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How to setup a bit-perfect digital audio streaming client with free software (with LTSP and MPD)


ronalde

Inspired by Bit-perfect audio in Linux at 88.2 and 176.4 now possible I created a setup at home which I share at How to setup a bit-perfect digital audio streaming client with free software (with LTSP and MPD).

 

In short, my wanted setup had the following requirements:

 

 

  • having bit-perfect audio for all digital formats accepted by my DAC
  • having the freedom of only using free software
  • having the freedom of only using open non patent encumbered file formats
  • having the freedom to use any client device as a streaming digital audio device
  • having the freedom to use any device for controlling the streaming digital audio device (acting like a “media controller”)
     

 

 

To achieve these goals I’ve made the following choices:

 

 

  • create a LTSP environment using my desktop PC as the LTSP server
  • installing a MPD client on the desktop PC or Android client acting as a media controller
  • using an old HP t5725 thin client as the LTSP client
  • this thin client will also act as a (headless) MPD server

 

 

The result is whopping! I can plug in any PXE-capable PC or laptop to my LAN and connect it to my DAC with USB. After booting it, 15 seconds later it is a bit perfect audio streaming client. I can use any PC, laptop or PDA to control the media and browse through my playlists and music library.

 

 

 

Have fun!

7 Comments


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Ronalde, thank you for the extensive write-up. Cool setup.

 

 

 

Much to my dismay, there seems to be more to digital sound quality than bit-perfection. Have you experimented with a real-time (low-latency) Linux setup? It does indeed improve the sound. This does not make sense to me or anyone else I know. I wouldn't have believed it if I hadn't heard it with my own ears and I don't expect anyone else to either.

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Much as I usually enjoy arguing with you, in this we are in perfect agreement mate. There is more to digital audio than ones and zeros. Much more. Makes it a lot of fun though. :)

 

 

 

Paul

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We've always agreed there's more to it than 1's and 0's. I hypothesized EMI and you "timing errors". I took exception to your explanation since it didn't explain how or why the sound was affected.

 

 

 

These days I'm suspecting there is more to it than EMI. Low latency seems to positively affect the sound, but how and why? :)

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Wish I knew - Damien from Auraliti has volunteered to do some testing if anyone can figure out exactly what needs to be tested and.or measured. It's in a thread about why high quality players sound different.

 

 

 

By the way, what do you think latency is, save timing? :)

 

 

 

-Paul

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Thanks for your reply.

 

 

 

Back in 2005, I used the preemptive Linux kernel patches (see Low latency for audio applications) together with Jack and alsa. Unfortunately Jack uses one (configurable) fixed sample rate and bit depth, so it was far from my goals.

 

 

 

I agree that every aspect in the system influences the sound quality. This system is big and complex because all computer devices are physically interconnected through ethernet. Furthermore the audio equipment and (switching) power supplies of the computer devices are interconnected through the power circuit in (and beyond) my house.

 

The designer of my Pink Faun USB-DAC even notices in his white paper that the chosen USB-port influences the resulting sound quality.

 

 

 

In what way latency adds to the stack is unclear to me for my purpose (music listening- not producing it). I guess that the asynchronous design of my DAC defeats this issue (for me).

 

 

 

Please keep me informed about progress you make regarding this issue.

 

 

 

Regards,

 

Ronald

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Hi Ronald. I use an asynchronous USB DAC too. It's a Wavelength Proton. Unfortunately, a low-latency real-time config improves the sound.

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