Building a home music server: looking for feedback

[aamwgm]

Hi all,

First off, I've gained a lot of knowledge from this forum. So thanks for that.

I'd like to discuss the system I'm building: a custom "hifi" system that is built with second hand components to attain the best quality of sound with the least amount of money. Now, with that last point: I've spent a fair bit of money on the system already as I'll list.

Currently, this system has:

• A "server" (old desktop machine)
• A "DAC": a second hand Focusrite Scarlett Solo Gen. 3 acquired second hand for pennies connected via USB
• An Accuphase E-202 vintage integrated amplifier acquired second hand, connected to the DAC via RCA
• PeLeon Kantor S-4 speakers, also acquired second hand (by far the most expensive component in my system)
• Lyngdorf CD-2 CD player, also connected via RCA to the Accuphase
• Decent quality (no snake oil) speaker cables
• Cheaper RCA cables:
  • Scarlett to Accuphase
  • CD-2 to Accuphase

On the software side: the Linux machine runs Pipewire on top of the usual ALSA audio server. Pipewire's sample rate is set to 192 kHz. A web browser has one tab open with Qobuz. I can control the playback via my phone (Qobuz connect is pretty neat!).

One of the issues I've noticed so far is that when raising the volume to a high level on the Accuphase, I notice some noise. At normal listening levels, this noise is not noticeable unless you get very close to the speaker. I have some suspects contributing to this:

• The RCA cable (long run between the DAC and the amplifier), however, the noise is also present (but with different colouration) from the CD player
• The DAC's in the Scarlett & CD-2 (the latter would surprise me given its repute)
• The amplifier itself: but this is unlikely since the noise on unoccupied inputs is very, very low, which is testament to the quality of the otherwise vintage amplifier (those of you who've heard it probably know how nice it sounds)

---

Now, I have a few questions:

• Where are the weak links of this system?
• What is the likely culprit behind the noise? Do I need better RCA cables?
• Do I gain anything in replacing the Scarlett with a DAC with balanced out and only converting to unbalanced when feeding the amplifier at the very end of the chain?
• Would you have done anything differently in setting up this system?

Thanks!

 

[fatoldgit]

Unrelated to this noise issue, you state:

On the software side: the Linux machine runs Pipewire on top of the usual ALSA audio server. Pipewire's sample rate is set to 192 kHz. A web browser has one tab open with Qobuz. I can control the playback via my phone (Qobuz connect is pretty neat!).

So that means you are using a fairly "run of the mill" resampling process (that which comes with the Linux OS).


You don't state what playback software you use... unless it's just a web browser and/or your phone for control?

MIght be worth looking at some software that can utilize some decent resampling. Something like LMS would give you Qobuz and decent resampling.

 

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[JeffS7444]

If you haven't already implemented some form of room correction, that's the next obvious step, IMO.

 

[aamwgm]

So that means you are using a fairly "run of the mill" resampling process (that which comes with the Linux OS).

The resampling quality in Pipewire is set to max (10). AFAIK, the default sampling algorithm used is Spa, which is pretty well regarded. Also, most of the time, Pipewire should be upsampling since most content I've seen on Qobuz has been in the 48-96 kHz range.

You don't state what playback software you use... unless it's just a web browser and/or your phone for control?

Indeed, I just use the browser (Firefox). pw-top also tells me which sampling rates are being used in the chain along with the PCM value. This seems to check out.
When playing 48 kHz 24 bit content on Qobuz, I see that these are the values set (dynamically).

MIght be worth looking at some software that can utilize some decent resampling. Something like LMS would give you Qobuz and decent resampling.

OK, thanks. Do you have any links handy where I can read more about LMS?

---

If you haven't already implemented some form of room correction, that's the next obvious step, IMO.

Yeah, this is what I've been afraid of doing (especially given the WAF).
The speakers are placed right at the end of a large carpet which provides some decent damping (I've recorded clap test samples which I've analysed in a DAW for the tail, the one without the carpet has a noticeably longer tail, given this test is rudimentary at best).
What would your steps towards improving that be?
The speakers are rear ported and are pretty close to a wall (one of them is in a corner).
I've been considering small bass traps that I can hide without really having my wife notice them too much.

 

[Slot]

OK, thanks. Do you have any links handy where I can read more about LMS?

Have a look here: Lyrion Music Server

And there's a really helpful forum here: Lyrion Forums

 

[Honken]

Yeah, this is what I've been afraid of doing (especially given the WAF).
The speakers are placed right at the end of a large carpet which provides some decent damping (I've recorded clap test samples which I've analysed in a DAW for the tail, the one without the carpet has a noticeably longer tail, given this test is rudimentary at best).
What would your steps towards improving that be?
The speakers are rear ported and are pretty close to a wall (one of them is in a corner).
I've been considering small bass traps that I can hide without really having my wife notice them too much.

Well, with room correction JeffS7444 probably meant EQ rather than room treatment. Fortunately, EQ is easier on the eyes than acoustic panels and diffusers. You just need a measurment mic and REW, no permanent installs required.

If you want to continue to use Pipewire for this setup, importing filters from REW for room correction EQ would be a breeze. Whilst the parametric equalizer module is designed for for squig.link use with IEMs and headphones, it makes implementing a global EQ a breeze. You can read more about it here:

PipeWire: Parametric-Equalizer

Buy a mic and measure your room if you haven't already.

--

Personally, I use the parametric equalizer module of Pipewire on my desktop - where I use it with headphones and IEMs, but I use CamillaDSP and an ALSA loopback device for my living room stereo system. The Pipewire module I find "just works" in a desktop setting where a multitude of simultaneous sounds might have to be mixed together at any given time, whereas the CamillaDSP approach gives me a lot of flexibility for a dedicated playback system, handling at the most one source of playback at once.

Importing filters from REW into CamillaDSP is also a breeze.

 

[aamwgm]

Have a look here: Lyrion Music Server

Thank you.

Buy a mic and measure your room if you haven't already.

Great, this gives a head-start.

 

[phofman]

One of the issues I've noticed so far is that when raising the volume to a high level on the Accuphase, I notice some noise.

PC with PE wire -> USB -> RCA shield being audio return path -> amp with PE wire - IIUC you have a ground loop. It may or may not produce a noise generated in PC ground traces.

The noise is also present (but with different colouration) from the CD player

If that noise in CD playback is present when the PC is running:
PC with PE wire -> USB -> RCA shield -> amp -> RCA shield being audio return path -> CD player with PE wire - again a ground loop, but the loopback currents would be reduced by the amp PE wire path.

The CD playback hypothesis would be easy to test - just disconnect the RCA from the PC/USB soundcard when playing CD.

Just 2 cents..

 

[aamwgm]

The CD playback hypothesis would be easy to test - just disconnect the RCA from the PC/USB soundcard when playing CD.

Thanks for the reply, btw. I just tried this.
I disconnected the RCA from the audio interface, changed the input to the CD player (using the Tuner input on the Accuphase), and increased the volume. The noise was present.
This happens at very high volume levels.
And the noise is only present on occupied inputs. It's incredible that the unoccupied inputs have almost 0 noise compared to the occupied ones on a 50 year old integrated.

 

[Sal1950]

The resampling quality in Pipewire is set to max (10). AFAIK, the default sampling algorithm used is Spa, which is pretty well regarded. Also, most of the time, Pipewire should be upsampling since most content I've seen on Qobuz has been in the 48-96 kHz range.

I'm also a Linux user, "PCLinuxOS" for around 25 years now, but by far not a command line pro.
I use Strawberry to serve the music files bit-perfectly to Alsa
I'd like to throw out a couple questions and comments at this point.
Firstly what do you hope to gain from the upsampling your doing ??
IMHO there's nothing to be gained that way, you can't add any musical information to a file that isn't already there, only possibly add something that don't belong there in the first place. The file's never going to sound better than by delivering the original file in a bit-perfect manner to your DAC.
JMHO

 

[aamwgm]

Firstly what do you hope to gain from the upsampling your doing ??
IMHO there's nothing to be gained that way, you can't add any musical information to a file that isn't already there, only possibly add something that don't belong there in the first place.

Indeed and I’m aware of that. The higher sample rate is simply because pipewire doesn’t dynamically adjust it based on the content. Since my Qobuz settings allow for up to 192kHz I’ve just set it to that rete.
Upsampling is harmless AFAIK.

 

[HoweSound]

I've built several music servers using mini-itx motherboards as fanless pc's with custom cases and Linux os and Kodi media software. Also have tried Raspberry Pi setups. My music library is all lossless flac. All my systems are now relegated to the storage closet; replaced by my Eversolo A6 with my library on ssd and music streaming courtesy of Qobuz. I think this is now a solved problem for most users. There are other streamers; including some with more features and pre-amp functions including some from Eversolo, but, the A6 provides everything I need and I doubt that I could tell the difference between the internal dac and another high end add-on external dac.

 

[aamwgm]

I've built several music servers using mini-itx motherboards as fanless pc's with custom cases and Linux os and Kodi media software. Also have tried Raspberry Pi setups. My music library is all lossless flac. All my systems are now relegated to the storage closet; replaced by my Eversolo A6 with my library on ssd and music streaming courtesy of Qobuz. I think this is now a solved problem for most users. There are other streamers; including some with more features and pre-amp functions including some from Eversolo, but, the A6 provides everything I need and I doubt that I could tell the difference between the internal dac and another high end add-on external dac.

I understand the appeal of a "no-nonsense" off-the-shelf streamer. But for me, part of the fun is to build one yourself.

I also use NixOS: so the system is reproducible down to the finest configuration. Upgrades are painless and declarative. It is very satisfying to have something you've built yourself rather than being beholden to a third party that may or may not decide to make you upgrade to their latest model.

I must also say that so far, the audio quality I'm getting out of this system is pretty great, I have no complaints. The tiny amounts of noise (only audible at un-listenable volume levels, or by holding your ears next to the tweeters) are present in every system, AFAIK.

The bottleneck is not the software, I think, but the vintage E-202 whose THD levels fall way short of modern alternatives. I think adding a more modern amplifier with balanced circuitry (I'm eyeing the Accuphase E-350 off the used market) might even further improve on the noise front and make this system competent up to modern standards. And that would be endgame for me.

 

[boxerfan88]

The RCA interconnect noise seems like the main issue you're highlighting.
For not much money, maybe upgrade the RCA interconnect -- fabricate a set using good RCA connectors with shielded cable like Belden RG6 family.

 

[antcollinet]

Thanks for the reply, btw. I just tried this.
I disconnected the RCA from the audio interface, changed the input to the CD player (using the Tuner input on the Accuphase), and increased the volume. The noise was present.
This happens at very high volume levels.
And the noise is only present on occupied inputs. It's incredible that the unoccupied inputs have almost 0 noise compared to the occupied ones on a 50 year old integrated.

The noise is almost certainly the result of ground loops resulting in common mode noise.

This is a common problem with unbalanced (RCA) inputs, especially with longer interconnects. You are unlikely to solve it with better cables; you need to look at your grounding and interconnect arrangements. Your amp also has very high gain (around 45dB), which will make things worse. Any noise just gets amplified more.

Ideally, your analogue sources should be right next to your amp with the shortest possible interconnect. As pointed out - getting low impedance ground bonding between all analogue devices might help (but don't create any additional loops while doing it, or things might get worse) - as might ensuring they are all plugged into the same supply socket.

It is very much better to place your amp next to the sources and have long speaker wires than it is to have the amp close to the speakers and have long unbalanced interconnects.

In the end, though - if you can't hear the noise from the main listening position at normal volume levels - then it is not a sufficient problem (IMO) to be worth spending lots of time on.