[nyc_paramedic]

Here it is:


# An example configuration file for MPD
# See the mpd.conf man page for a more detailed description of each parameter.


# Files and directories #######################################################
#
# This setting controls the top directory which MPD will search to discover the
# available audio files and add them to the daemon's online database. This
# setting defaults to the XDG directory, otherwise the music directory will be
# be disabled and audio files will only be accepted over ipc socket (using
# file:// protocol) or streaming files over an accepted protocol.
#
music_directory      "/mnt/tunes"
#
# This setting sets the MPD internal playlist directory. The purpose of this
# directory is storage for playlists created by MPD. The server will use
# playlist files not created by the server but only if they are in the MPD
# format. This setting defaults to playlist saving being disabled.
#
playlist_directory      "/mnt/tunes/mpd/playlists"
#
# This setting sets the location of the MPD database. This file is used to
# load the database at server start up and store the database while the
# server is not up. This setting defaults to disabled which will allow
# MPD to accept files over ipc socket (using file:// protocol) or streaming
# files over an accepted protocol.
#
db_file         "/mnt/tunes/mpd/tag_cache"
#
# These settings are the locations for the daemon log files for the daemon.
# These logs are great for troubleshooting, depending on your log_level
# settings.
#
# The special value "syslog" makes MPD use the local syslog daemon. This
# setting defaults to logging to syslog, otherwise logging is disabled.
#
log_file         "/mnt/tunes/mpd/mpd.log"
#
# This setting sets the location of the file which stores the process ID
# for use of mpd --kill and some init scripts. This setting is disabled by
# default and the pid file will not be stored.
#
pid_file         "/mnt/tunes/mpd/pid"
#
# This setting sets the location of the file which contains information about
# most variables to get MPD back into the same general shape it was in before
# it was brought down. This setting is disabled by default and the server
# state will be reset on server start up.
#
state_file         "/mnt/tunes/mpd/state"
#
###############################################################################


# General music daemon options ################################################
#
# This setting specifies the user that MPD will run as. MPD should never run as
# root and you may use this setting to make MPD change its user ID after
# initialization. This setting is disabled by default and MPD is run as the
# current user.
#
user            "root"
#
# This setting sets the address for the daemon to listen on. Careful attention
# should be paid if this is assigned to anything other then the default, any.
# This setting can deny access to control of the daemon.
#
# For network
bind_to_address      "any"
#
# And for Unix Socket
#bind_to_address      "/var/run/mpd/socket"
#
# This setting is the TCP port that is desired for the daemon to get assigned
# to.
#
port            "6600"
#
# This setting controls the type of information which is logged. Available
# setting arguments are "default", "secure" or "verbose". The "verbose" setting
# argument is recommended for troubleshooting, though can quickly stretch
# available resources on limited hardware storage.
#
#log_level         "default"
#
# If you have a problem with your MP3s ending abruptly it is recommended that
# you set this argument to "no" to attempt to fix the problem. If this solves
# the problem, it is highly recommended to fix the MP3 files with vbrfix
# (available from <http://www.willwap.co.uk/Programs/vbrfix.php>), at which
# point gapless MP3 playback can be enabled.
#
#gapless_mp3_playback         "yes"
#
# This setting enables MPD to create playlists in a format usable by other
# music players.
#
#save_absolute_paths_in_playlists   "no"
#
# This setting defines a list of tag types that will be extracted during the
# audio file discovery process. Optionally, 'comment' can be added to this
# list.
#
#metadata_to_use   "artist,album,title,track,name,genre,date,composer,performer,disc"
#
###############################################################################


# Symbolic link behavior ######################################################
#
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links outside of the configured music_directory.
#
#follow_outside_symlinks   "yes"
#
# If this setting is set to "yes", MPD will discover audio files by following
# symbolic links inside of the configured music_directory.
#
#follow_inside_symlinks      "yes"
#
###############################################################################


# Zeroconf / Avahi Service Discovery ##########################################
#
# If this setting is set to "yes", service information will be published with
# Zeroconf / Avahi.
#
#zeroconf_enabled      "yes"
#
# The argument to this setting will be the Zeroconf / Avahi unique name for
# this MPD server on the network.
#
#zeroconf_name         "Music Player"
#
###############################################################################


# Permissions #################################################################
#
# If this setting is set, MPD will require password authorization. The password
# can setting can be specified multiple times for different password profiles.
#
#password                        "password@read,add,control,admin"
#
# This setting specifies the permissions a user has who has not yet logged in.
#
#default_permissions             "read,add,control,admin"
#
###############################################################################


# Input #######################################################################
#

input {
        plugin "curl"
#       proxy "proxy.isp.com:8080"
#       proxy_user "user"
#       proxy_password "password"
}

#
###############################################################################

# Audio Output ################################################################
#
# MPD supports various audio output types, as well as playing through multiple
# audio outputs at the same time, through multiple audio_output settings
# blocks. Setting this block is optional, though the server will only attempt
# autodetection for one sound card.
#
# See <http://mpd.wikia.com/wiki/Configuration#Audio_Outputs> for examples of
# other audio outputs.
#
# An example of an ALSA output:
#
audio_output {
   type      "alsa"
   name      "My ALSA Device"
   device      "hw:0,0"   # optional
   format      "44100:16:2"   # optional
   mixer_device   "default"   # optional
   mixer_control   "PCM"      # optional
   mixer_index   "0"      # optional
}
#
# An example of an OSS output:
#
#audio_output {
#   type      "oss"
#   name      "My OSS Device"
#   device      "/dev/dsp"   # optional
#   format      "44100:16:2"   # optional
#   mixer_device   "/dev/mixer"   # optional
#   mixer_control   "PCM"      # optional
#}
#
# An example of a shout output (for streaming to Icecast):
#
#audio_output {
#   type      "shout"
#   encoding   "ogg"         # optional
#   name      "My Shout Stream"
#   host      "localhost"
#   port      "8000"
#   mount      "/mpd.ogg"
#   password   "hackme"
#   quality      "5.0"
#   bitrate      "128"
#   format      "44100:16:1"
#   protocol   "icecast2"      # optional
#   user      "source"      # optional
#   description   "My Stream Description"   # optional
#   genre      "jazz"         # optional
#   public      "no"         # optional
#   timeout      "2"         # optional
#}
#
# An example of a httpd output (built-in HTTP streaming server):
#
#audio_output {
#   type      "httpd"
#   name      "My HTTP Stream"
#   encoder      "vorbis"      # optional, vorbis or lame
#   port      "8000"
#   quality      "5.0"         # do not define if bitrate is defined
#   bitrate      "128"         # do not define if quality is defined
#   format      "44100:16:1"
#}
#
# An example of a pulseaudio output (streaming to a remote pulseaudio server)
#
#audio_output {
#   type      "pulse"
#   name      "My Pulse Output"
#   server      "remote_server"      # optional
#   sink      "remote_server_sink"   # optional
#}
#
## Example "pipe" output:
#
#audio_output {
#   type      "pipe"
#   name      "my pipe"
#   command      "aplay -f cd 2>/dev/null"
## Or if you're want to use AudioCompress
#   command      "AudioCompress -m | aplay -f cd 2>/dev/null"
## Or to send raw PCM stream through PCM:
#   command      "nc example.org 8765"
#   format      "44100:16:2"
#}
#
## An example of a null output (for no audio output):
#
#audio_output {
#   type      "null"
#   name      "My Null Output"
#}
#
# This setting will change all decoded audio to be converted to the specified
# format before being passed to the audio outputs. By default, this setting is
# disabled.
#
#audio_output_format      "44100:16:2"
#
# If MPD has been compiled with libsamplerate support, this setting specifies
# the sample rate converter to use.  Possible values can be found in the
# mpd.conf man page or the libsamplerate documentation. By default, this is
# setting is disabled.
#
#samplerate_converter      "Fastest Sinc Interpolator"
#
###############################################################################


# Volume control mixer ########################################################
#
# These are the global volume control settings. By default, this setting will
# be detected to the available audio output device, with preference going to
# hardware mixing. Hardware and software mixers for individual audio_output
# sections cannot yet be mixed.
#
# An example for controlling an ALSA, OSS or Pulseaudio mixer; If this
# setting is used other sound applications will be affected by the volume
# being controlled by MPD.
#
#mixer_type         "hardware"
#
# An example for controlling all mixers through software. This will control
# all controls, even if the mixer is not supported by the device and will not
# affect any other sound producing applications.
#
#mixer_type         "software"
#
# This example will not allow MPD to touch the mixer at all and will disable
# all volume controls.
#
#mixer_type         "disabled"
#
###############################################################################


# Normalization automatic volume adjustments ##################################
#
# This setting specifies the type of ReplayGain to use. This setting can have
# the argument "album" or "track". See <http://www.replaygain.org> for more
# details. This setting is disabled by default.
#
#replaygain         "album"
#
# This setting sets the pre-amp used for files that have ReplayGain tags. By
# default this setting is disabled.
#
#replaygain_preamp      "0"
#
# This setting enables on-the-fly normalization volume adjustment. This will
# result in the volume of all playing audio to be adjusted so the output has
# equal "loudness". This setting is disabled by default.
#
#volume_normalization      "no"
#
###############################################################################


# MPD Internal Buffering ######################################################
#
# This setting adjusts the size of internal decoded audio buffering. Changing
# this may have undesired effects. Don't change this if you don't know what you
# are doing.
#
#audio_buffer_size      "2048"
#
# This setting controls the percentage of the buffer which is filled before
# beginning to play. Increasing this reduces the chance of audio file skipping,
# at the cost of increased time prior to audio playback.
#
#buffer_before_play      "10%"
#
###############################################################################


# Resource Limitations ########################################################
#
# These settings are various limitations to prevent MPD from using too many
# resources. Generally, these settings should be minimized to prevent security
# risks, depending on the operating resources.
#
#connection_timeout      "60"
#max_connections      "10"
#max_playlist_length      "16384"
#max_command_list_size      "2048"
#max_output_buffer_size      "8192"
#
###############################################################################


# Character Encoding ##########################################################
#
# If file or directory names do not display correctly for your locale then you
# may need to modify this setting. After modification of this setting mpd
# --create-db must be run to change the database.
#
filesystem_charset      "UTF-8"
#
# This setting controls the encoding that ID3v1 tags should be converted from.
#
id3v1_encoding         "UTF-8"
#
###############################################################################

follow_outside_symlinks "yes"
follow_inside_symlinks "yes"
zeroconf_enabled "yes"
zeroconf_name "Voyage Music Player"
mixer_type "software"

Change this section:

audio_output {
   type      "alsa"
   name      "My ALSA Device"
   device      "hw:0,0"   # optional
   format      "44100:16:2"   # optional
   mixer_device   "default"   # optional
   mixer_control   "PCM"      # optional
   mixer_index   "0"      # optional
}

To this, please:

audio_output {
   type      "alsa"
   name      "My ALSA Device"
   device      "hw:0,0"   # optional
}

And please add a # in front of this line near the end of your mpd.conf, that is, this:

mixer_type "software"

should be this:

#mixer_type "software"

or just delete line.

Then issue the command, as root: /etc/init.d/mpd restart


Two problems with your mpd.conf:

1. You had everything resmapled to 16/44.1
2. You had the software volume control enabled.

Both degrade sound quality, especially if the volume in your mpd client was less than %100.

Try my changes and then take a listen for a couple of days.

[nyc_paramedic]

Here it is:..

Please also uncomment this line in the Volume Contol MIxer section of mpd.conf:

#mixer_type         "disabled"

So that it's:

mixer_type         "disabled"

[TJHUB]

Change this section:

audio_output {
   type      "alsa"
   name      "My ALSA Device"
   device      "hw:0,0"   # optional
   format      "44100:16:2"   # optional
   mixer_device   "default"   # optional
   mixer_control   "PCM"      # optional
   mixer_index   "0"      # optional
}

To this, please:

audio_output {
   type      "alsa"
   name      "My ALSA Device"
   device      "hw:0,0"   # optional
}

And please add a # in front of this line near the end of your mpd.conf, that is, this:

mixer_type "software"

should be this:

#mixer_type "software"

or just delete line.

Then issue the command, as root: /etc/init.d/mpd restart


Two problems with your mpd.conf:

1. You had everything resmapled to 16/44.1
2. You had the software volume control enabled.

Both degrade sound quality, especially if the volume in your mpd client was less than %100.

Try my changes and then take a listen for a couple of days.

I made these changes.  Thank you.  FYI, I knew to keep the volume at 100% for digital stuff, so it never left 100%.

I do hear an immediate but subtle difference.  I'd say it's a move in the right direction towards what I was looking for.  I'll give it some time to really know how I feel.

I'm off to spend 4 days in the woods with the family camping, so I'll be off the grid.  Next week I'll be testing a Wire World Silver Starlight USB cable and a Paul Hynes PS.  I'm looking forward to it!

For the record, I was not ultimately happy with my Duet.  It's very good, but has been lacking a bit of life in the upper midrange and treble.  It's not bad, but it seems to leave me wanting.  The Duet has wonderful tonality and near perfect weight in the midrange and bass. 

Here is what the Alix changed:
1.  The sound stage lost about 4' or more of depth.
2.  The sound stage is a couple of feet wider on each side.
3.  There is better treble extension and more life to the upper midrange/lower treble.
4.  There is a lack of overall clarity and definition with a sort of distorted sound, but it is subtle.
5.  Tonality seems to have shifted higher, as in the things sound a bit too thin with less harmonics.
6.  The bass is a bit lean and lacks some definition that I'm used to.
7.  There is more of a 3D effect to the sound with tracks that use phase technology.
8.  The sound stage lacks some separation.
9.  Lastly, and this is the big one, I just can't seem to get an attachment to the music.  It sounds ok, but there is no passion or wow factor for me.

There may be something I'm missing, but I think that's it. 

The Alix may sound great if I hadn't heard better.  I do like it better with some tracks, but most just seem a bit off to me.  I hope this makes sense.

I'll have more next week.  I'm very hopeful things work out because I like everything else about the Alix. 

 

[TJHUB]

Alix update (just a quick one with limited detail):

First, the changes that nyc_paramedic suggested really made a big difference.  The sound quality improved significantly immediately.  I also confirmed this after my camping trip this weekend.

Then I swapped my cheap USB printer cable with a Wire World Silver Starlight USB cable.  This also improved things slightly.  The main changes seemed to be clarity and separation in the sound stage and a darker background.  A "huge" difference, no.  But a significant enough change to warrant the used purchase price of the cable.  However, the Alix still wasn't floating my boat.  It sounded good, but there was no magic.

About an hour ago, I replaced the stock wall-wart with a Paul Hynes power supply.  I adjusted its output to 12v and fired up the Alix.  The change in sonics is unbelievable.  I had no idea how grungy the sound was with the stock PS.  Everything is absolutely clear and FAR more realistic sounding.  The sound now easily grabs me and I can often say "wow, that sounds GREAT!"  It sounds like more information is in the music because you can easily hear so much more.  The slight distorted sound the Alix used to have must have blended away a lot of detail.  With the linear PS, that distortion is replaced with crystal clear micro dynamics.  Instruments sound wonderful and vocals are so much cleaner and have a realistic presentation.  After hearing this, the Alix just sounds bad with the stock PS.

So now things have changed enough that the Alix is a very interesting source.  I need more days to listen to more material, but I am very willing to put in the time with the Alix. 

I'll try and do a better job of descriptions this weekend.

   

 

[ted_b]

Terry, so glad you are back from the camping trip (hot enough?) and that these changes (especially my Hynes package    ) have made things much more musical for the Alix.  Look forward to you extended impressions.  Question: why did you go to the trouble of adjusting the Hynes down to 12V from the 18V I had it set at?  Does it matter?  Wasted heat maybe?

[TJHUB]

Terry, so glad you are back from the camping trip (hot enough?) and that these changes (especially my Hynes package    ) have made things much more musical for the Alix.  Look forward to you extended impressions.  Question: why did you go to the trouble of adjusting the Hynes down to 12V from the 18V I had it set at?  Does it matter?  Wasted heat maybe?

The camping trip was VERY hot!  We had really high humidity, so the 88 degree temps felt more like 100 degrees.  Then add no wind whatsoever, and it's HOT!  But that's why they made pools and lakes. 

The Paul Hynes has transformed the Alix into something interesting.  I hope it only gets better for me as I listen.  The ONLY complaint I can muster is I'd like the sound to have a bit more weight to it.  It's not necessarily thin sounding, but I think the Duet had me used to heavier tone.  What's "correct"?

I adjusted the Paul Hynes supply down to 12v because it took me 30 seconds and someone in this thread made the statement that anything over 7 or 8v was regulated down and dissipated off as heat.  So I figured why do that?  12v seems high enough for extra power in that regard.  I could try the Tim Allen approach of "MORE POWER" just to see what happens.  I'll let you know after I get better acquainted with the sound I have right now.

Is what I'm hearing what you would say you heard with the Paul Hynes on the Alix?

[ted_b]

Yes, it's exactly what I heard with the Hynes/Alix.  I seem to remember, too, that the weight/heft tended to settle in after the Hynes was in place for, say, 48 hours.   The USB cable will settle in a bit, too (even though it is used).

[TJHUB]

Yes, it's exactly what I heard with the Hynes/Alix.  I seem to remember, too, that the weight/heft tended to settle in after the Hynes was in place for, say, 48 hours.   The USB cable will settle in a bit, too (even though it is used).

I hope you're right.  I really do. 

Maybe any weakness I'm hearing now could just be my DAC.  I do plan to try some others, but I would LOVE to get a Zodiac+.

And thanks for the opportunity to audition the Paul Hynes supply. 

[nyc_paramedic]

The Paul Hynes has transformed the Alix into something interesting.  I hope it only gets better for me as I listen.  The ONLY complaint I can muster is I'd like the sound to have a bit more weight to it.  It's not necessarily thin sounding, but I think the Duet had me used to heavier tone.  What's "correct"?

I don't know if what I am about to say is an apples to apples comparison, but when I inserted my Placette RVC (passive via the use of Vishay S-102 hand trimmed resistors) into my system I noticed something along the same lines as what you're describing. I thought the system sounded "thin", but after prolonged listening and contemplation I realized that my system became more transparent.

Some would suggest that the "warmth" people think they're enjoying in any given hi-fi system is most likely a layer of distortion introduced by some component. The more I listened to my system without an active preamp the more I realized that things sounded a bit more like they sound in real life: a violin at Carnegie; a bouzouki played at a Greek wedding (or by Stephin Merritt --  http://en.wikipedia.org/wiki/Stephin_Merritt); youths on an NYC subway platform banging drum sticks on 5 gallon buckets, etc. They all sound, crisp, clean and pierce the air with authority. They never sound "warm" but do sound beautiful nonetheless.

[TJHUB]

I don't know if what I am about to say is an apples to apples comparison, but when I inserted my Placette RVC (passive via the use of Vishay S-102 hand trimmed resistors) into my system I noticed something along the same lines as what you're describing. I thought the system sounded "thin", but after prolonged listening and contemplation I realized that my system became more transparent.

Some would suggest that the "warmth" people think they're enjoying in any given hi-fi system is most likely a layer of distortion introduced by some component. The more I listened to my system without an active preamp the more I realized that things sounded a bit more like they sound in real life: a violin at Carnegie; a bouzouki played at a Greek wedding (or by Stephin Merritt --  http://en.wikipedia.org/wiki/Stephin_Merritt); youths on an NYC subway platform banging drum sticks on 5 gallon buckets, etc. They all sound, crisp, clean and pierce the air with authority. They never sound "warm" but do sound beautiful nonetheless.

I understand what you are saying, but this situation is more than that.  The bass isn't defined or sounding right at all.  It's just plain anemic.  With my Duet as a source, the sound was a bit too heavy.  So it would seem I've gone too far in the other direction.  Until I swapped USB cables...

I was initially using an old printer cable when I first fired up the Alix.  Of course the Alix wasn't sounding too good, so I was looking at the USB cable and of course the power supply.  Once I installed the Wire World cable, I never removed it even after getting the power supply.  So as I was listening tonight, I was hearing what I've heard EVERY time I had any type of silver cable in my setup.  I figured if it truly was the silver USB cable causing the problem, then the printer cable should show a difference.  I decided to swap the USB cable back to the printer cable.

Sure enough, the sound had more weight to it and the timbre shifted back down to where I think it should be.  The sound changed exactly as I would have described when swapping in a silver cable.  So it seems that even silver USB cables won't synergize with my mix of components.  This same thing happens whenever I use any cable with silver. 

The sound has moved a long way in the right direction.  Now I need to listen for a while to understand if there is anything else I'm not liking with the sound.

This is lesson in changing too many things at the same time. 

[jrebman]

Terry,

Actually, the theory on the silver usb cable makes sense to me as I typically hear the same things (with a couple of noteworthy and expensive exceptions).  The standard starlight usb cable, if anything, is a tad heavy on the bass in my system, though not overdone.  If you can pick up a new or used db audo labs Essential usbcable, those too are quite nice, especially if cryoed.

Also, fwiw, just about every piece of digital gear I've ever had pretty much sounds it's best after 36-72 hours of being powreded on and are the only components in my system that I just leave powered on.  It often takes that long for everything to reach thermal equibrilium, and that seems most important to digital components.  Like I said, fwiw, but to my ears it definitely makes a difference.

-- Jim

[TJHUB]

Terry,

Actually, the theory on the silver usb cable makes sense to me as I typically hear the same things (with a couple of noteworthy and expensive exceptions).  The standard starlight usb cable, if anything, is a tad heavy on the bass in my system, though not overdone.  If you can pick up a new or used db audo labs Essential usbcable, those too are quite nice, especially if cryoed.

Also, fwiw, just about every piece of digital gear I've ever had pretty much sounds it's best after 36-72 hours of being powreded on and are the only components in my system that I just leave powered on.  It often takes that long for everything to reach thermal equibrilium, and that seems most important to digital components.  Like I said, fwiw, but to my ears it definitely makes a difference.

-- Jim

I'm glad to hear someone hears things the way I do, although I do realize that every room and system varies quite a bit. 

I've had the Alix powered up since getting it running a week ago last Friday (or it's been running for 11 days straight now if that didn't make sense).  I haven't heard any real changes other than the USB cables and power supplies.

After listening tonight, I think I'm ready to give up on the Alix.  Why?  It's just not consistent from track to track.  I listen to some things and it's good, other things it's ok, and others it's almost bad.  I just can't seem to love it, so that's a sign that spending more money on it is probably not a good idea for me.

Please understand that with my Duet, I could honestly play anything and really like it.  My only complaint is a slight lack of life in the treble and maybe a bit too much weight in the bass in midrange on some tracks.  As a total, the Duet's performance is actually surprisingly great.   I often get a very strong connection to the music.  I haven't ever felt that way with the Alix.  It can sound good, but it never really grabbed me.  I'm getting the feeling I'm going the wrong direction with it.

I've had friends from this forum to my home many times.  I think they believe my setup is the best sounding setup they've heard in any home.  I know I've heard nothing better...yet.  I doubt any of them would think much of my setup with the Alix.  I know it's all about synergy, and I just don't have it with the Alix at all. 

Maybe I'm a cable or DAC away from something special, but I don't have the cash right now to find out.  So I think I'm done.  I'll keep the Alix for now and maybe try again when I can afford a "better" USB cable and/or DAC. 

[TJHUB]

Ok, it's me again...

I decided to use a Squeezebox Touch I picked up on sale to replace my Duet.  I got the Touch up and running and I did the digital mods to it and thought it sounded pretty darn good.  The sound seems to be more forgiving overall than the Alix was, and maybe a little better than the Duet.  I listened to it as much as I could yesterday.

Today, I fired up the Alix again.  I had the Paul Hynes linear PS on it and my USB printer cable.  I'm very torn about this as the Alix does some things amazingly well.  The treble and midrange detail is amazing without really ever sounding bright.  Vocals are MUCH better than the Duet or Touch.  The only thing I seem to have is a slightly weak bass level compared to the Duet or Touch.

I mentioned a few posts back that I was going to try my netbook with my DAC again.  Well I did that today.  What's funny is that the result is exactly what I expected.  The netbook connected via USB to my DAC sounds VERY similar to the Duet and Touch connected via digital coax.  I do even think the netbook sounded better than the Duet or Touch, but I didn't really care to find out in detail.  So it appears that the Alix's USB output sounds very different from my netbook's (running Foobar 2000).

So I'm back to testing with the Alix.  I plan to give it more time this time around.  Maybe my ears will adjust to the sound.

[jrebman]

Terry,

Just curious, have you tried any footers and/or damping/weight on the box?  One of my alix bxes has some herbies baby booties on it and for the time I was using it, that did make a nice, audible difference, and as with the mac mini, the whole presentation can be tuned with different footers, mass loading, etc.  It is easyto try and can be dirt cheap, ultra expensive, or anywhere in between.  Just something you may want to try if you haven't already.

-- Jim

[TJHUB]

Terry,

Just curious, have you tried any footers and/or damping/weight on the box?  One of my alix bxes has some herbies baby booties on it and for the time I was using it, that did make a nice, audible difference, and as with the mac mini, the whole presentation can be tuned with different footers, mass loading, etc.  It is easyto try and can be dirt cheap, ultra expensive, or anywhere in between.  Just something you may want to try if you haven't already.

-- Jim

To be honest, I'm not sure I believe in this, but I will try it today. 

[TJHUB]

The Alix was completely transformed today.  I seem to be able to get it to sound anyway I want just by swapping interconnects and USB cables.  Sorry if you cable naysayers don't believe it.

I still could just not find love for the Alix as a source.  The treble was good, the midrange lacked a bit of weight, and the bass (read midbass) was down right anemic.  It wasn't horrible, but I just couldn't seem to learn to like it.

Earlier today, I decided to swap a couple of ICs around.  My first swap was the ICs from my preamp to my amps.  The change was HUGE.  I've always been able to hear differences in cables, but it's usually limited to the treble and rarely midrange.  This time these cables I've owned for a long time sounded VERY different.  I now had what seemed like 3 times the bass as before, midrange weight, and dulled treble.  To be honest, the sound was now very close to the Duet or Touch with just a bit more open treble and midrange.  I was shocked to say the least. 

Then I decided to swap the ICs between the DAC and my preamp.  This moved the sound even further in the wrong direction.  The sound was now getting thick with lifeless treble.  I'm being a little picky here, but you get my point. 

Since all of this testing was with my old USB printer cable, I thought I'd throw the Silver Starlight USB cable in to see what would happen to the thick sound and lifeless treble.  The change was very positive.  I'm only noticing a slight timbre shift up this time in the treble with a nice open and extended sound, and the midrange and bass are sounding very balanced.  As a matter of fact, this combination seemed the most interesting, so I left it.  After a couple of hours of listening, I'm finding little to complain about.

I might try a few more combinations of cables, but at least I'm finally hearing something very good from the Alix. 

There is something at play here that isn't making sense to me.  I've spent a lot of time with my current mix of gear and carefully picked what I thought was the perfect mix of cables to use.  I even thought I knew exactly what each cable did to the sound.  Now it seems everything has changed.  It's as if I changed much more than the source.  It also seems the cables aren't acting the same way.  And it now seems the changes are more drastic than they ever were before.  Anyone could pick out these changes as they are VERY obvious. 

So it looks like the Alix may just work out for me.  Especially now that I know the perfect cable combination seems achievable.   I may even keep the Silver Starlight USB cable that I thought was a goner for sure. 

I do also have a question regarding hi-rez playback.  If I shut off all down sampling and any mixer stuff, how is it my non-hi-rez DAC can still play 24/96 files?  And I tried a 24/192 file, and the DAC played it but it would play for 1 second, drop out for one second, play for 1 second, etc.  I didn't think it would play at all.  Strange.

I'm also very torn about what to do about a power supply.  The Paul Hynes PS is working really well, but it's a bit pricey for me.  I'm hunting for an alternative solution like the KECES or ?  I just hope I don't lose something over the Paul Hynes PS.

[nyc_paramedic]

I do also have a question regarding hi-rez playback.  If I shut off all down sampling and any mixer stuff, how is it my non-hi-rez DAC can still play 24/96 files?  And I tried a 24/192 file, and the DAC played it but it would play for 1 second, drop out for one second, play for 1 second, etc.  I didn't think it would play at all.  Strange.

The way we have mpd.conf setup is to pass along bit-perfect audio to your DAC without any molestation, i.e., don't convert streams to 48Khz, etc. But ALSA will try to it's best to play a file for you under certain circumstances...

My Ayre QB-9 will handle sample rates of 32Khz, 44.1, 48, 88.2, 176.4, and 192 at 16 bits and 24 bit word lengths. If I feed it any combination of these it will pass them to the DAC as is. But, if I send a 24/96 file (i.e., with this specific mpd.conf and ALSA setup) to a 16/44.1 DAC, it will do its best to down-sample the file and send it to the DAC.

I think there is a way to prevent this, but the end result is no audio and a vague error message. The reason you get pauses with 24/192 material is that the 500Mhz geode CPU isn't fast enough down-sample 24/192 to 16/44.1 in real time.

The alternative is to find a software sample rate converter you like and convert some of your hi-res material so that the Alix doesn't have to do it in real time.

I hope this simplified explanation clarifies what's going on.

P.S when all I had at one time was a 16/44.1 Wavelength Brick USB DAC, I did like the fact that the Alix could down-sample some of the 24/96 material I had. It also sounded quite good!

[TJHUB]

The way we have mpd.conf setup is to pass along bit-perfect audio to your DAC without any molestation, i.e., don't convert streams to 48Khz, etc. But ALSA will try to it's best to play a file for you under certain circumstances...

My Ayre QB-9 will handle sample rates of 32Khz, 44.1, 48, 88.2, 176.4, and 192 at 16 bits and 24 bit word lengths. If I feed it any combination of these it will pass them to the DAC as is. But, if I send a 24/96 file (i.e., with this specific mpd.conf and ALSA setup) to a 16/44.1 DAC, it will do its best to down-sample the file and send it to the DAC.

I think there is a way to prevent this, but the end result is no audio and a vague error message. The reason you get pauses with 24/192 material is that the 500Mhz geode CPU isn't fast enough down-sample 24/192 to 16/44.1 in real time.

The alternative is to find a software sample rate converter you like and convert some of your hi-res material so that the Alix doesn't have to do it in real time.

I hope this simplified explanation clarifies what's going on.

P.S when all I had at one time was a 16/44.1 Wavelength Brick USB DAC, I did like the fact that the Alix could down-sample some of the 24/96 material I had. It also sounded quite good!

Thanks for such a great explanation.  The good news is that I don't own any hi-rez I need to play.  I just have some sample tracks for testing purposes.  I plan to wait until I get a hi-rez capable DAC.  I just needed to understand why these files were playing.

[jkelly]

I really like the tone of my Alix/Tranquility/Keces combo.
I would like to have more bass.

Jeff

[nyc_paramedic]

I really like the tone of my Alix/Tranquility/Keces combo.
I would like to have more bass.

Jeff

Jeff,

Do you have any bass traps in your room? Maybe your front end is not at fault.