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Can anyone help me in building a small music server?


Edoardo

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Hello, I would like to build a music server that...

 

...Runs Linux MPD

...fits in a library

...feeds my DAC through coax. SPDIF

...supports even 88.2 and 176.4 KHz, plug&play (possible?)

...can be "cliented" through wi-fi - like, with a netbook or an iPod as remotes.

 

 

I have found that tiny server in the Voyage MPD store, but it would work only with USB as I/O, whereas I would like to get an SPDIF out also if I upgraded from my netbook...

 

All the solutions I've found up to know are either too big or too expensive.

Can anyone suggest a small motherboard in which I could put an ESI [email protected] and a wi-fi card, in a small case?

 

Edoardo

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  • 3 months later...

is it possible to make MPD switch AUTOMATICALLY between sample rates (between 44.1, 48, 88.2 KHz) or do I have to edit the conf file each time?

 

Do not specify any rate in alsa section of mpd.conf and MPD (just like most other players in linux as well as win ASIO) will ask the soundcard to switch to the samplerate of the current audio source.

 

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The plug plugin will take care of all necessary conversions

 

wait... I would not like the stream to be converted, but to be sent to the DAC in the original sample rate.

 

so shall I substitute:

 

----------------------------------------------------

audio_output {

 

format "44100:16:2" # optional

 

----------------------------------------------------

 

 

with

 

 

----------------------------------------------------

audio_output {

 

format "*:24:2" # optional

 

 

----------------------------------------------------

 

 

?

 

Edoardo

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Just use the following. On the [email protected] or on USB it will autoswitch the sample rate and pad the file to the correct bit depth for the interface. The audio data is untouched. You can then enable or disable the output you are not using from the control interface on the client.

Demian

 

 

audio_output {

type "alsa"

name "Analog"

device "hw:0,0" # optional

mixer_device "default" # optional

mixer_control "Master" # optional

mixer_index "0" # optional

}

#

audio_output {

type "alsa"

name "Digital"

device "hw:0,1" # optional

}

#

audio_output {

type "alsa"

name "USB"

device "hw:1" # optional

mixer_device "hw:1" # optional

mixer_control "PCM" # optional

mixer_index "0" # optional

}

 

 

Demian Martin

auraliti http://www.auraliti.com

Constellation Audio http://www.constellationaudio.com

NuForce http://www.nuforce.com

Monster Cable http://www.monstercable.com

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I set up an HTPC using a Sapphire Tech AMD Brazos set up. Dirt cheap. I had to buy a SPDIF connector (£5 and the mobo has provision for one) as it only comes with an optical o/p. Realtek chip 24bit 192khz output and the on-board graphics does blu-ray and mkv. 6 Sata connectors and USB3. Not bad for £110.

 

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wait... I would not like the stream to be converted, but to be sent to the DAC in the original sample rate.

 

The plug plugin (plughw) provides the major following functions:

 

1. resample IF required, to the nearest samplerate supported by the card. As you see there is no need to worry about unnecessary samplerate conversion if your card supports the input samplerate natively

 

2. change the bit width IF required to fit that supported by the card. The plugin always tries to keep information, that is to use the largest bitwidth possible. This is actually important since e.g. Envy cards support only 32bit format.

 

3. change the number of channels IF required to fit the closest combination supported by the card. Although most multichannel cards support stereo mode too, it may not be the case for all. E.g. playing mono file, stereo on pure multichannel (the other added channels stay at zero), etc.

 

As you see, the plug plugin is bit-perfect for 24+bit cards supporting all common samplerates natively. In most cases it does only conversion to 32bit for many PCI cards.

 

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