A novel way to massively improve the SQ of computer audio streaming

[cat6man]

I've visited this thread on and off over its life and I'm back for another visit.

 

I have a dx-usb HD with sCLK-2224 (formerly superclock upgrade) which I used for a few years, then put on the shelf while I experimented with microRendu and then ultraRendu, with LPS1 and LPS1.2 power supplies.

I recently went back to using the dx-usb HD to see how my totalDAC sounded with aes/ebu, since it had been a while since I used that mode (previously I used dx-usbHD with the sotm battery supply).

 

My current setup is ultraRendu (w/ LPS1.2 power) feeding usb into dx-usbHD (w/LPS1 power), aes/ebu into totaldac.

The sound is very very good and I plan to compare it to the ultraRendu feeding the dac via usb.

I also want to try swapping the LPS1 and LPS1.2, putting the LPS1.2 on the dx-usbHD feeding the DAC.

 

I see now, and wasn't paying sufficient attention earlier, that the newer EX clock fits in the dx-usbHD and is a good upgrade even from the sCLK-2224.  How many folks out there have a sotm unit withe the EX clock?  Anyone using the dx-usbHD with sCLK-13224EX clock?

Anyone doing so with aes/ebu output?  I'd appreciate any feedback or suggestions.  I see that Romaz had a unit upgraded and was using, I believe, spdif output.  Anyone else with experience here?

 

I've emailed sotm asking about an upgrade for my dx-usbHD unit, but have yet to get a response.

I do see that the EX clock is an option that can be chosen at checkout if buying a new dx-usbHD unit and the sCLK-13224EX clock costs $250 more than the sCLK-2224 (which I currently have).

 

Cheers

[cat6man]

On 7/5/2018 at 3:06 AM, jean-michel6 said:

I do had a dx-USB HD with sCLK 2224 and I did upgraded it to sCLKex.  

It is now powered by an lps 1.2

My dealer here in france bought back from me at half cost the sCLk224 board and i did purchase the sCLKex. 

I am using it to input into AES-EBU in an AMR dp 777 se dac. 

I did got some SQ improvement going from sCLK 2224 to sCLKex but it is not very big. 

In my opinion it is a worthwhile investment if you plan to use the two additional clock output of the sCLKex and adding later a 10mhz ref clock which I have not done yet. 

One other route you may try. 

I did try in my system the new matrix USB-Spdif converter.  

It is better than the sotm with sCLKex board and cost is small .... cheaper than sCLK ex 

 

Merci Jean-Michel,

 

The matrix x-spdif 2 looks like an interesting option for me to try, as you say that it outperformed the dx-usbHD with the sCLKex clock and is cheaper.  I assume you were comparing the aes/ebu outputs of the two and both were powered by the lps1.2 (also what I use for  the dx-usbHD).  You noted that the change from the sCLK 2224 to sCLKex was small.  How would you describe the improvement going from sCLKex to matrix x-spdif2?

[cat6man]

On 7/7/2018 at 5:49 PM, jean-michel6 said:

Yes I was comparing both unit on their aes/ebu output and both were powered by lps 1 . I did receive my lps 1.2after I made this comparison.  

 

The matrix was was more transparent than the sotm and medium high and top end was smoother .Overall it was nicer and sounding more analogic. Soundstage was also improved. 

 

For the time being I still run with the sotm because I am using the 2 extra clock of the clockboard and I want also to test the external 10mhz ref clock. 

But in the near future I intend to move to the matrix. 

 

My matrix arrived today (ordered last friday) and I'll be testing/comparing tomorrow.

It is 'warming up' on a generic 9v supply overnight, then I'll switch it in and out for the sotm using my lps1.2.

I have a lps1.0 on my ultraRendu and may try swapping the two power supplies, but I assume the better supply should

be closest to the DAC.

 

Thanks for the recommendation!

[cat6man]

will audiolinux and hqplayer work with just 4G of RAM? (I currently have a single 8G sodimm)

[cat6man]

2 hours ago, Johnseye said:

 

Yes

 

great.

i will have to try the Apacer 4Gig ram under discussion above.

 

update:  ordered from Mouser, delivery Wednesday

[cat6man]

20 minutes ago, ted_b said:

Sorry if I didn't explain my point.  I realize that an endpoint can be a renderer  But it can also be a rather simplified FIFO buffer, as in Jussi's NAA architecture.  I am asking whether you (or Romaz, in this example) believe these different functions would have different cpu, power, noise profiles....or is low-power-low-noise no longer the perfect endpoint box in ANY circumstance?

 

great question.

running hqplayer and having an NAA architecture at the endpoint, does increased processing power still sound better?

 

[cat6man]

3 minutes ago, matthias said:

 

Maybe you find the answer here:

 

Matt

 

if i recall, romaz was not using hqplayer/NAA so i'm not sure his conclusions re: processing power map into that case...........they might, but i don't see evidence yet.

[cat6man]

13 hours ago, bodiebill said:

 

For my audio endpoint I made new measurements on the ATX part as I found a 19V SMPS for it. Now the comparison is more fair as I used SMPS'es for both ATX and EPS/CPU. The result is:
 

 

	ATX		EPS/CPU
maximum	W	A	W	A
startup cycle	19.1	0.08	15	0.07
idle	16.7	0.07	4.5	0.02
playing music	16.9	0.07	7.4	0.03
stop cycle	17	0.07	7.8	0.03

 

Power draw at the ATX side (via HDPLEx 800W DC-ATX rail 1) is even lower than before when measured with the LPSU, but still higher than at the EPS side.

 

Note that the W & A values when playing music are independent of the audio format (redbook, hi-res PCM or DSD256). On the server there is a huge difference (as I can see from the temperatures) but for the endpoint, which just streams and does not process, there seems to be no difference.

 

These figures will save me some money 🙂

 

followup question for superdad and/or swenson

 

if we're looking for micro-bursts of current that are instantaneously pulling much more than the average current draw, would we be able to see this on a high speed oscilloscope in the time domain as a bursty dip in the voltage?  if the peak current was within the capacity of the power supply and the impedance was low (a fuzzy term, i know), i would expect the voltage to stay constant.

however, if the instantaneous draw is limited somewhere, wouldn't we see an instantaneous voltage change?  (assuming we had a fully buffered way of accessing the voltage)

 

just wondering..............or maybe its already in the white paper?

[cat6man]

9 hours ago, chrille said:

Have you actually heard your system against Roy's most recent ?

Or are you just envious that he seems to have reached Moksha?

 

 Regarding Mokhsa and music listening my current  take on it, after a  great week of downhill-skiing during which a good friend gave me 6 gigabytes of mainly Indian and  other world music, as 16/44.1 wave files, is that in particular Anoushka Shankar's album RISE brings me at least close to Moksha on a daily basis.

 

PURE BLISS via my quite big  electrostatic speakers ,my 900 watts per channel amp and mbp via usb and Qutest/HMS/Audirvana.

I really wonder if I could be even more amazed and thrilled by her fantastic music than my relatively  humble system already delivers?

And its freaking rbcd not even hi res!

 

For anyone who wants to explore a bit  beyond their Western Pop/Rock Comfort Zone I can very heartily recommend Anoushka Shankar's music.

She records among other labels also for DGG and there are some interesting Youtube videos of her playing LIVE at Music Festivals as well.

RISE is just one album among  several others she has recorded that is imho capable of giving  at least ME,a non drug induced REAL HIGH equalled by few other musicians, including her half sister Norah Jones whose music more people are probably a lot more  familair with.

Norah's music is nice and  very good at its best imho.

 

But Anoushka's is on a much more advanced level musically

Moksha music.

Oops  I almost forgot to mention that she plays the Indian instrument Sitar. And one comment about that instrument I've read said." Sitar is an instrument smuggled through Heaven"

I'd like to add, here it is played by a Godess

Cheers Chrille

 

@chrille

first, thanks for Anoushka Shankar recommendation!

 

@all

one thing i see that seems to differ between the diy and extreme, that perhaps warrants investigation, is the impact of storing the music in (lots and lots of) optane versus streaming from a NAS or using a hard drive conected via usb to the server.

any thoughts folks?  has anyone compared internal storage with external storage on the extreme?

 

 

[cat6man]

On 5/31/2020 at 1:17 PM, CJH said:

Tried the Slimrun USB3 and am getting very good results vs. an uRendu powered by double regulated MPaudio. Only problem is when using HQPlayer it sounds great until I use my Chromebook (via wifi) to control HQP (2 different rooms). When the HQP control app is used the soundstage shortens and high frequencies are exagerated. Turn it off and the magic returns. The Slimrun has an option to be externally 5v powered so I will try this as a remedy--problem is power connection is via micro usb so I am waiting for appropriate cable connectors to arrive. Anyone have similar experience?

CJH

 

quite strange but interesting.

some more details would be good (what are you running hqplayer on, wired/wifi, music on nas or on hqp server, etc)

is NAA on uRendu?

 

if i read this correctly, you have compared case 1 with either 2a or 2b (below)

 

1) hqplayer desktop sounds great

vs.

2a) hqplayer desktop set to remote control option (using what app for control?) sounds bad

or

2b) hqplayer embedded controlled by remote app sounds bad

[cat6man]

20 hours ago, lmitche said:

Good idea.

 

With two Monoprice USB Slimruns here, SQ has taken another jump. I don't use them for the USB DAC connection. Instead they connect a 3.5 inch HDD in a USB 3.0 enclosure and a Tplink USB 3.0  NIC to two CPU direct USB3 ports on my server. It is a tight fit. My two are each powered with a dual serial LT3045 chain of 1 amp regulators.

 

On well known tracks I'm hearing more detail, openness, clarity and depth. Image density has increased again. I've had to lower the volume by 2 db. LOL. The Monoprice cables deliver.

 

Tomorrow, more experiments on power options.

 

Larry

 

Larry,

 

so, you're running

 

external hdd ==> wired usb 3.0 ==> slimrun female ==> slimrun male (powered lt3045) ==> server

[cat6man]

2 hours ago, lmitche said:

Close:

 

external hdd ==> usb 3.0 male type A to male type B adapter==> slimrun female ==> slimrun male (powered lt3045) ==> server

 

 

got it.

thanks......now if only the slimrun were not all sold out.......

[cat6man]

On 6/3/2020 at 7:53 PM, CJH said:

Following up on my previous post. I received mini USB adapter to power my MP SR with a double regulated MP Audio 5V supply. This completely solved my sonic problems running HQP Control from a second Laptop. Sound is now consistent with or without HQP Control. Experiencing improvements in bass control, depth and width of soundstage, spacing between instruments and tonality resulting in a more natural, less mechanical presentation. One surprise was how good upsampled PCM now sounds, though sampling to DSD 256 still edges it out. One of my favorite test tracks is Zigeunerweisen, from Uncommon Ritual--Edgar Meyer.

 

since the MP SR are currently classified as unobtainium, i've ordered an alternate version which does not permit replacing the Vbus power via micro connector (ala monoprice slimrun).

 

has anyone found an adapter for usb A female to usb A male that includes the ability to insert the Vbus power?

i've searched/googled with no success.

[cat6man]

5 hours ago, MagnusH said:

This one is what you want I think: https://ifi-audio.com/products/idefender3-0/

 

thank you muchly!

just ordered.

[cat6man]

regarding audio-linux v. euphony

 

i see that many folks have moved from audio-linux to euphony over the past months with reports that euphony sounds better...............so i got a trial of euphony to check it out...........and the results are very interesting.

 

common baseline:

running HQPplayer embedded in server to NAA running on opticalRendu, much network optimization.

HQP running pcm only, no upscaling, no dither, just bit exact (theory being that having renderer at server and minimal processing at NAA is a good thing to do).  music on HDD attached via usb to NUC running music server

 

audio-linux running in ramroot, realtime set to extreme

euphony running on usb stick (in ramroot, it doesn't see HDD, so couldn't test ramroot)

 

tests:

audio-linux 2.5 -- meh

euphony 20200114v3 -- meh

audio-linux 1.4 -- wow!

 

in my system, the old build of audio-linux is vastly superior to the newer builds of both audio-linux and euphony and it isn't close.  i have not chosen to compare the first two as who cares if mehA is superior to mehB when you have wow as an option.

 

since YMMV, i'd love to hear from other audio-linux and euphony users.  try comparing your current servers with the old audio-linux 1.4 and let us know if you agree/disagree

 

something is going on here but damned if i know what............scheduling/queuing algorithms in the realtime scheduler, arch-linux build variations/issues, ?????.....................but whatever it is, it isn't minor.

i can't listen to the first two for more than a few minutes before the difference is obvious and i'm back to the audio-linux 1.4 build.

 

stay safe out there!

 

p.s. will be upgrading music server from hdplex200 to hdplex300 this week

 

[cat6man]

3 hours ago, drjimwillie said:

Did you see that AudioLinux had posted the wrong 2.5 they had posted the 2.5 for pink faun.    now they have posted the correct 2.5

 

yup..........a non-issue really since the 'wrong' one wouldn't have loaded anyway

[cat6man]

43 minutes ago, ted_b said:

Great post, but this would be even more relevant in the "Shootout at the Linux Corral: Audiolinux vs Euphony" thread, no?

 

 

excellent point...........off there i shall go

[cat6man]

1 hour ago, edwardsean said:

 

 

The Monoprice slimrun is hard to find. 

 

 

 

just recently back in stock.

got mine yesterday.

[cat6man]

2 hours ago, edwardsean said:

Did you get it from Monoprice?

 

Also, do you know if this works with USB 2.0 devices? 

 

yes

it should

 

will install here tomorrow

[cat6man]

may not be your problem but i found it needs to be plugged in deeply on some devices.........insufficient contact observed here until i pushed it in further on the NUC

[cat6man]

1 hour ago, ray-dude said:

 

I'm in the same boat.  I hear huge benefit under DACs and power supplies (and I'm comfortable that I at least have a hypothesis why), but I don't have a working theory why there would be an impact with transducers (not saying that there isn't!)

 

The physical displacement distances would have a vanishingly small impact on phase alignment. The only other thing I can think of is draining unwanted resonances from the case work, but that would be hugely speaker/room dependent.  This is an intriguing area to explore and understand.

 

i wonder if the benefits heard under DAC and power supply are also observed with headphones, as well as with your speakers?

 

this would give us an indication of whether the mechanism is immunity from the outside (i.e. the speakers setting the room ablaze with impulses) or removing inherent resonances/etc that originate in the electronics

[cat6man]

On 4/8/2019 at 11:05 PM, austinpop said:

So far, I have tried

1. -b 2097152:2097152 -a 52428800:4::
2. -b 1048576:1048576 -a 26214400:4::
3. -b 524288:524288 -a 13107200:4::
4. defaults (nothing in the Euphony expert field)

 

Only 4. is glitch free. 

 

So the answer lies beyond.

 

 

 

resurrecting a path that's a bit old.

i have audio-linux running on a nuc, streaming to an opticalRendu

i've been using hqplayerd for a few years now with all processing turned off and found that moving the renderer off of the Rendu (just NAA) made a nice improvement, though most users use hqplayerd for upsampling/etc.

 

i recently upgraded my networking (ubitquiti edgerouterX to isolate streams related to audio) and power supplies all over the place (plus many many fewer wall warts)

so i thought i would revisit some previous assumptions.

a long time ago, i started streaming with logitech music server, on a ubuntu based nuc, streaming to a logitech touch with digital usb out to my DAC.......come a long way since then

 

previous, but recent, conclusions:

hqplayerd in ramroot on AL is really good with "older" AL builds (e.g. 140) and this sounds much better than the same configuration on build 250........who knows why

 

so, i decided to go back and revisit logitech/squeezelite and running on build 250 was meh running on USB but was outstanding running in ramroot, definitely competitive with hqplayerd on 140 (i need a lot more time to do more detailed comparisons).......hmm, what is going on here?  now my old experience with squeezelite had been without AL or ramroot, but together this is a contender.

 

next questions: 

would squeezelite on AL build 140 be even better?  i don't know as i'm having trouble getting the LMS server to install.........stay tuned

 

the above observations was made with the default buffering for squeezelite.

i next tried -b 2097152:2097152 -a 52428800:4:: in the config file and it sounded good (but different).....again, i apologize for not having more detailed observations yet...........i had expected, based on the above, to have glitches or other problems but i have seen none whatsoever.

 

therefore, why not keep increasing the buffers further and see where that road leads?

i have 8G ram so i'm going to try doubling the ~2G input/output buffers and do some serious listening comparisons.

 

one thing i don't understand is where these buffers reside? are the input/output buffers (parameter -b) on the NUC?

what about the 52G -a buffer?  is that on the opticalRendu?  as a comparison i have set the opticalRendu buffer for hqplayer to 500ms and this has improved the sound. 

 

i also assume that if i had a nuc instead of the opticalRendu, i would have much more flexibility to buffer at the renderer side.

 

cheers (and stay safe everyone)

 

p.s. i've ended my euphony experiment after much wasted time and frustration trying to get external disk music played from memory.......in general i found the design/layout confusing.........my trial ran out as i ran out of patience

 

 

[cat6man]

1 hour ago, austinpop said:

 

Just to clarify: the glitches occurred when using Roon Core as the server, and SL as the endpoint. I had no issues — even with these large buffer settings — when the server was LMS. I hope that distinction is clearer now?

 

Thanks for your findings. Bummer that Euphony OS did not work out for you, as I have had very good experience with it. But ultimately, the responsibility is the Euphony team's, to ensure people — especially potential customers like you — get adequate support.

 

thanks for the clarification. 

 

do you have any suggestions as to buffer size without roon core?

did you ever compare root core server with LMS?

 

and to clarify on my part--i never requested support from euphony so they did not fail to support me.

my view is that stuff should work (it's the engineer in me--'we don't need no stinking manuals')

it didn't, and i found the interface very confusing

 

anyone know where these buffers reside in h/w or how large they can be made?

[cat6man]

3 hours ago, austinpop said:

 

The SL buffer sizes that I liked were the same, irrespective of which server (Roon Core or LMS).

 

 

Simple answer: in RAM.

 

Or more accurately, these buffers refer to blocks of virtual memory, in the address space of the "squeezelite" process, which are allocated by the application. If you search back, you'll find an earlier post of mine that gave some more detail: https://audiophilestyle.com/forums/topic/30376-a-novel-way-to-massively-improve-the-sq-of-computer-audio-streaming/?do=findComment&comment=946503

 

 

One thing I should add is that the definition of -a is different based on OS. The post above refers to the implementation on Linux, where -a refers to a buffer while interacting with the ALSA driver. As Emile recently reminded me, this is different on Windows, where -a refers to the buffer size to the sound driver, expressed as a time unit (in ms).

 

thanks again, i'm sorry i'm not being sufficiently clear with my question.

i've seen the earlier post.

 

what i mean to ask is, 'where are the buffers located, at the server or at the endpoint?'

 

both the -b and -a buffers are set in the sqeezelite.conf file which resides on the server.

are either of those buffers implemented at the endpoint (the opticalRendu in my case).

 

the Rendu has the following squeezelite settings available:

Buffer size (buffer time in ms)

Period count (period count in bytes)

 

since these look like the -a parameters, i'm wondering if they are the same as the -a parameters set in the config file

 

as a comparison, for hqplayer there is a buffer setting (in ms) that is implemented in the NAA (i.e. the Rendu)

[cat6man]

8 hours ago, austinpop said:

 

No worries. Squeezelite (SL) runs on the machine that is connected to the DAC. If you run a distributed configuration, where the server and endpoint are on distinct machines, then it would be on the endpoint.

 

 

The squeezelite.conf is irrelevant on the server if your DAC is connected to an endpoint machine. You'd need to change the sqeezelite.conf file on the endpoint.

 

 

The experiments I and Ray and others ran were on NUCs and other machines with ample RAM (see next point) running AL and Euphony OS, where it's possible to fully control the command line parameters with which SL is invoked. I'm not aware that oR gives you root access to a shell to allow this, or any ability to edit and change the sqeezelite.conf file. Also, to really exploit large buffers, like the 2GB each for input stream and output, you need to have at least 8GB of RAM. Does the oR have that much?

 

 

You'd have to check with Sonore to see if these allow you to achieve the same effect as the -b and -a options. You could ask them to expose these options as additional settings, I suppose.

 

 

sonore does not expose the config file, but do provide web access to a squeezelite settings page on the Rendu where you can change what look like the -a parameters.  they recommend not changing the (unknown value) default.

as for buffer size possible, i'm sure it is far smaller than what you guys did with a NUC as the endpoint........now i have that to consider, hmmmm..................has anyone done a direct comparison of a Rendu vs. a NUC endpoint, using a separate server box?

 

as for Rendu buffer available, if I can do 500ms of buffering in the Rendu/NAA with hqplayer, that would suggest for red book

(44.1k) x (16 bits) x (2channels) x (1/2s) x (1byte/8bits) = 88.2KBytes which is more than an order of magnitude smaller