plissken]

1 minute ago, R1200CL said:

I suppose @plissken can educate us if QOS and similar technology in managed switches will help.

There must be a reason why ip-phones gets a priority, so why not audio ?

 

QoS and DSCP are traffic markup technologies for priority traffic (delay sensitive, non-bursty) normally real-time. Audio playback isn't real-time so it's delay insensitive, and bursty in nature.

 

Take for example Vocera PTT devices that are used in hospitals. These are a puck that nurse and MD's wear that they can push a button and for instance ask for a crash team. This has to go over the intercom and it has to go now. So I setup IGMP at L2, PIM Sparse across the entire route to the Vocera server (this sets up the Reverse Forwarding Path) for the needed multi-cast address.

 

Next on the ingress/egress routed interfaces we set Diff-Serv code to EF and ToS to 5 to put these packet into queues that have preferential treatment by Routers and Switches. You have to do this from stem to stern. As soon as you hit a router that you don't control it's game over.

[R1200CL]

3 minutes ago, plissken said:

QoS and DSCP are traffic markup technologies for priority traffic (delay sensitive, non-bursty) normally real-time. Audio playback isn't real-time so it's delay insensitive, and bursty in nature.

Nice. So we don’t need managed switches (for whatever they offers) when it comes to audio. 
 

Thanks for clarifying that. 

[plissken]

1 minute ago, R1200CL said:

You test has only a value in some very specific environment. And as you know there is in general accepted that a PC is the worst endpoint. With expensive exceptions. 

Maybe why you couldn’t hear the benefits of the etherRegen?

 

No I don't know that it's generally accepted that a PC is the worst end-point. In several ways I think it's the best:

 

1. I can do 10/25/50/80/100GB connections

2. I can easily do fiber

3. I can buffer entire albums

4. I can use SoX and convolution

5. I drive a 50" display with mine

6. I can do MCH Audio

 

[plissken]

1 minute ago, R1200CL said:

Nice. So we don’t need managed switches (for whatever they offers) when it comes to audio. 
 

Thanks for clarifying that. 

 

Never did for audio playback. You can still benefit from managed switches and you can get them for not a lot. My Cisco 2360 with four 10GB SFP+ was $60.

 

My IoT and security sit in their own vlans and are ACL'd off from the rest of the network. My Wireless has Mangement VLAN, House Data Vlan, Guest Vlan. My file server sits in it's own Vlan with appropriate ACL.

[plissken]

Also don't forget about wireless. It addresses all the 'Audiophile Switch' Gremlins.

 

My WiFi is two TP-Link AC1350's with the TP-Link Omada controller for 802.11k/r/v. I routinely get ~38MB/s. Way more than needed for PCM 24/192.

 

Things are $56 a pop and PoE. I put one in the laundry closet and another in an office closet.

[The Computer Audiophile]

On 9/13/2020 at 1:27 PM, plissken said:

You are talking about .4 dB spread over 146dB range.

I'm late to this party, but for many of the measurement nuts, it means something is better engineered because it has better numbers so it must be better. 

 

I look at it as a difference that was measurable at the output of a DAC, which the objective crowd says is impossible. We can see here that it actually is possible. 

 

If the objective crowd was really into science and discovery, they'd try to determine why/how this happened. 

[R1200CL]

@plissken

What kind of jitter is this. What causes it ? And what is an acceptable number ? Should we even bother with it ?

Speedtest.net explain like this:

Jitter: Jitter is a measure of the variability in ping over time. High jitter can result in buffering and other interruptions. Jitter is measured in milliseconds (ms).

And should my ping to external servers matter ?

 

 

[The Computer Audiophile]

3 minutes ago, R1200CL said:

@plissken

What kind of jitter is this. What causes it ? And what is an acceptable number ? Should we even bother with it ?

Speedtest.net explain like this:

Jitter: Jitter is a measure of the variability in ping over time. High jitter can result in buffering and other interruptions. Jitter is measured in milliseconds (ms).

And should my ping to external servers matter ?

 

 

I'm not @plissken, but I'll say this jitter doesn't matter at all. 

 

Edit: Doesn't matter for audio playback. If you're doing VOIP or video calls across the internet, then you may have issues if the jitter is too high. Your numbers aren't bad. My jitter usually measures 1ms at Speedtest.