Re: [slim] Community Funded Squeezebox Replacement - Would you be interested?

2013-10-18 Thread Phil Karn
On 10/10/2013 08:17 PM, Mnyb wrote:
 
 If you run debian testing ? why not run the 7.8 beta version of the
 server that have some fixes for the perl issues ( perl 5.18 ).
 if your prepared to run a beta versions of an OS Why not of the server

I didn't even know there was a 7.8 beta version -- it's kept well hidden.

I managed to find and install it, but it still bombs out:


[13-10-18 01:55:07.8310] main::init (355) Starting Logitech Media Server
(v7.8.0, 1382017549, Thu Oct 17 20:07:31 PDT 2013) perl 5.018001
[13-10-18 01:55:07.8413] main::changeEffectiveUserAndGroup (971)
Warning: Logitech Media Server must not be run as root!  Trying user
squeezeboxserver instead.
[13-10-18 01:55:08.5945] Slim::Music::TitleFormatter::init (42) Warning:
Can't locate Slim/Schema/Track.pm:   Permission denied at
/usr/share/perl5/Slim/Music/TitleFormatter.pm line 42.

Mind you, I didn't find this in the logs; I had to invoke
squeezeboxserver manually to see these error messages.

All I want to do is to play music on my Squeezeboxes just as I've done
for years. I really shouldn't have to become a Perl hacker (or keep my
system way out of date) just to do that.

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[slim] Anyone have a succinct summary of the squeezebox server situation?

2013-10-10 Thread Phil Karn
I've heavily used several Squeezeboxes and Boom Boxes for years, but I'm
ready to throw them all in the trash.

I just wasted several hours in an unsuccessful attempt to get
squeezeboxserver (or logitechmediaserver, or slimserver, or whatever
it's called this week) running again on a Debian Jesse system after an
apt-get upgrade broke it.

I've lost track of how many times in the past my server has abruptly
stopped working after I've updated some obscure library or piece of Perl
on my system, and I've had to dig into the code to find out why. Users
shouldn't have to be skilled programmers just to play the music in their
libraries.

The Squeezebox hardware is nicely designed, but it is quite frankly
useless without reliable software to drive it. Perl was designed to
crunch Linux system logs and generate reports, which it barely does.
It's far too fragile a foundation on which to build a huge application
like a music storage and retrieval system with thousands of gratuitous
bells and whistles. It's also extremely slow even on fast modern hardware.

Has anyone considered a simple server to drive Squeezebox hardware as a
bare-bones network audio output device? I suspect I'm like most people
in that I control my player from my computer, not the Squeezebox's own
display and IR remote, so much of the software functionality just isn't
necessary. It doesn't really need to do anything but accept a raw audio
stream from a player like VLC or iTunes, possibly transcode it, and ship
it over the network to the player.

--Phil

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Re: [slim] Community Funded Squeezebox Replacement - Would you be interested?

2013-10-10 Thread Phil Karn
On 10/10/2013 06:12 PM, castalla wrote:

 
 I have LMS running on Fedora, Ubuntu and Debian - none of which are the
 latest and greatest -  end result, rock solid  systems and virtually no
 maintenance.

That's your choice, but it's not mine. I strongly believe in updating
software on a regular basis, primarily to fix security holes and bugs
and only secondarily to get new features. I also use my servers for
other purposes that require that they be kept up to date.

Well-written applications should never break when this happens.

Even when a non-backward-compatible change must be made to some module
(e.g., a library) in Debian Linux, an application can specify that it
needs the previous version and the package manager will handle this
automatically. It can even allow multiple versions to coexist so that
each application can use the one it wants. But the .deb files with the
Squeezebox media server don't seem to do this -- probably because there
are just far too many dependencies in the first place.

I think these bit rot problems are merely a symptom of the real
problem, which is that the Squeezebox media server tries to do far too
much in one huge program written in the wrong language. Countless media
player and library management programs with full-blown (and very
complex) GUIs already exist for every OS, ranging from iTunes to the
obscure, and I don't really see why I should have to use yet another one
just to send audio through Squeezebox hardware. Why reinvent the wheel?

What I really want is something like Apple's AirTunes or AirPlay, only
with an open and unencumbered protocol specification and support for all
the major codecs, including open-source ones like Ogg Vorbis and FLAC
that the commercial guys refuse to support. It was this support for open
codecs that attracted me to the Squeezebox in the first place.

--Phil

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Re: [slim] Community Funded Squeezebox Replacement - Would you be interested?

2013-10-10 Thread Phil Karn
On 10/10/2013 06:50 PM, castalla wrote:

 That's your choice!  hardware is so cheap these days that you can easily
 run a headless server for LMS at about 35 USD.
 
 I think you are just giving yourself unnecessary grief.

Oh, I know hardware is cheap. I even bought a separate server recently
to primarily host our media archive (though it cost more than $35).

But I explained my main reason I keep my systems updated is to patch
security holes and bugs. Perhaps that's why I've never had a detected
break-in.


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Re: [slim] Community Funded Squeezebox Replacement - Would you be interested?

2013-10-10 Thread Phil Karn
On 10/10/2013 06:16 PM, garym wrote:
 
 You might be interested in the ickstream project. 
 
 http://forums.slimdevices.com/showthread.php?98467-Pre-Announcement-ickStream-Music-Platformhighlight=Ickstream

Thanks for the pointer. There is very little in the way of specifics,
and a proprietary platform isn't very appealing, but any alternative to
the present software is good to have.


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Re: [slim] Community Funded Squeezebox Replacement - Would you be interested?

2013-10-10 Thread Phil Karn
On 10/10/2013 07:22 PM, paulster wrote:
 
 Use Debian then. I've updated mine regularly over the last 5 years and
 keep LMS up to date as well, and have never had a hiccup. There are more
 bleeding-edge distros than Debian but their security patches are up to
 date.


That's exactly what I run -- Debian testing -- but slimserver
frequently breaks during upgrades. It is now broken again and I've not
been able to fix it. Usually I have to wait for another release but I
haven't seen anything past 7.7.3.


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Re: [slim] Squeezeboxserver on Linux broken

2011-07-20 Thread Phil Karn
My Squeezebox servers have been down for a week because of this most
recent problem and I get the very definite impression that no one really
understands what's wrong (I certainly don't).

Perl is fine for its original purpose of running quickie, one-off
scripts on text and system log files. That's how I use it. But I think
it's simply the wrong platform for a large, complex and unique package
like squeezeboxserver with many dependencies on a huge and only loosely
standardized run-time environment that it does not and apparently cannot
fully control.

I understand that squeezeboxserver is an open source project that relies
on volunteer contributions. That and its support for open audio formats
like FLAC and Ogg Vorbis initially attracted me to it. But I've also
dropped a fair chunk of change on Logitech's proprietary hardware
players, and they're bricks without a working squeezeboxserver. I think
Logitech owes its customers at least one working version of the server
for each advertised platform that runs without constant attention from a
local Perl hacker.

Even when it does work, squeezeboxserver has always seemed much too slow
for what it does. It seems to suffer from a serious case of feature bloat.

All I've ever really wanted is a simple, fast and above all *reliable*
way to play my large, multi-format music collection at home. Even a
command-line interface would be enough. A barebones textual web
interface would be a plus, along with the ability to stream a simple
audio-over-HTTP feed. I certainly don't need RSS feeds scrolling across
my player display, or dozens of other plugins and options of
questionable utility. When I want to read the news I fire up Firefox on
my laptop. It does a much better job.

I really hate to say this but it's probably time to re-survey the
network music player market for something that doesn't depend on unique
client-server protocols, even when the server is open source. There's
something to be said for widely implemented application protocols like
HTTP even when they lack a feature you might think you could use
someday. Sometimes all you really want is to play your music without
first having to parse a Perl error traceback.

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[slim] Squeezeboxserver on Linux broken

2011-07-19 Thread Phil Karn
I accepted the package update on both of my Linux servers, and
squeezeboxserver stopped working on both. The logs showed only that the
wrapper was continually restarting the daemon, so I invoked it manually
and got this very cryptic message. What should I make of it? I'm not a
perl hacker, I just want to play my music... Thanks.

Use of inherited AUTOLOAD for non-method YAML::Syck::DumpYAML() is
deprecated at /usr/share/squeezeboxserver/CPAN/YAML/Syck.pm line 65.
The following modules failed to load: DBI DBD::mysql EV
XML::Parser::Expat HTML::Parser JSON::XS Digest::SHA1 YAML::Syck GD
Sub::Name


***

NOTE:

If you're running some unsupported Linux/Unix platform, please use the
buildme.sh
script located here:

http://svn.slimdevices.com/repos/slim/7.5/trunk/vendor/CPAN/

You should never need to do this if you're on Windows or Mac OSX. If the
installers
don't work for you, ask for help and/or report a bug.

If 7.5 is outdated by the time you read this, Replace 7.5 with the
major version
of Squeezebox Server you are running.

***


Exiting..
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Re: [slim] Squeezeboxserver on Linux broken

2011-07-19 Thread Phil Karn
On 7/19/11 9:47 AM, aubuti wrote:
 
 You would be much more likely to get a helpful response if you indicated
 (a) which Linux distro and version you are using, (b) which version of
 SBS you are using, (c) what version of SBS was previously working on
 the server, (d) what method you used to install SBS, and (e) what
 hardware platform you are using.
 
 

a) Debian testing
b) 7.5.5, picked up by apt-get with sources.list line deb
http://debian.slimdevices.com stable main
c) Whatever was previously served through debian.slimdevices.com
d) apt-get update;apt-get dist-upgrade
e) x86_64, custom compiled 2.6.39.3 kernel.

I see that others are having problems too, I'll just find a .deb package
for a previous version that works and revert until the bugs can be
fixed. Thanks.

Phil
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[slim] GPG keys for Debian/Ubuntu packages?

2010-12-12 Thread Phil Karn
Are the .deb packages for Debian and Ubuntu Linux signed with GPG/PGP keys?

This is standard practice for the Debian repositories to prevent the
distribution of malware should a repository be compromised.

I don't see a key for the Squeezeboxserver packages distributed from
debian.slimdevices.com. Should there be there?

Thanks,

Phil
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[slim] Excessive traffic to www.mysqueezebox.com

2010-09-29 Thread Phil Karn
Every 5 minutes, like clockwork but sometimes even more often, the
squeezeboxserver running on my Linux box opens a HTTP connection to
www.myqueezebox.com and performs a GET.

I've turned off all sorts of network syncing options and other features
I've never used but the GETs continue. It appears to repeatedly pull
down some sort of menu that never changes.

I run a local server to be as self-sufficient as possible, and aside
from connecting on request to external audio sources such as Sirius
Radio and the Internet Archive, I see little reason for my server to
ever talk to the outside world, much less every 5 minutes even when it's
idle. Is there any way to turn it off, or at least knock the frequency
way down? Thanks.

Phil

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Re: [slim] Playing .m4a files?

2008-11-01 Thread Phil Karn
On Thu, Oct 30, 2008 at 3:31 PM, Books 
[EMAIL PROTECTED] wrote:


 Somebody give me the low down - I am trying to play .m4a files (AAC and
 stuff) through SqueezeCentre (latest version). I use SqueezeCentre and
 then connect to the stream file with WMP at work. Is this possible,
 what needs to be done?


What about Linux? I used to have it working after manual editing of
custom-config.conf, but it broke in one of the upgrades over the past month
or two. I can't even figure out where the config files are supposed to be
now.
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[slim] Boom aux input gain low?

2008-10-22 Thread Phil Karn Jr, KA9Q
I just got my Boom. Works great as a network player. I hooked my iPod up
to the aux input and found that the gain is VERY low. With the iPod and
Boom both cranked all the way up, I can just hear it across the room.

What is the rated sensitivity of the Boom aux input?
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Re: [slim] Boom aux input gain low?

2008-10-22 Thread Phil Karn
I just found the aux gain setting in the squeezecenter config page. It was
at 50%. Turning it to 100% helped, but it's still not as high as it should
be.
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Re: [slim] Does anyone buy MP3s from Amazon?

2008-08-16 Thread Phil Karn
On Thu, Aug 14, 2008 at 9:58 PM, Goodsounds 
[EMAIL PROTECTED] wrote:


 So, what is your take on the CD quality issue?  I personally prefer
 vinyl to plastic based on my listening experiences, but do you think
 that to be unfounded?


Wow. A vinyl fan who is actually open to the possibility that CDs might
actually be better? I think you are the first I've ever seen.

This has been a hotly debated topic ever since the CD came out in the early
80s. This is unfortunate because there's a perfectly good way to settle the
issue scientifically, once and for all.

My reasoning is as follows. The purpose of an audio reproduction system is
to reproduce the original signal as faithfully as possible. It should not
add or take away from that signal in any way. Making a recording sound
good is the job of the recording engineer; it is not up to the
reproduction system to modify the engineer's work product in any way.

Ergo, if you can't tell the difference between the original and reproduced
signal, then you can't complain about the quality of that reproduction
system.

So here's what you do. You start with your favorite test signal in analog
form. It can be from any source you like, including a vinyl record; your
choice. You produce a digital version of that signal by running it through a
44.1 kHz 16 bit/sample A/D and D/A, being careful to set the overall analog
gain to exactly unity (0 dB). Now we have two analog signals, one direct
from the input source and the other having been passed through the codec.

Next you construct two audio switches. The first switch, the listener
switch, has two positions labeled simply A and B. The second switch, the
control switch, has four positions and is physically placed so that the
listener can't tell its setting. The control switch affects the behavior of
the listener switch as follows:

1. Positions A and B both play the analog signal.
2. Position A gives the analog signal, position B gives the digital signal.
3. Position B gives the digital signal, position A gives the analog signal.
4. Positions A and B both give the digital signal.

It's important to construct the circuits so the listener has absolutely no
cues as to the setting of the control switch. For example, there must be no
audible switching transients or changes in gain, latency, bandwidth or
phase. This means ensuring there is no perceptible latency in the A/D - D/A
chain.

You can see what comes next. The experimenter randomly chooses a control
switch position. The listener must then determine whether the listener
switch does anything. Note: the listener is NOT asked to determine which
position is analog or digital, or to evaluate which one sounds better. He
only has to tell if the switch does anything or not.

You do this a number of times, each time setting the control switch to a
random position determined by a pair of coin tosses (the experimenter should
NOT choose the switch position by himself).

The bottom line is simple. If the listener can't tell with better than
chance accuracy whether his switch actually selects between the original and
digital signal sources, then it is clear that the digital path is not
modifying the signal in any detectable way. And if he can't detect the
difference, he can't claim that the digital system is somehow worse.

The listener may have other perfectly reasonable reasons to prefer a vinyl
version of a recording over the CD. For example, the mixing and equalization
on the LP might be subjectively better. But that is not something you can
blame on the CD (or digital audio) per se; the fault is the recording
engineer's who prepared the signal given to the CD mastering system.

--Phil
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Re: [slim] problem with ogg file playback

2008-06-14 Thread Phil Karn
Mnyb wrote:
 I don't have any OGG files myself, so i can not experiment with this,
 somebody else have to chime in here.
 Do experiments with this, you wont damage anything, you can always
 restore the settings to what it was before.
 
 A guess would be that setting the OGG Vorbis box in OGG Vorbis to
 disabled (is native per default ) would do it.
 
 Then I suppose SC would use either AIFF, FLAC or MP3 to play ogg files.
 
 

I've just run into this same problem with a set of ogg files from the 15 
October 2004 Live at the Patio concert by Robert Walter's 20th Congress 
hosted on archive.org:

http://ia310125.us.archive.org/2/items/RWTC2004-10-15.akg.flac16/

If I disable built-in player ogg decoding in the files configuration 
tab, then Squeezecenter transcodes ogg to FLAC that my Squeezebox 2 
plays just fine.

I can't determine which encoder or options were used to produce the 
troublesome ogg files.

I downloaded the FLAC version of a song from that set. It plays fine, of 
course. I encoded it into ogg using oggenc version 1.2.0 on Linux with 
default options. Unlike the version on archive.org, my locally created 
ogg file plays fine on my Squeezebox 2.

So it's confirmed. The problem is in the Ogg Vorbis decoder in the 
Squeezebox firmware. It cannot decode some Ogg Vorbis files that other 
decoders handle just fine. Since there appears to be no easy way to tell 
which files can't be decoded in the player, the only viable workaround 
until the firmware can be fixed is to completely disable built-in Ogg 
Vorbis decoding and decode it on the server.

I like Ogg Vorbis and the other open audio formats as much as anyone, 
but I've been wondering about its utility built into a player. I guess 
it's useful when streaming remote radio stations (or song archives) 
that generate it when your local computer is off.

Since Ogg Vorbis is a better lossy codec than MP3, in theory it could do 
better than MP3 for bit rate limiting. But it looks like MP3 is 
hardwired into the bit rate limiter, probably because it was the only 
lossy codec available in the player at the time. There's no reason Ogg 
Vorbis couldn't become the default bit rate limiter for Squeezeboxes 
that implement it correctly. That could also avoid possible legal 
problems in relying on LAME.


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[slim] Minor nit: lossless codecs are never CBR

2008-01-20 Thread Phil Karn
I know this is a really minor nit, but I've noticed it for a long time.

Slimserver displays, on the Squeezebox and the web page, the bit rate for FLAC
and other lossless material as CBR (Constant Bit Rate). Lossless compression
algorithms like FLAC, ALAC, Monkey's Audio, Shorten, etc, are never constant bit
rate. They must vary with the redundancy in the audio stream.
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[slim] Replacing mDNSResponder with avahi

2007-12-26 Thread Phil Karn
I already run avahi (a Bonjour daemon) on my Linux box and I'd rather have it
advertise my slimserver instead of having slimserver run its own copy of
mDNSResponder.

In the past I've done this by manually creating the appropriate avahi config
file and commenting out the mDNSResponder invocation in Slimserver. But avahi is
getting pretty popular now so I'm wondering if any thought has gone into making
slimserver work with it a little more naturally.

--Phil
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Re: [slim] Re: SlimServer - A pain in the ass...

2006-12-31 Thread Phil Karn

Lost Viking wrote:


It takes ~45secs from clicking to diplaying the result page. Is this
normal? Will it be as slow using the Transporter (once I have it in my
hands, it is still on its way..)?? *shudder*


No, that's definitely not normal. IMHO, the Slimserver web interface 
could be a lot faster, but here it's still only a couple of seconds at most.


One of the recent Slimserver updates apparently added a lot of internal 
threading. That noticeably sped up the web interface. Make sure you're 
running recent code.


Let me add a hearty amen to the other suggestions to run Slimserver on 
Linux, not Windows. Windows is a terrible server OS. Scratch that, 
Windows is a terrible OS, period. Jettisoning all that virus scanning 
cruft is just one of many reasons to dump it.


Here are some other suggestions to speed up a file server. First, avoid 
external USB hard drives. In my experience, a USB 2.0 external drive is 
less than half as fast as an internal drive.


Second, if you have the cabinet space and the budget, consider RAID. 
Without RAID (or a current backup), a failed drive is a major disaster. 
(Do you really want to re-rip a thousand CDs?) With RAID, a failed drive 
is just a minor nuisance. RAID can also speed up read performance quite 
noticeably. Linux has a very nice software RAID subsystem that avoids 
having to spend money on hardware RAID controllers.


--Phil

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Re: [slim] Re: Ogg Encoding Options

2006-12-31 Thread Phil Karn
I haven't seen any comments on my problems playing Ogg files with 6.5.1. 
Can anybody confirm that they *can* play Ogg files on 6.5.1 and a 
Squeezebox 2 or 3 with firmware version 71 without server decoding?


Phil

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Re: [slim] Re: First sec of song getting clipped off

2006-12-30 Thread Phil Karn

Eric Seaberg wrote:


I'd really rather NOT re-rip everything with FLAC.  I have thought
about it just to get the extra quality bump, but iTunes won't read them
which, for me anyway, REALLY puts a crimp in my tag editing and
organization for SlimServer.  As a Mac user, there just aren't as many
options to work with FLAC yet.


This was a problem for us too until I discovered mt-daapd, an open 
source music server (functionally somewhat similar to Slimserver) that 
speaks DAAP, Apple's protocol for sharing iTunes collections over a LAN.


I run mt-daapd alongside Slimserver on the Linux box that has our music 
library. The clients see iTunes' native file formats (.mp3, .m4a, .m4p, 
.wav) unchanged, while non-iTunes formats like .flac and .ogg 
automatically appear to iTunes as WAV files.


The two servers coexist nicely. We can now use iTunes/mt-daapd for its 
superior search and seeking features and Slimserver for when we want to 
play something through a Squeezebox and good speakers.


Unfortunately, DAAP is read-only, so iTunes cannot edit tags on a remote 
server.


--Phil
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Re: [slim] Re: First sec of song getting clipped off

2006-12-29 Thread Phil Karn

Eric Seaberg wrote:


I'd really rather NOT re-rip everything with FLAC.  I have thought
about it just to get the extra quality bump, but iTunes won't read them
which, for me anyway, REALLY puts a crimp in my tag editing and
organization for SlimServer.  As a Mac user, there just aren't as many
options to work with FLAC yet.


I've standardized on FLAC for my library collection (with subsequent 
conversions to Ogg Vorbis, MP3 or AAC as needed for portable jukeboxes), 
but I also like to rip with iTunes for the convenience.


So my technique is to use ALAC when ripping, copy the ALAC files to the 
Linux server running SlimServer, and batch convert ALAC to FLAC with a 
little perl script I just wrote. The only hard part is converting the 
tags, but I think I do pretty much the right thing.


Anybody who wants my script is welcome to it.

--Phil
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Re: [slim] Re: Ogg Encoding Options

2006-12-28 Thread Phil Karn

rgmiller1974 wrote:


Just tried that; no luck.  Tried both compiling from source and a
pre-built 32-bit binary (I'm running an Athlon64 so the one I compiled
was 64-bit.)  No matter what I use, I get the same errors from
slimserver.pl:

2006-12-19 22:11:59.1207 Error: Decoder does not support file format,
skipping track
2006-12-19 22:11:59.1209 Error opening current track, so mark it as
already played
2006-12-19 22:11:59.1213 Backtrace:

frame 0: Slim::Player::Source::errorOpening
(/opt/SlimServer_v6.5.0/Slim/Player/Source.pm line 634)
frame 1: Slim::Player::Source::notSupported
(/opt/SlimServer_v6.5.0/Slim/Networking/Slimproto.pm line 657)
frame 2: Slim::Networking::Slimproto::_stat_handler
(/opt/SlimServer_v6.5.0/Slim/Networking/Slimproto.pm line 387)
frame 3: Slim::Networking::Slimproto::client_readable
(/opt/SlimServer_v6.5.0/Slim/Networking/Select.pm line 238)
frame 4: Slim::Networking::Select::select (./slimserver.pl line
487)
frame 5: main::idle (./slimserver.pl line 440)
frame 6: main::main (./slimserver.pl line 1039)


I seem to have run into this same problem immediately after updating to 
today's (12/28/2006) release of 6.5.1 from 6.5.0. I'm not sure of the 
build date of the previous version; the timestamp on slimserver.pl was 
October 25.


Today's build came with a player firmware update to version 71, and now 
neither my Squeezebox 2 nor my Squeezebox 3 will play any of my Ogg 
Vorbis files. WAV, FLAC and MP3 still play fine. Unfortunately, I 
updated both players' firmware files so I can't test with a 2 or 3 that 
hasn't been updated, and I don't remember the previous firmware version.


My old Squeezebox 1 still plays Ogg Vorbis fine. I expected this because 
it doesn't handle Ogg natively, nor has it seen a firmware update. So 
server transcoding from Ogg to WAV still works with the current version.


My guess is that either the current firmware load for the Squeezebox 2/3 
somehow broke Ogg Vorbis decoding, or the slimserver somehow broke its 
handling of Ogg Vorbis for those models.


If the firmware is indeed broken, is there a simple workaround to force 
transcoding of Ogg Vorbis to WAV for Squeezebox 2/3?


Phil


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Re: [slim] Re: Ogg Encoding Options

2006-12-28 Thread Phil Karn

Phil Karn wrote:

Today's build came with a player firmware update to version 71, and now 
neither my Squeezebox 2 nor my Squeezebox 3 will play any of my Ogg 
Vorbis files. WAV, FLAC and MP3 still play fine. Unfortunately, I 
updated both players' firmware files so I can't test with a 2 or 3 that 
hasn't been updated, and I don't remember the previous firmware version.


I figured out how to downgrade my Squeezebox 2 to the previous firmware 
version, 64. That does *NOT* fix the problem; Ogg Vorbis files still 
won't play on the 12/28/2006 version of 6.5.1. The log shows the 
following backtrace:


2006-12-28 21:50:42.6919 Backtrace:

   frame 0: Slim::Player::Source::errorOpening 
(/usr/local/SlimServer_6.5_v2006-12-28/Slim/Player/Source.pm line 635)
   frame 1: Slim::Player::Source::notSupported 
(/usr/local/SlimServer_6.5_v2006-12-28/Slim/Networking/Slimproto.pm line 
658)
   frame 2: Slim::Networking::Slimproto::_stat_handler 
(/usr/local/SlimServer_6.5_v2006-12-28/Slim/Networking/Slimproto.pm line 
387)
   frame 3: Slim::Networking::Slimproto::client_readable 
(/usr/local/SlimServer_6.5_v2006-12-28/Slim/Networking/Select.pm line 238)
   frame 4: Slim::Networking::Select::select 
(/usr/local/slimserver/slimserver.pl line 492)

   frame 5: main::idle (/usr/local/slimserver/slimserver.pl line 445)
   frame 6: main::main (/usr/local/slimserver/slimserver.pl line 1071)


FLAC, MP3 and WAV (e.g., transcoding from AAC) still work fine.

I'll keep digging.

--Phil
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Re: [slim] Re: Ogg Encoding Options

2006-12-28 Thread Phil Karn

Phil Karn wrote:

I figured out how to downgrade my Squeezebox 2 to the previous firmware 
version, 64. That does *NOT* fix the problem; Ogg Vorbis files still 
won't play on the 12/28/2006 version of 6.5.1. The log shows the 
following backtrace:


I rolled everything back to 6.5.0 and my Squeezebox 2 and 3 *still* 
can't play any Ogg Vorbis files. Yes, their firmware loads reverted to 
version 64. I even forced a player factory reset.


I'm absolutely sure these files played fine before I installed 6.5.1 today.

Now I'm *really* confused.
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Re: [slim] Re: Ogg Encoding Options

2006-12-28 Thread Phil Karn

Phil Karn wrote:

If the firmware is indeed broken, is there a simple workaround to force 
transcoding of Ogg Vorbis to WAV for Squeezebox 2/3?


I found the workaround: simply uncheck the Ogg Vorbis/Ogg Vorbis line 
in File Formats under Server Settings. Slimserver now transcodes to WAV 
and avoids whatever problem kept the player from decoding Ogg Vorbis.


Still trying to figure out why the original problem appeared, and why it 
didn't go away when I reverted to 6.5.0 and the earlier firmware.

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Re: [slim] Re: Radioio Deadhead?

2006-05-24 Thread Phil Karn

Peter wrote:

Or try this:
mms://radioio-dead-hi.wm.llnwd.net/radioio_dead_hi?MSWMExt=.asf

(note the underscores)


Works for me in 6.2.2. Thanks!

--Phil
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Re: [slim] Re: SB sends no DHCP host name

2005-12-22 Thread Phil Karn

0xdeadbeef wrote:

Unfortunately, in my case the DHCP server is running on my DSL
modem/router and there's nothing to configure there. Though it should
give the SQ the same IP every time due to MAC caching, I have the
feeling it doesn't. Indeed, I observed at least 3 different IP
addresses in the few days since I obtained my SQ.
Then again, all devices whith a DHCP name get the same IP all the time.


You really need to trace the packets to know what's really going on, as 
I know of at least two different ways that a given host might keep a 
stable IP address.


First, the client might cache leases in nonvolatile memory across power 
cycles or reboots and simply request the previous IP address when 
rebooted. A DHCP server will almost always grant a request for a 
specific IP address as long as it's in that server's pool and no other 
client has a lease for it.


Second, the server might cache client MAC addresses and previous leases 
(expired or still valid) and reassign the previous IP address even when 
the client doesn't ask for it.


It's helpful to know the DHCP protocol exchange. A cold start looks 
like this:


Client: DHCPDISCOVER [who can give me an IP address?]
Server: DHCPOFFER a.b.c.d [I'm a DHCP server, and I can offer you this 
address]

Client: DHCPREQUEST a.b.c.d [I'd like to use this address]
Server: DHCPACK a.b.c.d [Okay, you've been officially granted this address]

A booting client that wants to use a previously assigned address will 
jump directly to the DHCPREQUEST message. The server can respond with a 
DHCPACK as before, or it can respond with a DHCPNAK (negative 
acknowledgement), rejecting the request. At this point the client will 
fall back to the DHCPDISCOVER step.


Ideally the Squeezebox would save DHCP leases across reboots, and your 
DHCP server would hand out very long leases to minimize the chances of 
that address being given to another client. But there are other ways to 
ensure that a Squeezebox always gets the same IP address, including 
bypassing DHCP altogether and assigning a static IP address.


That might be your best bet assuming your Squeezebox doesn't move often. 
Just make sure that you set up separate blocks of addresses for static 
use and dynamic assignment, so your DHCP server doesn't hand out the 
address already in use by your Squeezebox. DHCP clients are supposed to 
detect this by ARPing for an assigned address before actually using it, 
but it can fail if the conflicting host with the static assignment 
happens to be offline at the time.


--Phil



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Re: [slim] SB sends no DHCP host name

2005-12-15 Thread Phil Karn

0xdeadbeef wrote:

When the Squeezebox requests an IP adress from a DHCP server, it doesn't
seem to send its own host name.
E.g. in Linux (Debian/Ubuntu), you can send the hostname to the DHCP
server by adding the following lines to /etc/dhclient.conf:

send dhcp-client-identifier 00:00:00:00:00:00;  // enter mac address
here
send host-name host name; // change to your
host name

Since the Squeezebox doesn't send a host name, it's named noname by
the DHCP server, which is kinda ugly.

So I'd suggest to send the host name that was entered in the player
setup.


I do it in a way that doesn't require the Squeezebox to send a host name.

My DHCP server is ISC DHCPD on Linux. /etc/dhcpd.conf has an entry for 
every regular host on my LAN, including Squeezeboxes. Each entry 
specifies the MAC address and a fixed-address entry for that system. So 
each Squeezebox gets the same IP and DNS name address every time it boots.


Here's a sample entry in /etc/dhcpd.conf:

host skinner {
hardware ethernet 00:04:20:06:1a:56;
fixed-address skinner.local.ka9q.net;
}

The zone local.ka9q.net contains A records in the 192.168.1/24 subnet 
for hosts that don't need global IPv4 addresses. skinner.local.ka9q.net 
resolves to 192.168.1.13, so that particular Squeezebox always gets that 
same IP address.


The DHCP server keeps a separate pool of IP addresses for clients whose 
MAC addresses don't have specific entries (e.g., a visitor's laptop).


--Phil
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Re: [slim] SB3 review at Silent PC Review

2005-12-13 Thread Phil Karn

mflint wrote:

Forgive me if this has been mentioned before:

http://www.silentpcreview.com/article287-page1.html


Overall it's a good review, and I'm glad he liked the Squeezebox 3, but 
early on he makes a few erroneous statements that might reduce his 
credibility with other readers. And it's clear that he comes from the 
expensive has to be better than cheap school of audio that has kept so 
many speciality audio companies in business over the years.


--Phil
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Re: [slim] Best way to back up music on Macs

2005-12-13 Thread Phil Karn

Marshall Clow wrote:

So I use a Mac, and most of my music is on a laptop.  Last year I had a
laptop stolen, and had to rip everything again.

So what i sthe best way to back up music on a Mac?  Is there an easy
way to get iTunes to access an external harddrive, or should I just
burn everything to DVDs?


I would burn everything to DVDs - it's not like the music is going to 
change.


I recommend backing up to an external hard drive (or drives) rather than 
(or at least in addition to) burning DVDs.


Hard drives have gotten really cheap. External drives are now less than 
$1/GB, and some internal drives are below half that cost. But even more 
important, hard drives are *much* roomier than DVDs so you don't have to 
keep switching disks during your backup.


Writing to a hard drive is also *much* faster then writing to a DVD. 
Combined, these two features of hard drives makes it much more likely 
that you actually will spend the time to keep your backups complete and 
up to date.


Also consider giving yourself some protection against drive failure with 
RAID. My ripped CD collection sits on a RAID-5 array, and that is in 
turn backed up on offline hard drives. We also keep the original CDs off 
site, making them less of a target for burglars.


Phil
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Re: [slim] The servers that a Squeezebox sees

2005-12-13 Thread Phil Karn

JJZolx wrote:

I recently assigned my Windows XP computer a second IP address and
switched to HTTP port 80 so that I could more easily address the
SlimServer without a port, and so that I could run Apache on port 80 at
another IP address on the same machine.  SlimServer had been at
192.168.9.11:9000 previously.  I'm using the --httpaddr command line
option to designate the IP address.

192.168.9.11 (audrey)
192.168.9.30 (slim) running SlimServer, HTTP port 80

I have two SB2's on my network.  Both see the server named 'audrey'
at the .11 address, but don't see the server at .30.  I had to key in
the IP address to connect.  If I get into the network setup on either
SB, it always shows the nonexistent server.

Is this a bug, with SlimServer incorrectly announcing that it's running
on the .11 IP address?


I suppose it is a bug, but I don't understand the point of what you're 
trying to do. By using the alternate port 9000, a Slimserver can easily 
co-exist with Apache on the same machine with a single IP address.


I haven't checked, but it seems likely that the Squeezeboxes discover 
the slimserver by their zeroconf advertisements. They're made 
symbolically, not numerically, so the Squeezebox that sees a Slimserver 
advertisement will assume that it can connect to it on port 9000.


Again, what's the point of what you're trying to do?
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[slim] Client/server docs

2005-12-13 Thread Phil Karn
Two related questions. I've been looking at the protocols between the 
Squeezebox and the Slimserver with ethereal and I've noticed that the 
protocol specifications bundled with the Slimserver appear to be 
incomplete. I see commands and data structures that aren't documented. 
Is a more recent version available?


If I understand the protocol correctly, the server hands the Squeezebox 
a command that basically says fetch the following URL, play the 
resulting data stream with the following DAC settings, and let me know 
when you're done. (The DAC settings are ignored with MP3 and FLAC 
files, as those settings are embedded in the data stream.)


This seems pretty straightforward and flexible.

During playback the Squeezebox continually lets the server know how it's 
doing, e.g., by reporting buffer utilization, etc, and the server tells 
the Squeezebox exactly what to show on its screen. Even something as 
simple as screen scrolling happens only when the server tells it to.


But I notice that every HTTP 'get' operation is handed back to the 
Slimserver, even when it's an Internet Radio stream. The Slimserver 
fetches the data from the source and relays it to the Squeezebox. I 
suppose that makes it possible for multiple Squeezeboxes to play the 
same stream without duplication over the wide area network connection.


Still, it would be handy if I had a hook to let me pass my own URL 
directly to the Squeezebox, without application-layer relaying through 
the Slimserver. This might be useful if I wanted to use my Squeezebox as 
a dumb D/A converter for some special application such as Internet 
telephony, or driving the Squeezebox DAC directly from a program of my 
own. Does anybody know if such a raw DAC access hook already exists in 
the Slimserver?


Phil
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Re: [slim] Re: 20Kb/sec bandwith used and it's not even playing anything?

2005-12-13 Thread Phil Karn

seanadams wrote:

Who cares about 20Kb/s on a LAN?


Nobody, at least not on a LAN. But it can be a problem if your 
Squeezebox and its server are separated by a shared wide area network.


The Squeezeboxnetwork server appears designed to minimize network 
overhead, e.g., by updating the clock only every minute.


I once took my Squeezebox to work and had it tunnel back home over SSH. 
It worked, but the overhead was rather high and I didn't want to put 
that much continuous load on my DSL uplink. It seems better to just take 
your music with you.


Phil
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Re: [slim] Weird DHCP bug on Slimserver 3 with latest firmware

2005-12-08 Thread Phil Karn

kdf wrote:

I have 192.168.1.34 as a static assignment for the mac address of the 
player (listed on the bottom and in player settings).  however, the 
lease table shows an entry for 20:05:73:00:00:0C and has been given 
192.168.1.100.  i have 100-149 as a 7-day dynamic block of leases.


Whether DHCP works under these conditions may depend on the particular 
DHCP server you're using.


My packet traces showed that the Squeezebox DHCP requests were being 
sent with the correct MAC address in the Ethernet source field, but the 
hardware address field in the DHCP request packet was the funny one you 
observed (20:05:...).


My ISC DHCP server running on Linux then answers with a unicast response 
to the funny (20:05: ...) address, and since the Ethernet controller 
in the Squeezebox is apparently not listening for that address, it never 
gets the response. So eventually the DHCP attempt fails and the 
Squeezebox reverts to a self-assigned address in the 169.254 block.


But it's my understanding that it's legal for DHCP servers to broadcast 
all of their responses (i.e., to the Ethernet broadcast address 
ff:ff:ff:ff:ff:ff). This lets multiple DHCP servers serving the same 
subnet to easily observe each others activities. If this happens then 
the Squeezebox, which is also presumably monitoring broadcasts, will get 
its DHCP responses and complete the exchange.


I'm guessing that's what's happening on your network, but without packet 
traces I can't say for sure.


I suppose I could see if my DHCP server can be configured to broadcast 
its responses, and to try some more packet tracing experiments to flesh 
out the details of just how the SB behaves with bridging enabled. But as 
soon as I saw that it was munging frames I turned it off just so I could 
get my box playing music again.


I don't have an immediate need for the bridging feature, but it would be 
nice to have it working someday. 802.11g bridges are surprisingly 
expensive, so it's a nice feature to have in the Squeezebox.


Phil
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[slim] Weird DHCP bug on Slimserver 3 with latest firmware

2005-12-07 Thread Phil Karn

I just received a pair of Slimserver 3s and opened one of them up.

When I configured it for my wireless network, it quickly configured 
itself with DHCP and downloaded new firmware, version 28. It then came 
back up, auto-configured itself again and worked fine for a while.


Then I unplugged it to move it into another room. I also enabled 
Ethernet bridging, to try it out.


This time the unit was unable to complete DHCP.

Some packet tracing showed why. The unit sent out a DHCP request with 
its factory-assigned MAC address 00:04:20:06:1a:50 in the Ethernet 
source field but the sender hardware address in the DHCP request was the 
unusual value 20:05:73:00:00:0c. When my DHCP server responded, it put 
this second MAC address into the destination field. Since the Squeezebox 
was presumably listening for its factory-assigned MAC address, it never 
heard the response. So it kept resending its request and my server kept 
sending the same unheard answer.


Eventually the Squeezebox timed out of DHCP and gave itself a 
self-assigned IP address in the 169.254 block, which I do not use.


If I manually set the appropriate IP addresses, all works fine. Only the 
DHCP client appears to be broken.


The DHCP server is ISC dhcpd running on Linux. It works fine with 
everything else in my house.


Anybody else see anything like this?

Phil
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[slim] Re: Weird DHCP bug on Slimserver 3 with latest firmware

2005-12-07 Thread Phil Karn
A followup: I forced a reset to factory defaults (by holding down the 
ADD key and cycling power) and the problem I encountered with DHCP 
disappeared. DHCP worked fine.


Then I turned on Ethernet/WiFi bridging, and the problem appeared again. 
Neither the Squeezebox nor the Powerbook I have plugged into the 
Ethernet port on the Squeezebox can do DHCP.


I see some broadcast traffic from my wireless network coming through on 
the Ethernet on the Powerbook, but I can't seem to connect to anything 
on my home network with either IPv4 or IPv6.


Packet traces at the DHCP server seem to show that the Squeezebox is 
writing its own MAC address over the source address fields of the 
packets it is relaying from the Ethernet, something a bridge shouldn't 
do (bridges are transparent to MAC frames). And when bridging is 
enabled, the Squeezebox's own DHCP packets have the correct MAC address 
in the Ethernet link level source address field, but carry a bogus MAC 
address in the DHCP application-level packet so the Squeezebox doesn't 
hear the DHCP server responses.


Basically, it appears that Ethernet bridging is completely broken, and 
enabling it disables the Squeezebox as well.


Again, this is firmware version 28 on the Squeezebox 3.

Phil
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Re: [slim] Weird DHCP bug on Slimserver 3 with latest firmware

2005-12-07 Thread Phil Karn

kdf wrote:

This sounds a lot like this report:
http://bugs.slimdevices.com/show_bug.cgi?id=2221


Thanks for the pointer. This is indeed exactly the problem I'm seeing.

That report specifies firmware version 22. I'm running 28, so apparently 
the bug has been in several versions. Is version 28 the latest?


i've had the same issue, as far as MAC.  The difference in my case it 
that it does eventually get a dynamic IP, and works.  The router has a 
static IP assignment set up, but that is not the resulting IP.


I assume this was with Ethernet/WiFi bridging enabled?

When you say that you eventually get a dynamic IP address, is it in the 
block you normally use on your network, or is it in the 169.254 block? 
That's the zeroconf address block set aside for self-assignment by 
hosts when they're unable to get an answer from a DHCP server, and the 
Squeezebox seems to follow this convention as well.


Your network might have been set up to work with 169.254 addresses as 
well as your usual block, so that may be why it seemed to start working.


--Phil
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Re: [slim] Obtaining mp3s/flac/whatever legally

2005-10-20 Thread Phil Karn

max.spicer wrote:

I've got a list of tracks that I'd like to own, but don't necessarily
like the artists enough to go out and by the relevant albums (yet). 


I can't help you if you've got a list of specific tracks by artists on 
RIAA labels. But if you're looking for a good way to obtain MP3, FLAC 
and other digital music files without DRM, and with very liberal copying 
and usage rules, check out Magnatune (www.magnatune.com).


They're *not* a music service like iTunes or emusic. They're an 
independent record label, that happens to do business very differently 
than most.


You will *not* find well-known acts on Magnatune, as they can only sign 
new artists that haven't already signed away their souls to a regular label.


Sure, a lot of what Magnatune has doesn't really interest me, but that's 
just as true (if not more so) for the big, evil RIAA labels. But you 
probably will find *something* you like.


Phil
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Re: [slim] mDNS broadcasts on all IP addresses

2005-08-15 Thread Phil Karn

slimbls wrote:

Hi, slimserver newbie here...I have a suse 9.2 linux system that's my
firewall and media server (please, no bottle tossing this way).

After I installed slimserver (6.1.1) I asked it very nicely to only use
my internal ethernet, but I'm still seeing UDP packets from myself on
port 5353 to 224.0.0.251.

How can I nicely tell mDNS to not broadcast on my external ethernet
adapter besides a couple of lines in iptables?


mDNS stands for multicast DNS and that's how it works. Multicasting is 
a lot like broadcasting, but scales much better because you can be 
selective about who receives your packets.


mDNS is designed to let you find services and hosts on your local LAN 
without needing a server on your local network. Requests for IP 
addresses and services are multicast on the local LAN. Whichever machine 
has the requested information will respond, and the others will ignore 
it. mDNS is currently used most heavily by Mac OS X, which calls it 
either Rendezvous or Bonjour, and there are ports of mDNS to Linux.


224.0.0.251 is the standard multicast destination address for a mDNS 
packet. Your router won't forward multicast packets unless it's 
specifically designed and configured to be a multicast router, which it 
almost certainly won't. Otherwise it'll just ignore them, and they'll 
stay on your local network. So there's nothing to worry about.


Phil


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[slim] Source for bin programs?

2005-08-15 Thread Phil Karn
Where can I find the source for the various binaries included in the 
Slimserver package?


I'm particularly interested in the source for the mDNSResponderPosix 
program, as I'm trying to set up mDNS on all my Linux machines to make 
them play better with our Macs. It appears that the Slimserver perl 
script writes entries for three services into the mDNSResponder's config 
file and then starts it. I'd rather start a separate system-wide 
mDNSResponder daemon so I can easily configure it to advertise all the 
services on my Linux box, not just the Squeezebox stuff.


Do the Squeezeboxes actually use the mDNS stuff in any way, or is it 
just there to make the server's web pages visible to Macs that happen to 
be browsing the network with Bonjour/Rendezvous?


Phil
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Re: [slim] AAC playback background noise. Slimserver6/FreeBSD

2005-05-07 Thread Phil Karn
icestorm wrote:
However...When I try to play AAC music from my FreeBSD server, (which
is most of my music collection), I can hear a sound like a constant
wind in the background, which becomes intolerable when I  increase the
volume. Please note that I can hear the song fine! This problem does
not happen with mp3 encoded music or radio.
I seem to be having a somewhat similar problem.
I just set up AAC decoding on slimserver 6.0.2 on Linux. It works fine 
with my original Squeezeboxes, but when playing on a Squeezebox2, the 
audio breaks up into loud noise about a second into each AAC song.

I just isolated the problem. It only occurs when two conversions are 
done in tandem, e.g., AAC-PCM-FLAC or AAC-PCM-MP3. This is why the 
problem only appeared on the Squeezebox2, because the transcoding from 
AAC to FLAC is done in two steps. The original Squeezebox doesn't 
support FLAC, so faad converts directly to WAV.

To verify this theory, I set bit rate limiting on one of my original 
Squeezeboxes to force transcoding to MP3. Sure enough, the same problem 
appears.

I don't yet have a clue as to *why* this is happening. But it does, 
every single time. It seems almost as if a byte of raw PCM is somehow 
being dropped in the pipe between the two coders, causing a 
mis-synchronization that sounds like loud noise. Whether that's actually 
happening, I don't know yet.

--Phil
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Re: [slim] Noise Spike at Song Startup

2005-05-07 Thread Phil Karn
warc1 wrote:
have experienced. The most annoying is a loud noise spike that occurs
during the start of some songs when playing a long playlist. 
Is the spike really, really short and sharp, and right at the beginning 
of the track?

If so, this could be the sound of the WAV header on the front of the 
file being misinterpreted as raw PCM.

--Phil
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Re: [slim] Apple Lossless

2005-05-07 Thread Phil Karn
Frank Xavier Ledo wrote:
First, It seems that all songs start out with a pop.  I am guessing that 
the command line options I set in convert.conf were incorrect and are 
resulting in a linefeed or space at the start of the song.

What I used is:
mov wav squeezebox *
[alac] $FILE$
As a test I tried using alac to convert the file from *.m4a to *.wav, and 
added the wav file to my playlist.  There were no pops.

Second, alac supports PCM output.  Is there any benefit to using this 
instead of wav output?
The pop is almost certainly the sound of the WAV file header being 
misinterpreted as raw PCM audio. Try selecting raw PCM output from alac 
and see if the problem goes away.

Are you sure you're using the convert.conf entry you gave above, and 
aren't actually transcoding to another format?

In the convert.conf file entries that pipe two codecs together to 
transcode from one compressed format to another, the two codecs must 
agree on their interchange format. The existing convention *seems* to be 
raw little-endian 16-bit stereo PCM *without* a WAV header. If the first 
program outputs wav format when the second expects raw PCM, you'll get 
the pop.

I'm not exactly sure which native format the Squeezebox 1 expects, WAV 
or raw PCM. Anybody know?

--Phil
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Re: [slim] SB1 buffer size

2005-04-08 Thread Phil Karn
Sean Adams wrote:
When I started designing the Squeezebox1 hardware in 2002, I fully 
realized that an enormous buffer would be desirable. However, the 
architecture (which was chosen based on a long list of goals including 
ethernet/wireless capability, ability to drive a graphic VFD, OS 
performance, etc) unfortunately did not have an SDRAM interface, only 
SRAM, which meant that a very large (several MB) buffer would not be 
possible. Dean and I did a great deal of testing and never saw any 
problems on wired ethernet under any OS. On wireless networks there were 
some issues at first, which were quickly eliminated, most of them 
anyway, with some careful tuning of the TCP stack.
I understand your design constraints. Memory is nice, but it can also be 
expensive and you don't want your box costing more than it really has 
to. That's quite reasonable.

What we found was that in the real world there were still a few 
problems. I spent many weeks studying packet dumps and optimizing TCP 
(http://www.seanadams.com/ip2k), testing on Windows, Mac, and Linux, 
getting feedback from customers, and making improvements.
Yup. Many stacks do have glitches, although I think the later Linux 
stacks behave very well, especially when the entire path is just a few 
high speed Ethernet hubs. A bigger issue, I think, are the OS and disk 
scheduling algorithms.

It is no secret that a bigger buffer would be nice - in fact I think 
this request has been in our public bug database since its inception. 
Unfortunately the only way we can make the buffer larger is to make new 
hardware with a larger buffer, which is exactly what we've done. 
Optimizing streaming performance is still an active goal.
But I don't think that buffer *has* to be huge. If it does, then that's 
a symptom of brokenness someplace else, specifically in the server and 
the way it is treated by the operating system on which it runs.

PCM streaming on Squeezebox1, even with its 256K buffer capacity, works 
as advertised in nearly all environments, unlike my car which can only 
reach its advertised top speed on less than 0.01% of the road surface in 
the USA. I realize this is a retarded analogy and not a kind response 
but consider this: we are more open than any hardware company (that I 
know of) in terms of admitting where there's room for improvement 
(bugs.slimdevices.com). We view this information not as a liability, but 
as priceless guidance in how to make our products better.
Um, I assume you didn't mean that as a serious analogy! A better analogy 
would be to the per-mile probability of my car suddenly and without 
warning, seizing up and stopping dead in the middle of the freeway. 
Then, after a few seconds, the tires screech and it suddenly lurches 
back up to 65 mph just as unexpectedly. :-)

Certain issues are simply not addressable in software (802.11g support, 
as another example). For those issues we develop new hardware.
Understood. Since you advertised those hardware limitations, I can't 
complain. Your transcoding scheme to optionally limit the network bit 
rate to something it can support was a very nice touch.

Finally, as protection we provide a 30-day no questions asked return 
period. If the product doesn't work the way you want it to, or even if 
you just have a change of heart for some reason, we'll take it back.
Well, that's problematical when the problem is in software, not 
hardware. Again, I have no complaints about the Squeezebox hardware, 
either version 1 or version 2. If the hardware were broken, mis designed 
or otherwise unusable, then it would be easy to decide to return it in 
the 30 day period.

But the hardware is all fine; all of the problems I've encountered have 
been in server software, and that makes returning the unit a much harder 
call. You never know whether a given problem that you consider really 
serious will be fixed tomorrow or next week or next month or next year. 
Believing strongly as I do in the power of open source, I have decided 
to keep all three Squeezeboxes and wait for the software to improve. And 
that can be frustrating. Yes, I could roll up my sleeves and jump in and 
hack the code myself, but the Slimserver package is already a very large 
chunk of code that presents a dauntingly steep learning curve to anyone 
who wants to modify it. Confronted by it, I'd be strongly tempted to 
start over from scratch, or by adapting existing, mature mechanisms that 
I already know to work well (e.g., adapting the standard printer 
spooling system to manage queues and play lists for a Squeezebox.)

I don't know what else to say - we vastly improved PCM streaming 
performance by adding a huge buffer, adding 802.11g wireles support, and 
adding lossless compression. At the same time we added a whole slew of 
features in SlimServer 6 which improve SLIMP3, a product which we began 
selling out of my garage in 2001, and of which we have not sold a single 
unit since 2003. I realize that these are not huge 

Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-04-08 Thread Phil Karn
Robin Bowes wrote:
Can you show me where this claim is made? I'd be very surprised if this 
is true. For a start, the wired port on the SB1 is only 10MB/s (I can't 
find a spec. sheet to confirm this, but the last point here [1] says:

 - faster 100Mbps wired ethernet interface
I do believe you're right; it's 10 Mb/s on the SB1. I checked it by 
plugging it directly into my PowerBook and looking at the interface 
settings, and also by plugging it into a different hub that indicates 
link speed on the LEDs.

But this doesn't really matter. A raw, uncompressed PCM audio stream 
still takes up only 14% of a 10 Mb/s Ethernet port, so there's no reason 
it should cause any kind of playback interruption unless it's used on an 
ancient 10 Mb/s non-switched hub with lots of contending traffic. I got 
rid of all my hubs years ago. Every network port is switched and 
supports full duplex up to 100 or 1000 Mb/s, depending on the specific 
switch. The Linux box running SlimServer has a 1GB/s connection, and 
nothing (server or network) is ever loaded very heavily.

[1] http://www.slimdevices.com/su_faq.html#about2-difference
I've got a wireless SB1, and until recently, I had no problems with 
dropouts caused by network bandwidth (I too play flac files decoded on 
the server and streamed as raw PCM).
I have two SB1s, but both are wired. I wanted to wait until 802.11g was 
supported before buying a SB, so I got one of the SB2s with wireless.

But my playback interrupt problems were all over wired paths. I would 
have expected problems on 802.11b, but not on an all-wired path with 
plenty of capacity.

I agree with the point you make about threading and making the streaming 
part of slimserver work better, but I don't think the SB or slimserver 
is particularly at fault in causing your dropouts - there must be 
something in your environment that is causing the problem. Maybe the 
port you've got the SB plugged into is not auto-detecting the 10Base-T 
link? Have you checked?
I'm quite sure all my LAN switches are working fine. The playback 
interrupt problem is clearly due to sub optimum task scheduling on the 
server. If I renice the slimserver and related tasks up a few points, 
the playback interruptions disappear, but at the expense of slowing down 
everything else on the server whenever the Slimserver wants to do some 
sustained maintenance activity like scanning the music filesystem and 
rebuilding its database. The Slimserver code should run at high priority 
only in its real-time-critical sections, i.e., those directly involved 
in playing music. Everything else (e.g., the UI) should run at normal 
priority. Database reconstruction could even run at below-normal 
priority, just to be nice to the other users.

Another way to minimize playback interruptions is to use lots of read 
ahead buffering inside the Slimserver playback path. That covers you in 
case the disk is suddenly pre-empted by another urgent I/O-intensive 
application. And if you can decompress and buffer up a lot of raw PCM 
inside the server, then you're also covered in case a high priority 
compute-bound job comes along and pre-empts the FLAC/Ogg Vorbis/AAC/etc 
decompresses (which are at least slightly CPU-intensive). If you can 
manage to buffer up a lot of raw PCM ready to transmit, then even if the 
server is busy running other applications it should be able to spare the 
few quick interrupt service calls needed to move that PCM to the 
Ethernet transmitter and then to the Squeezebox.

Obviously if the system running Slimserver doesn't have, on average, 
enough disk I/O cycles and CPU cycles to supply a full-rate stream to 
the Squeezebox, then none of this will help much. But there's no reason 
why a fast, lightly loaded machine like mine should have frequent 
problems keeping the Squeezebox happy even when the occasional unrelated 
load-burst comes along.

Phil
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[slim] Re: SB2 startup glitches

2005-03-30 Thread Phil Karn
Additional comment: the crunching during startup does *not* occur with 
the original SB running off the same v6.0.0 server. The problem happens 
only with the SB2, and only when playing FLAC or OGG, not MP3. Haven't 
tried other file types.

Phil
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Re: [slim] kinks in the SB2?

2005-03-30 Thread Phil Karn
Phillip Kerman wrote:
First, mine goes blank for no apparent reason sometimes.  Blank screen.
I see that too. I don't think the unit actually crashes, but it can be 
disconcerting.

Also, I'm trying to compare the digital out into my DAC to the internal DAC
but the digital out can't stay locked (I think that's it... my receiver
keeps reseting like it used to between PCM and MP3).
I wonder if this is could be related to the crunchy distortion I 
reported last night with my own SB2. I can't confirm it, but it very 
definitely sounds like the firmware in the box is starving the DAC when 
it has to handle the sustained, high speed burst of incoming traffic 
that fills up the playback buffer with raw PCM data. If the DAC is 
indeed being intermittently starved, perhaps this is killing the clock 
on the S/P-DIF output and making it difficult for your external DAC to 
maintain lock.

Does the problem happen on all files, or only those transcoded to raw, 
high speed PCM? And does it persist through the entire track, or does it 
only happen at the beginning? The crunchy distortion I notice on my 
SB2 only occurs on raw, high speed PCM, and only  until the buffer 
fills. When that happens and the server backs off to its maintenance 
speed of about 1.4 megabits/sec, the distortion goes away.

It just occurred to me that I could try forcing the Ethernet down to 10 
megabits/sec and see if the problem still occurs. If it's indeed due to 
DAC starvation from lack of proper CPU real-time management, this might 
work around the problem by limiting the peak speed of the buffer-fill 
burst to only 10 Mb/s.

--Phil
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-23 Thread Phil Karn
Jack Coates wrote:
http://www.google.com/search?hl=enlr=c2coff=1client=firefox-arls=org.mozilla%3Aen-US%3Aofficialq=linux+encrypted+swap+partitionbtnG=Search 
I'm aware of the various ways to encrypt a swap partition. They are 
obsolete in this age of cheap, large RAM modules.

Again, a fun theoretical argument -- or are you saying this box has a 
lot of untrusted root-level users who are frequently working in shells 
on it? I know my server won't let me do anything of the sort without 
being root.
No, I'm the box's only interactive user, though it also runs web and 
mail services that my wife also uses. However, the machine sits in my 
house, not a co-lo, and is thus potentially vulnerable to physical 
theft. I use encryption as appropriate to protect my employer's 
information and other sensitive information, but that can be rendered 
ineffective if encryption keys can persist on an unencrypted swap device.

OK, I give -- the key here, as you identify in another message is that 
you're playing FLACs, and with that fact I think there's just not a lot 
to do except for SB2 and its bigger buffer.
Yes, clearly things are stressed by my large collection of FLACs and the 
relatively small buffer in the SB1. But...

Buffer starvation could 
theoretically be slowed or alleviated by reducing other tasks. For 
instance, I bet your vmstat looks like a few miles of bad road when 
you're browsing those 17K tracks on two ATA drives through the Web UI or 
rebuilding the cache. In theory a bunch of componentry could be isolated 
off into other processes so it would be non-blocking, but in the end 
your performance would still be questionable (UI and streaming and data 
management rely on each other and are going to block, sooner or later). 
You could try prioritizing traffic (tc) and processes (nice), but at the 
end of the day I'm guessing that it's probably not going to work unless 
you transcode to a lossy format on the fly.

This will work not just in theory, but also in practice. I have a fair 
bit of experience in networking and real-time applications, and I'm 
convinced that carefully restructuring and reprioritizing the server 
tasks and threads responsible for real-time playback -- and separating 
them from non-real-time operations such as managing the database and 
driving the UI -- could completely eliminate *all* playback interrupt 
problems, even with my many big FLAC files and a small SB1 playback buffer.

My LAN is a combination of switched 100-Base-T and 1000-Base-T segments 
that are never loaded anywhere near capacity. (Yes, it's also quite 
reliable; I've measured my packet latency and loss rates, and they're 
always near zero or zero, respectively.) My server has plenty of all the 
required resources (disk I/O bandwidth, CPU cycles and RAM) to decode a 
FLAC file and buffer the raw PCM far ahead of the playback pointer in 
the SB. So there's simply no fundamental reason why playback 
interruptions should ever occur.

The problem with the existing server, just as you point out, is that the 
real-time and non-real-time server functions are bundled into the same 
tasks and threads, and insufficient attention has been paid to meeting 
real-time-critical playback requirements. Yes, the bigger buffer in the 
SB2 will definitely help (I have one on order) but I disagree that it's 
the only way or even the best way to solve the problem.

Phil
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-21 Thread Phil Karn
Jack Coates wrote:
You might want to do some research on modern virtual memory 
management... swap is necessary no matter how much RAM you have.
I disagree. I see no reason for a swap partition when you already have 
far more physical RAM than most swap partitions used to be, and tasks 
never fail because of memory exhaustion.

Swap partitions are also a potentially serious security hazard; 
passwords, encryption keys and other sensitive cleartext information 
have been known to persist on them for years.

Anyway, 
I haven't seen how many tracks you have, but that machine is about four 
times more than the average Slimserver installation runs on.
As I just posted in another message, I count 17,115 files in two 
separate 200GB filesystems built from RAID-5 partitions spread across 5 
300GB SATA disks, with an additional disk as a hot spare.

homer$ vmstat 1 25
procs ---memory-- ---swap-- -io --system-- 
cpu
 r  b   swpd   free   buff  cache   si   sobibo   incs us 
sy id wa
 1  0  0  58992 298524 11343720029 8   2333 43 
 1 56  0
 0  0  0  58992 298524 113437200 0 0 1023  1054  0 
 0 100  0
 0  0  0  58992 298524 113437200 036 1028  1073  1 
 0 100  0
 0  0  0  58992 298524 113437200 060 1062  1092  0 
 0 100  0
 0  0  0  58992 298524 113437200 0 0 1021  1059  1 
 0 100  0
 0  0  0  58992 298524 113437200 0 0 1020  1059  0 
 0 100  0
 0  0  0  58992 298524 113437200 0 0 1018  1048  0 
 0 100  0
 0  0  0  58992 298524 113437200 036 1031  1088  0 
 0 100  0
 0  0  0  58840 298524 113437200 0   105 1045  1080  0 
 0 98  1
 0  0  0  58840 298524 113437200 0 0 1021  1051  0 
 0 100  0
 0  0  0  58840 298524 113437200 0 0 1025  1055  0 
 0 100  0
 0  0  0  58840 298524 113437200 0 0 1020  1051  1 
 0 100  0

[this is a little unusual in that I normally have two copies of 
[EMAIL PROTECTED] running -- this is a hyperthreaded 3.2 GHz P4 -- but the Boinc 
server has been off the net for several days, and my machines have run 
out of work. Those tasks are always niced to 19 and consume minimal memory.]

homer$ more /proc/1410/status
Name:   slimserver.pl
State:  S (sleeping)
SleepAVG:   98%
Tgid:   1410
Pid:1410
PPid:   1
TracerPid:  0
Uid:0   10080   1008
Gid:0   0   0   0
FDSize: 256
Groups:
VmSize:98800 kB
VmLck: 0 kB
VmRSS: 95240 kB
VmData:94804 kB
VmStk:84 kB
VmExe:   996 kB
VmLib:  2756 kB
VmPTE:   132 kB
Threads:1
SigPnd: 
ShdPnd: 
SigBlk: 
SigIgn: 00011080
SigCgt: 80004007
CapInh: 
CapPrm: feff
CapEff: 
homer$ top - 17:34:46 up 5 days,  6:08,  1 user,  load average: 0.00, 
0.01, 0.00
Tasks: 110 total,   1 running, 109 sleeping,   0 stopped,   0 zombie
Cpu(s):  0.3% us,  0.0% sy,  0.0% ni, 99.2% id,  0.3% wa,  0.0% hi,  0.2% si
Mem:   2076460k total,  2019228k used,57232k free,   298616k buffers
Swap:0k total,0k used,0k free,  1134960k cached

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND 

 1410 slimserv  15   0 98800  93m 2204 S  0.7  4.6  77:48.69 
slimserver.pl
 1356 mysql 16   0  127m  15m 2804 S  0.0  0.8   0:00.54 mysqld 

 4082 karn  15   0 10308 7476 3884 S  0.0  0.4   0:02.74 emacs 

 1467 asterisk  18   0 38568 5852 3580 S  0.0  0.3   0:01.08 asterisk 

20875 proxy 15   0  7648 5360 1628 S  0.0  0.3   0:01.77 squid 

 1785 www-data  15   0  6808 4236 2868 S  0.0  0.2   0:00.03 apache-ssl
[...]
[obviously slimserver has quite an appetite for memory. However, it's 
still only a small fraction of the 2GB on the system.]

I note several other slimserver related tasks on the system:
homer$ ps -elf | grep slimserver
5 S slimser   1410 1  1  75   0 - 24700 -  Mar16 ? 
01:17:52 slimserver
0 S slimser   1443  1410  0  76   0 -   406 -  Mar16 ? 
00:00:00 
/usr/local/slimserver/Bin/i386-linux-thread-multi/mDNSResponderPosix -n 
SlimServer -t _http._tcp -p 9000
0 S slimser   1444  1410  0  76   0 -   406 -  Mar16 ? 
00:00:00 
/usr/local/slimserver/Bin/i386-linux-thread-multi/mDNSResponderPosix -n 
SlimServer -t _slimhttp._tcp -p 9000
0 S slimser   1445  1410  0  76   0 -   406 -  Mar16 ? 
00:00:00 
/usr/local/slimserver/Bin/i386-linux-thread-multi/mDNSResponderPosix -n 
SlimServer -t _slimcli._tcp -p 9090

Apparently the clients are currently all quiet (I'm not at home).
--Phil
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-20 Thread Phil Karn
Jason wrote:
The overwhelming majority of users do not seem to have the problems you have
with the server crashing or grinding to a halt.
It sounds like it might be worth your time to try Slimserver on another
machine. 
Perhaps I did overstate the unreliability of the 5.4 code. What I said 
certainly *does* apply in spades to all of the 6.x versions I've tried 
recently. None of them were at *all* usable.

5.4.0 does usually stay up for a day or two, and as long as I don't try 
to add stuff to the database it usually works modulo occasional, long 
and unexpected pauses in responding to even trivial commands from the 
remote control or the web interface, and unexplained and sometimes long 
interruptions while playing back FLAC files over a 100 Mb/s wired 
connection. Renicing the tasks up a few points usually gets rid of the 
pauses, but only at the expense of impairing other interactive processes 
on the same machine whenever slimserver does crunchy things like 
rescanning the music database.

Speaking of which, when I add something to the database, or merely 
change a FLAC tag, I don't even bother with the rescan button as I 
know it probably won't work correctly. I just blow away the database and 
rebuild it from scratch.

This is an otherwise solidly reliable 4U rackmount 3.2 GHz Pentium 4 
running Linux 2.6.11, with 2 GB of ECC memory and 2.3 TB of EIDE and 
SATA disk storage organized into RAID-1 and RAID-5 arrays. With that 
much memory, I don't bother with a swap partition, so VM thrashing is 
not a problem. The Antec box is loaded with big cooling fans, and 
internal temperatures are well controlled.


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Re: [slim] SB2 really doesn't support 96kHz PCM on S/PDIF?

2005-03-20 Thread Phil Karn
Richard Elen wrote:
First, with the vast majority of users storing their music in 
lossy-compressed formats with sample rates seldom exceeding 44.1 kHz and 
often lower, this may be moot for a lot of people.
Agreed.
However with the 
advent of lossless compression format availability in the system, very 
likely those of us who care about audio quality will jump on them, and 
if we do so, we might feel that the ability to handle sample rates above 
48 kHz would be an advantage.
This is not obvious.
But as we cannot hear above about 20 kHz, honestly, at the best of 
times, except perhaps when the level is above about 120 dB SPL at which 
point we certainly can't tell the pitch, what's the point of higher 
sample rates?
Good question.
The traditional reason for proposing higher rates was due to analogue 
filtering in the conversion process. With an audible band ending at, 
say, 20kHz, and a Nyquist limit (the highest frequency you can capture 
with a system, which is half the 'carrier' frequency) at fs/2, eg 
22.05kHz, this meant only a small bandwidth to get signals out of the 
way before they started aliasing and imaging (frequencies above the 
Nyquist limit are intepreted as frequencies below - see 
http://www.apogeedigital.com/pdf/apogeeguide.pdf for more details).

This meant very steep filters that were initially in the analogue 
domain. As a result they cause all kinds of problems such as ringing, 
and phase errors all the way down to below 10kHz. This is why early 
digital recordings sounded clangy and brittle (we also didn't know about 
the audibility of jitter then either).
While it's true that sharp reconstruction filters were difficult to 
implement in analog hardware, this problem -- if it ever *was* a problem 
-- hasn't really been around for almost two decades now. Every modern 
DAC uses oversampling techniques to implement the high-performance 
part of the reconstruction filter in DSP, requiring only a simple and 
cheap analog filter to complete the job.

That said, there was *never* any serious evidence that the 
much-discussed phase shifts of the early analog reconstruction filters 
were *ever* audible in any controlled listening environments. The effect 
was comparable to tiny position changes of a tweeter vs a midrange 
speaker in a cabinet. I certainly heard all those claims about 
brittle-sounding digital audio, but never did I see one based on 
solid, scientifically controlled evidence.

The solution was to get the filters way out of the way by multiplying 
the sample rate - multiplying by integers was simplest and produced the 
fewest artefacts. But you don't really have to work at the higher 
rate; you can instead 'oversample', where you interpolate imaginary 
samples between the real ones, the Nyquist limit is way up there and 
there is tons of room for filtering - which you could also do digitally 
by now and avoid artefacts by and large anyway. Virtually all converters 
used today do many times oversampling and as a result this problem has 
gone away.
Exactly. This is sometimes cited by the golden ears as a vindication of 
their claims -- why would the manufacturers do it if it wasn't necessary 
-- but the real reason is that it's simply cheaper and easier to 
oversample and filter in DSP than to do it with lot of expensive analog 
components. We just happen to be lucky that the cheaper way is also the 
better way in some sense. Of course, the vendors knew a marketing 
windfall when they had one, so they made lots of hay over their 
oversampled DACs. But they never made one whit of audible difference. 
They just saved the vendors a few pennies per unit in component costs.

The other argument for high sample rates is that there may be stuff 
going on up there even if we can't hear it, and we really ought to 
capture everything - especially if we are archiving the material and 
think that inaudible stuff might just be important one day. It's also 
possible that inaudible frequencies might interfere with each other to 
produce something audible - for example if you close-mic violins and 
record each mic on its own track (please don't), not capturing the 
ultrasonics might adversely affect the timbre when you mix, which would 
not have been the case if you had used more distant mics and the mixing 
of ultrasonics had happened in the air between the musicians and the mics.
That basically argues that intermodulation distortion is a good thing, 
the effects of which we should preserve. Seems very strange to me, 
especially since we already agree that the components involved in the 
intermodulation are not directly audible. So if we were to eliminate 
those ultrasonic components by low pass filtering before A/D conversion, 
the sole effect would be the elimination of intermodulation products in 
the audio range that wouldn't exist if the intermodulation distortion 
weren't present.

Right. Makes perfect sense. Not.
The fact is, however, that the human hearing capability can be satisfied 

Re: [slim] Prefered Archiving Format

2005-03-17 Thread Phil Karn
Robin Bowes wrote:
Advice: Make sure you safeguard your collection against HDD failure in 
one or more ways. For example:

 - buy an external 400GB HDD and copy your rips onto that
 - buy another internal 400GB HDD and mirror your first disc.
 - burn your rips to DVD
 - stream your rips to high-capacity tape
 - you get the gist?
   - Use RAID in addition to external backups. RAID-5 is a good 
alternative to RAID-1 that lets you get by with less overhead on larger 
disk arrays. RAID-5 is well suited to relatively static data like a 
music collection.
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-16 Thread Phil Karn
Pat Farrell wrote:
The problem with open source is that the contributors want to 
do the cool stuff, especially when it is pretty easy.
Redesigning the basic engine is hard, slow, and has zero sizzle
Well, most people would say the same thing about an operating system 
kernel. Yet Linux routinely runs for months without panicking.

So 6.* is the first major redesign, and 7.* will likely be
a lot more like you'd want.
I'm beginning to think whether maybe it's time to think about a complete 
server rewrite from scratch, probably in C rather than Perl. The menus 
would be ugly and simple, but the system would actually *work* without 
being restarted several times per day, without unpredictable 5 second 
pauses during playback, etc, etc.

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Re: [slim] Apple Lossless vs. FLAC

2005-03-16 Thread Phil Karn
Todd Larason wrote:
Another option would be to run a server which speaks iTunes' sharing
protocol.  I haven't paid much attention to this space since doing the initial
work figuring out the protocol, so I'm not sure if any of the pre-packaged
servers would do quite what you want.  The perl module 'Net::DAAP::Server'
might be a good place to start looking, or daapd[1].
[1] http://www.deleet.de/projekte/daap/daapd/index.html
Thanks for the pointer.
I'm trying to figure out, though, why we need yet *another* network file 
system protocol, this one just to share music. Is there some essential 
functionality in DAAP that I can't somehow provide over NFS, AFS, SMB or 
even http?

Or is this just Apple up to its old tricks? Maybe I was wrong when I 
thought that with Mac OS X, Apple had finally seen the wisdom of 
building on open platforms.

Phil
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-16 Thread Phil Karn
Danny Rego wrote:
Ummm...as does Windows XP (although the more geek-sided may refuse to 
believe that)what's your point?
Slimserver is open source, so Linux is the closer analogue. Windows XP 
is 1) proprietary and 2) owned by Microsoft, so it has its excuses for 
being so unreliable.

Making slim server less user-friendly/ugly would be a very dumb move. 
Besides, after a bit of tweaking off the bat, slimserver runs reliably, 
unless you are one that enjoys downloading the latest nightlies...then 
you're asking for it.
Little is less user friendly than a massive package of Perl code that 
crashes frequently in a mass of cryptic Perl error messages. Or just 
grinds slowly to a halt for no apparent reason, gobbling all CPU time 
until it has to be restarted manually.

This is on a Linux machine with ECC memory and RAID disks that otherwise 
runs flawlessly for months. So it's not hardware.

v5.4.1 runs beautifully on my Windows XP machine...so on a decent linux 
machine, I would only imagine how smooth it would run.
I see various nightly versions of 5.4.x, but nothing officially labeled 
5.4.1. If this is supposed to be the most stable release, recommended 
for those that want reliability over features, it should be labeled and 
distributed as such.
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Re: [slim] SB2 really doesn't support 96kHz PCM on S/PDIF?

2005-03-16 Thread Phil Karn
Patrick Dixon wrote:
Phil, are you seriously saying that you can design a 'brick-wall'
reconstruction filter at the Nyquist rate (fs/2)?
With reasonably modern DSP techniques such as those employed in most 
modern DACs, yes. It's not *totally* brick-wall because all such filters 
are impossible, but it is a very good approximation.

All it takes to provide an excellent LPF is a FIR filter with enough 
taps, and this is again readily doable with modern DSP hardware and 
oversampling so that the final analog filter can be trivial.

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Re: [slim] Ogg Vorbis Support on SB2?

2005-03-16 Thread Phil Karn
Christian Pernegger wrote:
Sorry if I missed something in this thread, but what's the down side
of ogg-pcm on the server?  Bandwidth to the SB?

Yes. Raw PCM is just below 1400kbit while the ogg stream itself would be 
much larger.
Don't you mean that the ogg stream would be much *smaller* than raw PCM?
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Re: [slim] Open firmware for SB2?

2005-03-13 Thread Phil Karn
Patrick Dixon wrote:
IMHO, the two biggest threats to Slim Devices' competitive advantage are:
* Product design - most 'normal' people think the Roku styling is better.
Maybe. Personally, I think basic functionality and reliability are far 
more important. Then again, my Squeezeboxes are all black.

* Simple software installation - most 'normal' people can't (or can't be
bothered) to spend hours reconfiguring their computer to get an application
running - if it doesn't work reliably straight from the tin, they'll just
send it back and move on.
Absolutely!
The second produces a major dilemma - the opensource community is
notoriously geeky and seems to just love wading though reams of poorly
documented or undoccumented source code to re-configure it for some strange
combination of a Linux installation.  But if the company concentrates on
supporting and making the software work seamlessly with Windows and iUnix,
it will probably alienate the geeks.
I don't think that's really a big dilemma. These sorts of sponsored 
open source projects work best when the volunteers work on the features 
that personally interest them, and the commercial sponsor acts as the 
project glue -- merging patches, conducting regression testing, and 
managing the release cycle. I can't see how any geek could oppose the 
mere existence of a stable Windows version (though that's arguably a 
contradiction in terms) so long as the code he's interested in remains 
open and hackable.

What I *do* find discouraging is the distressing unreliability of even 
the 5.4 version of the server software. I shouldn't have to install the 
version du jour just to get a fix for a bug that keeps crashing my 
server in routine usage.

There ought to be two code bases: a relatively stable, no-frills version 
with an emphasis on robustness, and an experimental version with all the 
latest gimmicks. As new features prove themselves and become stable, 
they can be backported to the stable version. This is hardly a novel 
concept; it's worked very well for Linux.

Most of the volunteers would probably prefer to play with the 
experimental release, while the people at Slim Devices would maintain 
the stable version. After all, their product is pretty much useless 
without a server to drive it.

--Phil
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Re: [slim] Re: problems with symbolic linking

2005-03-13 Thread Phil Karn
Dan Sully wrote:
I agree completely Phil - we're focused right now on making 6.0 stable. 
There
was a big jump in functionality, speed and stability in changing out the
entire backend between 5.4 and 6.0. However, as you well know, bugs are
introduced along the way.
I think this is exactly where the problem is. I shouldn't be forced to 
choose between an old version with lots of bugs, and a new, 
bleeding-edge version that fixes (most of) the older bugs, but also 
introduces many new bugs associated with lots of new features that are 
largely useless to me.

There really should be two, separately maintained code bases: a stable 
version where stability and robustness are paramount, and an 
experimental version with the latest new features that may not be 
particularly stable.

This *doesn't* mean that you never update the stable version. You 
update it as often as you have to to fix a bug or security hole. Even 
daily, if you have to. But new features, even seemingly simple ones, are 
*prohibited*. Those go only into the experimental tree.

I think the Debian Linux project does it exactly right. Debian has three 
flavors: stable, which is regularly updated *only* with security 
fixes; unstable, with all the latest, bleeding edge features; and 
testing, which is something like stable, but periodically updated with 
features that have proven themselves for a while in the unstable 
version. Once in a while, you rev the stable version, including only 
those new features that have already withstood the test of time. It 
works pretty well for me.

Phil
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Re: [slim] Apple Lossless vs. FLAC

2005-03-13 Thread Phil Karn
Todd Larason wrote:
Can you point me to documentation for the Vorbis metadata?  I've scanned
vorbis.com and xiph.org, and I'm just not finding it.
Start here:
http://www.xiph.org/ogg/vorbis/doc/v-comment.html
I decided on Vorbis for my meta data partly because I have a large 
collection of classical CDs, and whoever defined the MP3 ID tags 
obviously just wasn't thinking about classical music. The Vorbis comment 
 conventions aren't complete, but they're a big improvement over MP3 
tags, and more importantly they're free-format and extensible. E.g., 
when ripping piano and violin concertos I easily added SOLOIST tags 
even though no software currently pays attention to them. I figure 
they're not that important right now, and support for it can always be 
added later.

The other convention I strongly recommend when tagging classical music 
is to make sure each TITLE tag is self contained. If a work spans 
several tracks, I'll include the name of the work in each TITLE comment. 
For example, if a CD contains Beethoven's 9th symphony in 4 tracks, then 
I'll use TITLE tags that look like this:

TITLE=Symphony 9 - I. Allegro ma non troppo un poco maestoso
TITLE=Symphony 9 - II.Scherzo, Molto vivace
TITLE=Symphony 9 - III. Adagio molto e cantabile
TITLE=Symphony 9 - IV. Presto, Allegro assai
This is important because many classical CDs contain several unrelated 
symphonies or concertos, sometimes by multiple composers. You can't use 
the ALBUM comment for this purpose, because it really ought to be the 
same for every track on the same CD, even if it contains more than one 
work. iTunes has a GROUP tag that seems to be designed for this exact 
purpose, but it's not in the Vorbis comment conventions, no software 
supports it, and it's important enough that I wanted something that 
would be displayed by existing software like the Slimserver.

Because there's already a Vorbis COMPOSER comment, there's no need to 
include the composer's name in the title tags. I use it to contain the 
composer's complete name, e.g., Ludwig van Beethoven. The ARTIST tag 
contains just the composer's last name, which I also use to name the 
actual directories. (I make exceptions for names like Bach and Strauss, 
where there's more than one composer with the same surname.)

The name of the orchestra goes into the PERFORMER tag, the conductor 
into the CONDUCTOR tag, and so on. The existing Slimserver software 
seems to deal reasonably well with all this except for my nonstandard 
SOLOIST tags, which aren't all that urgent anyway.

The iTunes practice isn't that simple.  For mp3 files, for most things, the
definitive metadata is stored in ID3 tags in the file; the information is
cached in a binary file and reflected in an exported XML file.  The definitive
volume adjustment, equalizer, star rating, and start and stop time information
is stored in the binrary file and relfected in the exported XML file.  The
part of a compilation field is confusing: as nearly as I can make out, it is
initialized from the nonstandard TCMP ID3 tag, and when changed through iTunes
the TCMP tag is updated; unlike all the other ID3 tags, though, changes made
outside iTunes post-import are never reflected inside iTunes.
Thanks for this description! I've looked at the XML files on occasion, 
and I've wondered which files iTunes reads and which it writes. I have 
noticed that I can delete the binary database file and force iTunes to 
rebuild it from the XML file. You can also blow away the database 
entirely and re-import it from an imported XML file. I've done this a 
few times after making manual edits to the XML file for various reasons.

--Phil
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Re: [slim] Apple Lossless vs. FLAC

2005-03-13 Thread Phil Karn
Todd Larason wrote:
The iTunes practice isn't that simple.
Speaking of the iTunes database structure, do you happen to know of any 
utilities that can scan a music folder and build the iTunes XML 
structure from the tags in the music files (other than iTunes itself, 
that is)?

The reason I ask is that I have an Ogg Vorbis plug-in for iTunes that 
plays just fine, but it doesn't read the Vorbis tags into the database 
when I add an Ogg Vorbis file to the library. I'm also thinking of 
writing a NFS shim that would make my FLAC archive appear to iTunes as 
if it was full of WAV files, and WAV files don't have any metadata at 
all for iTunes to import.

Phil
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Re: [slim] Apple Lossless vs. FLAC

2005-03-13 Thread Phil Karn
Michael Peters wrote:
There are issues in other areas - for example, some CD's will have
more than one artist - U2 for example, the album Rattle and Hum.
In that case I'd just use a different ARTIST tag for each track, just as 
for a compilation album like Greatest Hits of the '70s. Unlike the 
ALBUM tag, which really ought to be the same for every track on the same 
physical CD, there's no reason that the ARTIST tags all have to be the 
same. If a particular track is a joint effort of several artists, then 
just list them all in the ARTIST tag for that track. Both iTunes and the 
Slimdevices databases do a pretty good job of indexing all this stuff so 
you can find what you want, and that's what really matters. That leaves 
just the question of how to name the directory that contains the album 
directory; iTunes' use of Compilations is as good as any.

The Vorbis COMPOSER and PERFORMER tags give you even more indexing 
flexibility, if you need it. I don't usually bother to set the COMPOSER 
tags on popular music, but some songs are *so* widely covered that it 
makes sense to do so. E.g., I must have a dozen different versions of 
Dave Mason's song Feelin' Alright by a half dozen different 
performers: Dave Mason, Grand Funk Railroad, Joe Cocker, Three Dog 
Night, Traffic, etc.

Phil
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Re: [slim] Open firmware for SB2?

2005-03-13 Thread Phil Karn
Patrick Dixon wrote:
it's worked very well for Linux.
Really?  As someone struggling to get FC3 configured, googling for
information produces many more people with Linux problems than there are
solutions out there.
I was talking about Linux, which is just an OS kernel. There has 
traditionally been a production branch and a development branch, with 
features from the development branch integrated back into the production 
branch after they've been adequately tested. As a result, the production 
branch has been remarkably stable. All of my Linux boxes routinely run 
for months at a time without crashing; when they do, it's almost always 
because of a hardware failure like a crashed disk or a power failure.

BTW, I use Debian, not Red Hat.
--Phil
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Re: [slim] Re: shorten

2005-03-13 Thread Phil Karn
kdf wrote:
to make things worse, systems like windows dont like /dev/null so the server
spits out a warning for every shorten file it finds and never adds them to the
db properly.  I have only about 10 shorten files.  Only 2 are playable with
slimserver. I have tried to find alternative ways of getting the info, but I
give up after losing too many evenings.  tips would be useful.
Well, transcoding to FLAC as Michael Peters does would seem to be the 
right answer. I've already decided to do that with the shorten files I 
get from etree trading.

The only problem with this is if I want to continue to help make these 
files available via Bit Torrent, then I have to keep both formats 
around, and that's extra disk space.

Phil
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Re: [slim] Slim Devices SB2 disappointment SB for sale.

2005-03-13 Thread Phil Karn
momerath wrote:
having odd problems connecting after power off, and, of course, pcm
streaming skips sometimes, even in situations I would have thought
would certainly work. 
I think the PCM skipping (buffer underflow) problem could be 
substantially improved if a little attention were paid to thread 
priority and CPU scheduling in the server. That's something Slim Devices 
could do to help keep the faith of those of us with original Squeezeboxes.

Phil
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Re: [slim] Suggestions for 802.11g Wireless Router?

2005-03-13 Thread Phil Karn
Chris Glushko wrote:
I would suggest staying away from Netgear.  I just
replaced my WR614 today.  Even with the latest
firmware, my netgear router was never very stable and
under intense use (i.e. a few hours of Bittorrent w/
Azureus) the netgear would essentially freeze, needed
to power cut off manually to unfreeze it.  
I agree. Netgear's plain vanilla Ethernet hubs and switches have always 
worked great for me (I also like their simple metal cases) but their 
WR614 wireless access point was always a problem. It had poor coverage, 
and even at close range we had unacceptably high packet loss rates to 
our Power books. Firmware upgrades and appeals to customer support 
didn't help. So it's been sitting unused in a box ever since we got our 
first Linksys WRT54G.

I really wish Consumer Reports could evaluate stuff like this. They 
could save a lot of people a lot of time and money.

Phil
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Re: [slim] Apple Lossless vs. FLAC

2005-03-13 Thread Phil Karn
Phillip Kerman wrote:
I have a proof of concept thing I built in Flash that parses iTunes's XML.
What do you need to extract exactly?  I'm pretty sure it'd be easy to adapt
this thing I have to output a string (in any form you want) to your
clipboard so that you can paste it into another tool or text file.  Or, do
you just want to view the data with something other than iTunes?
No, I want to build the iTunes database by extracting FLAC and Ogg 
Vorbis tags (they're the same format). iTunes itself does this 
automatically when importing MP3 and AAC files, but when I import Ogg 
Vorbis songs with the iTunes Ogg Vorbis plugin installed, the Vorbis 
comments are ignored. I have to enter them by hand.

There's a Sourceforge project that's supposed to be working on a FLAC 
plug-in for Quicktime/iTunes, but it doesn't seem to work. So while I'm 
waiting, I thought I'd build a simple shim for NFS that would make my 
library of FLAC files on Linux look like a network filesystem full of 
WAV files to my desktop Mac. I can then import them into iTunes and play 
them from the server onto my Mac. However, there wouldn't and couldn't 
be any meta information in iTunes' database, because WAV files don't 
have meta tags. That means I'd have to add the meta info manually to the 
iTunes database, or preferably use a tool to extract the tags from the 
FLAC files and build an XML file that I could then import into iTunes.

I actually like iTunes. Although it doesn't have native support for my 
preferred formats, it has one of the best user interfaces of any music 
jukebox program around. And it's pretty stable. So when I want to listen 
to music while I'm at my computer, I'd much rather use iTunes than 
SoftSqueeze. I'd prefer to limit my use of SlimServer to just my 
Squeezeboxes, at least until it becomes a lot more stable.

Phil
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Re: [slim] Suggestions for 802.11g Wireless Router?

2005-03-13 Thread Phil Karn
Michael Peters wrote:
Just for the record - I have the same issue with my Linksys wrt54g router.
I should point out that I use only the wireless base station and 4-port 
Ethernet switch features on our WRT54Gs. I don't use the built-in DHCP 
or NAT/routing functions at all; that's all handled by a Soekris 
Engineering box running Linux. So I wouldn't have encountered any bugs 
in the WRT54G's routing or DHCP functions.

Phil
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[slim] Virgin Radio Ogg Vorbis streaming problems

2005-03-12 Thread Phil Karn
Has anyone tried to listen to Virgin Radio (London) on their Ogg Vorbis 
streams?

The streams are:
http://ogg.smgradio.com/vr160.ogg
http://ogg.smgradio.com/vc160.ogg
http://ogg.smgradio.com/gr160.ogg
Their equivalent MP3 streams work fine, but their Ogg streams close at 
the end of each song. At first I thought they were kicking people off 
intentionally, but a little packet tracing showed that slimserver, not 
Virgin's server, is closing the TCP connection.

I also see this message in slimserver.log each time it happens:
Only one logical bit stream currently supported
This message, plus the fact that the problem always occurs during the 
segue between songs, suggested to me that Virgin sends separate streams 
for each song, they overlap during the transition, and this overlap 
apparently isn't supported by oggdec. Does anyone know of an Ogg Vorbis 
stream decoder that does work in this application?

I wonder if *anyone* is listening to Virgin's Ogg streams. It's a shame, 
since the audio quality is noticeably better than their 128k MP3 stream.

I'll probably dig into this problem myself, but I wanted to see if 
anyone else has solved it first.

Phil
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[slim] Bombout in 6.0b1

2005-03-12 Thread Phil Karn
I brought up 6.0b1 tonight, and while it was rebuilding the database it 
abruptly exited with these messages:

Use of uninitialized value in multiplication (*) at 
/usr/local/slimserver/Slim/Player/SqueezeboxG.pm line 65.
Can't use an undefined value as an ARRAY reference at 
/usr/local/slimserver/Slim/Player/Squeezebox2.pm line 209.

Is the server supposed to just exit when it encounters an error like 
this instead of trying to continue?

--Phil
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Re: [slim] problems with symbolic linking

2005-03-12 Thread Phil Karn
kdf wrote:
that's a font crash.  are you trying to run slimserver 5.4 with a softsqueeze
session that is still running from 6.0?
SB2 (and softsqueeze 2.0a11) use different fonts, so they will cause earlier
servers to barf.
It may also be that you ran with 6.0 and it had the same MAC addy, so the player
prefs have been updated with SB2 stuff.  You may have some better luck if you
play with the softsqueeze settings to have different mac addresses.
I just figured that out for myself a few seconds ago. When I killed the 
copy of softsqueeze I had been running, slimserver 5.4 came back up.

You know, I hate to say this but it's really been bothering me for some 
time and I think I need to say it. The Squeezebox is a *really* nice 
piece of hardware. When it works, it's wonderful. The design philosophy 
of having the box do just the essential functions and do them well while 
moving all the smarts to a much more capable and open server is exactly 
the right approach.

But this also means that without its server, the Squeezebox is a totally 
useless lump of plastic. So that server has *got* to be reliable.

Since I got my first Squeezebox, I've been more than a little surprised 
at the overall brittleness of the server software, at least on Linux (I 
haven't tried any of the other OS versions). It seems to abruptly exit 
at the slightest provocation, such as this little mixup on fonts. 
Various cryptic Perl error messages appear from time to time in the log; 
whether they're cause for concern, I haven't a clue. Seemingly simple 
web operations, like walking down a directory tree, often take undue 
amounts of time. Sometimes I'll see it gobbling all available CPU time 
for no apparent reason; aborting and restarting seems to make it behave 
again for a while. Clicking rescan after adding music rarely seems to 
work right; I almost always have to blow away the cache and restart. And 
so on.

That's with the 5.4 version. The 6.0 versions have been even worse; I 
haven't gotten one to play yet without crashing within the first minute 
or two with another flurry of cryptic Perl error messages.

This is not quite what I expected from a consumer product seeking a mass 
market. It is just not reasonable to expect every user of a device like 
this to be a skilled Perl hacker, able to know that a cryptic message like

Can't use an undefined value as an ARRAY reference at 
/usr/local/slimserver_5.4/Slim/Player/SqueezeboxG.pm line 123.

followed by an abrupt server crash means that you're running an 
incompatible version of the client somewhere on your network. Not only 
should the error message be a little more informative, but the user just 
might be forgiven for foolishly expecting that software written as a 
multi-user network server should be just a little more robust against 
unexpected input in general.

I don't know why it should be so difficult to make a simple and robust 
server for this thing. Perhaps the other users consider a rich selection 
of 17 web skins and the various plugins and news ticker crawls more 
important than basic functionality and robustness; I don't know. But I 
do know that, as nice as the Squeezebox is, I just can't recommend it 
yet to my less technical friends until it comes with a *far* more robust 
and foolproof server. When it does, I'll recommend it wholeheartedly.

--Phil
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Re: [slim] 60b1 buggy?!

2005-03-12 Thread Phil Karn
[EMAIL PROTECTED] wrote:
Tried the new version 60b1 yesterday, but immediately downgraded to 5.4.1
Where's 5.4.1? I don't see it in http://www.slimdevices.com/downloads. I 
only see 5.4.0 and the v6 betas.

If there's a stable, no-frills version of the server software, I would 
very much like to use it.

Phil
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Re: [slim] SB2 and FLAC

2005-03-11 Thread Phil Karn
Phillip Kerman wrote:
For my ears, my SB plays FLACs back to back perfectly.  I've yet to find an
instance where it doesn't... and there are many cases where you'd definitely
hear a pause if one were inserted.
Right, thanks for confirming my observation.
As an aside, I have noticed that, unlike flac, shorten (.shn) files play 
through the Squeezebox with a tick between tracks. This seems to be 
because the shorten command used for decompression has no option to 
produce a raw, headerless stream. The tick sound is the sound of the 
reconstructed .wav header being fed to the DAC.

Shorten still seems to be popular on bt.etree.org and other sites where 
audience recordings of taper-friendly bands are traded. I can't figure 
out why, as flac seems to be superior to shorten in every way. FLAC 
files are about 5% smaller, and they support metatags.

--Phil
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Re: [slim] Open firmware for SB2?

2005-03-11 Thread Phil Karn
Sean Adams wrote:
That was SLIMP3 firmware, which was all code I wrote myself, so we could 
release it however we wanted. Squeezebox and Squeezebox2 include 3rd 
party proprietary OS and wireless drivers and require a $30,000 
development kit in order to write code for them. These are just a few of 
the obstacles to opening up firmware development.
Well, I suppose an alternative would be to open up all the necessary 
hardware specs (if they're not already open) and encourage an 
independent open source firmware development from scratch. If or when it 
surpasses the stock firmware in features and stability, you could switch 
to it.

Phil
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Re: [slim] MP3 quality on repeat copies

2005-03-11 Thread Phil Karn
Patrick Dowling wrote:
On average what level of compression are you seeing?  I know there are 
many factors, but for space planning I'm looking for a general estimate.
A very rough rule-of-thumb is about 50% (2:1). You get better than that 
on quiet classical music, and worse on loud, volume-compressed rock.

Phil
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Re: [slim] MP3 quality on repeat copies

2005-03-11 Thread Phil Karn
Here's another data point for comparison. I recently bought the 
Magnatune label album Nocturne by The West Exit and was given my 
choice of file format to download. Here are the sizes:

# 44k/16bit WAV: 507meg zip of perfect quality WAVs.
# FLAC: 357meg zip file of perfect quality FLAC files.
# OGG: 65meg zip file of high quality Ogg Vorbis files.
# 128kb MP3: 47meg zip file of 128kb MP3 files.
# MP3 VBR: 84meg zip of high quality MP3 VBR files.
# Mac iTunes: 58meg iTunes OSX AAC files.
# Mac MP3: 96meg OSX  OS9 compatible 256k MP3s.
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Re: [slim] Suddenly can't start SlimServer on Linux

2005-03-11 Thread Phil Karn
Jack Coates wrote:
Either slimserver is already running, or something else is using its 
network port. As root, ps aux | grep slimserver.
That won't show if another process is listening to that port.
If you have the lsof (list open files) command, that's a better way to 
get the information you need. E.g., if I do

sudo lsof | grep TCP
on the Linux machine that runs my slimserver, I'll get (heavily edited):
slimserve 32635 slimserver8u  IPv45470505  TCP 
*:3483 (LISTEN)
slimserve 32635 slimserver9u  IPv45470506  TCP 
*:9000 (LISTEN)
slimserve 32635 slimserver   10u  IPv45470507  TCP 
*:9090 (LISTEN)
slimserve 32635 slimserver   11u  IPv45709439  TCP 
homer.local.ka9q.net:3483-192.168.1.31:35348 (ESTABLISHED)
slimserve 32635 slimserver   12u  IPv45709440  TCP 
homer.local.ka9q.net:3483-otto.local.ka9q.net:33030 (ESTABLISHED)
slimserve 32635 slimserver   13u  IPv45520472  TCP 
homer.local.ka9q.net:9000-milhouse.wi.local.ka9q.net:51407 (ESTABLISHED)
slimserve 32635 slimserver   32u  IPv45718345  TCP 
homer.local.ka9q.net:9000-otto.local.ka9q.net:33036 (ESTABLISHED)

This shows that process ID 32635, named slimserve(r), is listening on 
TCP ports 3483, 9000 and 9090. Chances are you'll find some old 
slimserver process still bound to one of the needed ports, and that's 
keeping you from starting a new one. Kill the old ones and try again.

--Phil
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Re: [slim] Apple Lossless vs. FLAC

2005-03-11 Thread Phil Karn
Chris Glushko wrote:
If you are an iPod user and you like using iTunes to
manage your music, wouldn't Apple Lossless be the most
logical choice for your primary archive?
Well, it might be -- if you're willing to spend all that precious iPod 
disk space on a lossless format. If you insist on lossless even on your 
iPod, then Apple Lossless is your only choice.

I understand the preference to Open Source, but aren't
you just adding extra work for yourself for nothing
more than to spite apple if you are an iTunes/iPod
user?
Well, for one thing I didn't buy an iPod. Even though I have several 
Macs and do use iTunes, I bought an iRiver 340 specifically to get Ogg 
support. If Apple were to support Ogg Vorbis on the iPod, I'd buy one in 
a heartbeat.

That said, the numbers I've seen tend to indicate that FLAC achieves 
somewhat better compression ratios than Apple Lossless. Also, FLAC 
supports Vorbis-style metatags, which I consider vastly superior to 
either MP3 ID tags or the iTunes practice of keeping all the meta 
information in a separate massive XML file. I invest a lot of time 
getting the metainfo right, especially on classical music, so the right 
tag format matters.

Phil
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Re: [slim] Out of curiosity

2005-03-11 Thread Phil Karn
Joshua Uziel wrote:
Oh, I'm well-aware of SCO and a whole bunch of other open source related
cases over the years. :)  Sure, this is a conceivable outcome.  To be
honest, while I'd prefer to be able to play with the firmware for the
SB2 and add stuff like native ogg support, the fact that it requires a
totally expensive toolchain suite ($25k was it?) kinda shoots it in the
head.  I'm happy enough with the source for the server anyways.
This is about as good an argument as you can make for putting as much as 
you can in the server and keeping the client (the Squeezebox) as simple 
and generic as possible.

The Squeezebox should focus on a very basic set of hardware functions 
and doing them as well as possible. Everything else -- all the bells and 
whistles -- belong on the server where you have essentially unlimited 
CPU and memory and a powerful, open development environment to do 
whatever your heart desires.

I presume Slim Devices chose its name because they felt much the same way.
IMHO, I can think of only two really valid reasons to justify additional 
functionality in a device like the Squeezebox.

The first would be features that make it easier for the server to meet 
its real-time constraints to prevent buffer exhaustion. The SB1 has a 
rather small buffer that empties in less than one second at PCM rates, 
and this can be difficult to meet on a general purpose server that's 
also doing other things. The SB2 has essentially solved this problem 
with a much bigger buffer. If you can't get your server to refill the 
SB2's buffer often enough, you've got bigger problems to solve.

The other class of enhancements would alleviate the Squeezebox's 
bandwidth requirements on wireless channels. (PCM is already a trivial 
load on a 100 Mb/s wired connection. If you still have 10 Mb/s hubs, 
upgrade!) The built-in MP3 decoder in the original Squeezebox certainly 
helps with this, as does the addition of 802.11g and built-in FLAC 
decoding in the SB2 for the purists who insist on a totally lossless 
audio path under all conditions.

With these new features to the SB2, I really can't think of anything 
else that I *really* can't live without. I think Slim Devices has done 
an excellent job with both the SB and SB2, and wouldn't want to see it 
sunk with creature feep as has happened to so many other products of its 
type.

Phil

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Re: [slim] Gapless playback with mp3 - another shot

2005-03-09 Thread Phil Karn
Can somebody give me some background on gapless playback? I don't 
understand why it's a big problem.

With FLAC, at least, I don't seem to have any problem at all playing 
tracks that flow together seamlessly. Occasionally I'll get a FLAC file 
that was created with an erroneously embedded WAV header that results in 
a tick at the beginning of each new track, but there is no actual 
interruption.

Phil
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Re: [slim] SB2 and FLAC

2005-03-09 Thread Phil Karn
Steinar Bjaerum wrote:
I have started to rip my record collection to FLAC. I am using one FLAC file
per album, internal cuesheets with tag information included as numbered
Vorbis Comments.
This works great with SB1 and server-side decoding.
I'm curious as to why you did this. I've also ripped my own collection 
to FLAC, but with one track per file.

Before I did this, I ran a test using everybody's favorite continuous 
album, Pink Floyd's Dark Side of the Moon. I ripped it as a single WAV 
file, and then again as individual tracks. I found that the 
concatenation of the individual track files was exactly bit-for-bit 
identical to the single WAV file. This told me that I could rip 
everything as single tracks and have them play correctly as long as the 
player doesn't insert any pauses. This seems to be the case for the 
Squeezebox/Slimserver.

What support is there in Slimserver/Squeezebox for cue points within a 
single large FLAC file? How do you easily play just one track on an album?

Phil
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Re: [slim] Open firmware for SB2?

2005-03-09 Thread Phil Karn
Jason wrote:
The obvious risk to slim devices if they do this is that some other company
can produce a super budget version of the SB, load the SB firmware onto it
and completely undercut Slim Devices potentially putting them out of
business. 
Quite frankly, I just don't see all that much value in the existing SB 
firmware that couldn't be easily duplicated from scratch by someone else 
building a similar product. After all, it *is* a slim device, meaning 
that the box does very little on its own; nearly all the heavy lifting 
is done back at the server, and that's already open source.

So open-sourcing the Squeezebox firmware would result in Slimdevices 
picking up a lot of new features from its users for free, and since they 
already make a very good hardware box at a reasonable price I don't 
think they have much to worry about from being knocked off.

Phil
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Re: [slim] Squeezebox 2, could it be an access point also?

2005-03-09 Thread Phil Karn
Andrew Lucas wrote:
Hi,
Just curious, seeing you can get to the Squeeze 2's wireless interface via
the ethernet connector, would it be also possible to use this this a basic
wireless access point.
The SB2 as described is a wireless bridge, not an access point. A 
wireless bridge is a *client* of an access point, and that's a very 
different (and much simpler) set of functions than being an access point.

Besides, with excellent access points like the Linksys WRT54G now down 
to the $60-70 level, there's simply very little point in adding access 
point capability to the SB2. I'd rather keep the box cheap, simple, and 
focused on doing one thing (playing music) very well.

Phil
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Re: [slim] Squeezebox2 wireless interference question

2005-03-09 Thread Phil Karn
Christian Pernegger wrote:
Question: Are there any disadvantages in having the radio and one 
antenna internal? I'm thinking interference,  added noise in the analog 
stages ...
Probably very little, if any. Spurious noise from computing equipment is 
generally strongest at the lowest frequencies, and it falls off fairly 
rapidly in intensity the higher you go. That's why you tend to have so 
much more trouble with AM radio interference (~1 MHz) than FM (~100 MHz).

802.11 operates on the 2400 MHz band, which is high enough for most 
computing equipment to generate relatively little noise. Even high-end 
CPUs like Pentium 4s and Athlons with clock speeds in the multi-GHz 
region confine signals at those frequencies to inside the chip; the 
external interfaces to memory and peripherals that can radiate 
interference run at slower speeds, e.g., 800 MHz or 1 GHz. Harmonics can 
certainly exist at higher frequencies, but they probably won't be all 
that strong.

The biggest source of radio frequency noise from Squeezeboxes has been 
the little power supplies that they bundle with them. Slimdevices ought 
to return them for a refund and use a cleaner model. Quite a few people 
are now complaining that they can't listen to an AM radio anywhere in a 
house that has a Squeezebox.

Phil
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Re: [slim] (a bit OT) firewall in the router (was Squeezebox2)

2005-03-09 Thread Phil Karn
Ken Hokugo wrote:
Dean or anyone,
Would the firewall feature in these routers (wireless or wired) be good 
enough so that I can get rid of Zonealarm Pro which contributes 10 to 
15% more of CPU usage when playing Slimserver?  If I could get rid of 
the sw based firewall, that would be great.
A particularly powerful and flexible firewall is a Linux box with 
multiple Ethernet interfaces. If you'd rather not dedicate a full-blown 
PC to the job, Soekris Engineering (www.soekris.com) makes a line of 
single-board PC-compatible machines specifically designed as network 
engines. They come without any software, so you have to roll your own, 
but there are many people who can help you.

I have a Soekris net4801 acting as my primary router. It provides QoS 
(Quality of Service) in the upstream direction to my DSL line, along 
with DHCP, IPv6 routing/tunneling and IPv4 NAT for any local machines 
that need it.

Except for the filtering inherent in a NAT, it doesn't actually filter 
any packets because I basically don't believe in firewalls; I'd much 
rather just keep my individual machines as secure as possible. 
Basically, that means banning anything and everything from Microsoft; 
we're in the process of getting rid of the very last Windows machine on 
our network (my wife's desktop) and replacing it with an iMac. The 
combination of Mac OS X on the desktop and Linux on servers can do 
pretty much everything Windows can do, and do it a whole lot better and 
with far better security.

Phil
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Re: [slim] Squeezebox2

2005-03-09 Thread Phil Karn
Christian Pernegger wrote:
I'd been holding out for a public domain linux apple
lossless decompression solution that would run
compfortably on a squeezebox, but this might push me
to dual libraries (one flac, one aac for ipod)...

There is a preliminary FOSS decoder for ALE out:
http://craz.net/programs/itunes/alac.html
You could have SlimServer transcode ALE -- FLAC
without loss of quality.
That said, I'd much rather people didn't use closed
formats.
Agreed!
I don't see much reason to ever use ALE, though it's certainly nice to 
now have the ability to decode it if necessary. Lossless formats are 
still too bulky for portable players, so even though an iPod supports 
it, it doesn't seem very practical.

That leaves AAC and MP3, since the iPod supports them but not Ogg 
Vorbis. Since AAC seems to give better quality than MP3 for the same 
data rate, then it makes the most sense to keep your primary archive in 
FLAC and convert to AAC for the iPod as necessary. Both AAC and MP3 are 
patent encumbered, so you're forced to choose between two evils.

But if you're lucky enough to have some other portable player that 
supports Ogg Vorbis, then you can produce Ogg Vorbis versions from your 
FLAC masters and not deal with proprietary formats at all. That's the 
route I've chosen by buying an iRiver 340.

--Phil
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Re: [slim] Squeezebox install creates IP address conflict -- resolved!

2005-03-06 Thread Phil Karn
Roy Owen wrote:
Don't be suprised if the problem re-occurs.  It's been my experience
that problems that fix themselves will bite you in the backside at the
most inconvient time.
Yeah. That's why you should take the time to assign permanent addresses 
to each one of your devices, and use dynamic addressing only for guests.
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Re: [slim] Squeezebox causing RFI on AM

2005-03-06 Thread Phil Karn
Maurice Poirier wrote:
Since I installed my Squeezebox, there is constant static interference 
on AM radio reception. Is it the ethernet cable that is causing this -- 
or the Squeezebox itself??
 
The AM tuner is on same receiver that the Squeezebox is connected to, 
and the Squeezebox is located on a shelf ca. 16 inches below the 
receiver. (The Squeezebox itself is working fine). I temporarily 
disconnected the ethernet cable (cat 5E) from the Squeezebox, but the 
interference remains. Any advice on how to resolve this problem would be 
much appreciated.
It's almost certainly the switching power supply that came with the 
Squeezebox. My boss (who initially turned me onto the Squeezebox) had 
exactly the same problem, and he eventually found a different model of 
power supply that greatly improved the situation.

You might also try adding an inline RFI filter to the standard power supply.
I don't know how some of these power supplies ever get FCC certification.
Phil
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Re: [slim] MP3 quality on repeat copies

2005-03-06 Thread Phil Karn
Others have already pointed out that MP3 files are like any other kind 
of computer file; as long as no errors occur during copying, the copies 
will be exactly identical to the original. Only if you decompress and 
recompress will there be any further degradation.

Having said that, let me make a plug for FLAC (Free Lossless Audio 
Compression). Not only does FLAC not lose *any* quality from the 
original PCM WAV format (the format on standard audio CDs) but the FLAC 
header also includes a built-in MD5 hash of the uncompressed audio. This 
lets you verify a FLAC file to ensure that it's not corrupted. (You use 
the command flac --test file.flac). I don't know of any other commonly 
used audio format with this *very* nice feature.

FLAC uses much more space than the lossy formats like MP3, Ogg Vorbis 
and AAC, but modern disks are *so* roomy and cheap that this just isn't 
a big deal anymore for a desktop machine.

I chose FLAC as my primary format on my Slimserver, and I can still 
produce lossy compressed versions from the FLAC masters as needed for 
various portable players.

Phil
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[slim] new open source decoder for Apple Lossless

2005-03-05 Thread Phil Karn
http://craz.net/programs/itunes/alac.html
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Re: [slim] Modifying squeezebox clock

2005-03-04 Thread Phil Karn
momerath wrote:
Phil is absolutely correct that the burden of proof is on the
audiophile, and that a double blind study is the only practical way to
prove the audibility of jitter.  I intend to do this in the near
future with some skeptical friends. 
By all means, if you can run a controlled test, then do it!
I'm not sure how to best go about that. I know of some audio ABX testing 
software, but I think it's designed mainly to evaluate lossy codecs. 
You're testing something at a much lower layer. It might be possible to 
simulate the effect of jitter in software by using similar algorithms to 
those that convert between sampling rates, although here the rates would 
be very nearly identical. This might not satisfy everyone, though, as it 
wouldn't actually test the real hardware, so an actual hardware switch 
would probably be the best as well as the simplest.

The most crucial thing in any audio A/B test is level matching. 
Otherwise, the louder signal will almost always sound better. This 
shouldn't be hard to do in an all-digital system; just make sure 
identical bit streams come out both cables. Also, make sure that all 
switching transients and propagation delays are the same; if you do a 
test that involves A vs A and A vs B, you don't want the lack of a 
switching transient or change in delay in the first case to give it away.

Keep in mind that a positive result will be less conclusive than a 
negative result, at least with that particular subject. A positive 
result could be due to some artifact other than the property you're 
trying to test (e.g., different audio levels, different time delays, 
etc) and you've got to carefully exclude every possible artifact other 
than the one property (jitter) you're trying to test.

There's a classic example given in a MIT science textbook about just how 
incredibly hard it can be to design a proper experiment on human 
perception. It asked the question Can humans detect magnetic fields? 
The answers kept coming up Yes, but closer examinations kept finding 
various bugs in the experiment that explained the positive results 
(e.g., the test subjects could hear  the buzzing of the electromagnets, 
or saw the slight dimming of the lights as the switch was thrown, etc.) 
Eventually, after going to very great lengths to eliminate all 
artifacts, they got their answer: No, humans cannot detect magnetic 
fields, at least not at the intensities tested.

Let me know what happens!
--Phil
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Re: [slim] Proposal: more lists

2005-03-03 Thread Phil Karn
Robin Bowes wrote:
I'd like to propose that additional mailing lists are created to split 
up the traffic more, particularly on the general list. I'd also like 
to see the names of the lists change as indicated below.
In general, I think this is a bad idea. Topics have a way of crossing 
finely-drawn boundaries like these, and besides most of us now have good 
mail-reading software that can easily organize mail by topic and thread. 
This makes it easy to follow the threads and topics you want while 
ignoring the ones you don't want.

My observation is that when a list splits, one usually ends up bearing 
all the traffic while the others go unused. Even worse, people start 
cross-posting to all the lists. So nothing is really accomplished.

--Phil
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Re: [slim] New Squeezebox?

2005-03-02 Thread Phil Karn
Christopher Jacob wrote:
and a hard drive
and a keyboard
and a mouse
Then build yourself a mini-ATX PC, or buy a Mac mini and run Softsqueeze.
The whole point of a Squeezebox is to have a small, light, low-power, 
and **fanless** box that just plays music and plays it well. I think the 
current design is just fine, though it took me a while to get used to 
the complete lack of local controls on the box. I soon realized I'd 
probably never use them anyway. Heck I hardly even use the IR remote, as 
a web browser on a wireless laptop is usually much faster.

The key to a successful product like this is keeping it simple!
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Re: [slim] Magnatunes vs. slimserver

2005-03-02 Thread Phil Karn
Chip Hart wrote:

...after hearing a few references to Magnatunes here, I decided
to check it out.  Cool idea, surprisingly good music.  I was
about to work on parsing their WWW pages to extra playlists when
I realized _they'd already done the hard work_ and supply
xml/csv lists of all their music.  Wow.
Yup, Magnatune is run by one remarkably enlightened dude, John Buckman. 
As far as I can tell, it's a one-man operation. He participates in the 
Magnatune forums quite regularly, if you're interested.

Parsing these lists and creating groups of related playlists is
fairly easy (I just did it for a much grosser radio1234 file),
but I figured I'd see if anyone else has already tried this.
So - anyone played with Magnatunes and automatic playlist
generation?
I just grabbed his .m3u playlists, massaged them a bit with Perl and 
stuffed them into my playlist directory. I also grabbed his Shoutcast 
streams; he currently has 6 of them for different genres. I actually 
find myself listening mostly to these streams, and then going on his 
site when I hear something I like. His ultimate goal *is* to get you to 
buy an occasional album, to help cover his streaming costs.

One thing that seems to be missing from m3u and pls formats is
the ability to have comments or information about the tracks.
Sure, I can crank out 100s of playlists, even sort them by
genre, but that's not as handy is understanding that Phebe
Craig and Katherine Westine feature harpsichord duets and not
the Ukranian Children's Choir.  Try fitting that on the slim
display...I just wonder if anyone has come up with a smart(er)
way to do this.
I don't think so. I did notice that his playlist entries are URLs with 
file names that just give the track #, artist and song, no album, year 
or comment. But if you suck them into iTunes and quickly click on each 
one, iTunes will grab the MP3 tags from the stream and display them in 
the library window. I haven't run Slimserver on the machine with iTunes, 
so I don't know if Slimserver can pick this data out of the iTunes 
database and display it on the Squeezebox. But even if it doesn't, the 
Slimserver ought to extract and display the MP3 tags when you actually 
play the track.

Phil
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Re: [slim] Magnatunes vs. slimserver

2005-03-02 Thread Phil Karn
Chip Hart wrote:
Oh, unless I am misunderstanding you, check this out:
http://magnatune.com/info/api
Look at the XML and csv files - tons of data.  I've got it
parsed and all, what I really struggle with is a cool way to
I guess I wasn't aware of all that stuff. Cool! Now to see if I can do 
anything with it myself...

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Re: [slim] Modifying squeezebox clock

2005-03-02 Thread Phil Karn
Robin Bowes wrote:
What information on that image tells you that? It seems to me you're not 
understanding what you're seeing.
The legend says the time trace is 5 ns/div. A pair of measuring lines 
implies that the jitter on the uncorrected is somewhat less than that, 
about 3 ns.

As I recall (I can't find good, recent references), each subframe in 
S/P-DIF carries a (usually) 16-bit PCM sample in a 32-bit subframe, so 
the clock rate of the composite stereo signal would be 64 times the 44.1 
KHz CD sampling rate, or 2.8224 MHz. At this rate, a bit time is 354.31 
ns, so the 5 ns jitter is a tiny fraction of the bit time. That's why 
you only see one bit transition on the scope -- because its horizontal 
sweep rate has been blown way up to make the small jitter visible.

Even if that jitter were directly imposed on the local VCO, which it is 
not because of loop filtering, it would still be reduced by a factor of 
64 as the VCO clock is divided by 64 to produce the DAC sample clock. 
5ns of jitter would become 78 ps. Even tinier when you consider that a 
cycle of 20 KHz (the highest frequency the CD can reproduce) takes 50 
*microseconds* to complete. What fraction of an audio cycle is that? 
What's the FM modulation index? What is the resulting spectrum of 
sidebands around the 20 KHz signal? What about lower frequency audio? 
I'll leave the precise numbers to the reader, but it should already be 
obvious that they're already tiny for the 20 KHz signal, and even 
smaller at the lower frequencies.

Not only that, but it's showing us the raw signal jitter, before the 
bit clock has been reconstructed by the receiver's PLL and divided 
down to the sample rate, which decreases the jitter accordingly. And 
furthermore, there's no indication of the exposure time, so we don't 
know anything about the frequency spectrum of the jitter. That's 
important too.

It's the same for both cases. Level playing field, anybody?
Except that, in both cases, it's just too small to matter!
But you don't believe in all this audiophile mumbo jumbo about good 
cables and bad cables, do you?
Don't put words in my mouth. You don't need cables that can pass 2 MHz 
if you're carrying baseband analog. If you're carrying S/P-DIF, which 
has its spectral peak at 2.8 MHz, then you do.

It would be hard to find a coaxial cable that couldn't do an adequate 
job of passing the spectral components of a 2.8 MHz S/P-DIF signal over 
a few feet in a home stereo system. We regularly use even smaller coaxes 
to carry far higher frequencies from cell phones to external antennas.

I *can* see how there might be a problem with ordinary shielded, 
twisted-pair microphone cables such as the kind long used in 
professional work to run 600 ohm analog signals over significant 
distances. Here you'd probably want a redesigned cable better suited for 
megabit digital signals. Something like Cat-5, for example, which is 
really cheap and goes up to 100 MHz. It doesn't have to be expensive or 
gold plated to be good.

You're general POV seems to be that if *you* can't hear a difference 
then no-one else should either.
Not at all! For one thing, I'm 48 and my hearing is not what it was at 
18. But if I can't hear a difference, and some calculations cast strong 
doubt on *anyone* hearing the difference, then I think it reasonable to 
ask those who claim to hear a difference if they have conducted any 
proper blinded listening tests. If not, then I question their assertion. 
Audiophiles have a very long history of hearing all sorts of amazing 
differences that seem to disappear as soon as proper controls are 
introduced. That's a fact, and it would be foolish to ignore it here.

Have you ever heard the effect of phase modulation on an audio signal?
Sure I have. Remember I said I help design modems for a living. Phase 
modulation (e.g., PSK) is one of the modem designer's standard methods. 
I'm well aware of what large amounts of phase modulation sound like; 
I've spent many hours listening to these things while developing and 
using modems on satellite links. But that doesn't mean very tiny amounts 
of phase modulation sound at all alike.

The sort of improvements audio enthusiasts wax lyrical about can often 
be attributed to phase. Soundstage, depth, clarity, all that sort of 
stuff. Even a small phase error can radically smear the sound.
Yeah, but can you do it in a properly controlled test? How do you know 
you're not just fooling yourself?

I don't hear discrete sounds or tones, no, but I do hear relationships 
between sounds and positional information that is encoded in the HF 
band. If you lose that you lose clarity and detail in the sound.
Again, so you say. People can claim to hear anything. Prove it with a 
properly controlled test. That's all I ask.

And what happens if the crystal is inaccurate?
If it's in a wall clock, the clock runs a little fast or slow. If it's a 
local oscillator in a radio receiver, then its dial frequency 
calibration is a 

Re: [slim] Modifying squeezebox clock

2005-03-02 Thread Phil Karn
Phil Karn wrote:
Even if that jitter were directly imposed on the local VCO, which it is 
not because of loop filtering, it would still be reduced by a factor of 
64 as the VCO clock is divided by 64 to produce the DAC sample clock.
I'm going to have to revise and correct this. (In my defense, I've been 
home sick with the flu for the past few days, and I'm not firing on all 
cylinders.)

When the VCO clock is divided by 64 to obtain the 44.1 KHz sample clock, 
 the jitter time is *not* divided by 64.

With this correction, I believe the rest of my analysis remains valid. 
That is, the clock division by 64 reduces the modulation index of the 
jitter on audio components by that same factor. So 5ns of jitter (a 
little more than that scope showed) represents only 1/10,000 of a cycle 
of a 20 KHz sine wave (period 50 microsec).  And the modulation index 
would be proportionately lower at lower audio frequencies; down at 1 
KHz, where there's far more energy in a typical audio signal, 5 ns would 
be only 5/100 (5 millionths) of an audio cycle! These are truly 
*tiny* phase modulation indices that would produce very little in the 
way of PM/FM sidebands. I just can't imagine that anyone could hear them.

And I've left out the effects of PLL smoothing, which reduces jitter 
even further.

But if you feel otherwise, conduct some properly controlled listening 
tests. I'm all ears.

--Phil
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Re: [slim] Modifying squeezebox clock

2005-03-01 Thread Phil Karn
Triode wrote:
For some science, try the following:
http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C41%20SPDIF%20interface%20flawed.pdf 
Okay, I've read it. And I'm still not convinced.
Although his math looks fairly solid, he makes a lot of questionable 
assumptions that lead to very questionable conclusions. But he also 
makes some interesting observations.

The first observation is that significant (i.e., measurable, though not 
necessarily audible) jitter appears only when a composite S/P-DIF signal 
is severely bandpass filtered. Indeed, his entire paper is all about the 
jitter caused by such filtering. This surprised me, as I had initially 
assumed that people were complaining about the jitter from timebase 
oscillators.

That too-tight filtering can generate jitter is no surprise to me at 
all. I'm a digital communications engineer with experience in modem 
design, and we call this very well known and thoroughly understood 
phenomenon inter-symbol interference. (See the Nyquist Sampling 
Theorem for the math details.) Intersymbol interference is something we 
take great pains to avoid, as it can, if severe, impair the 
bit-error-rate performance of the modem even at high signal-to-noise ratios.

But here, the author concedes that the jitter isn't so bad as to cause 
any bit errors. The *only* problem that remains has to do with jitter in 
the recovered clock stream.

By now it should be obvious that if the only significant source of 
jitter is bandpass filtering of the S/P-DIF channel, then there is 
absolutely no point in replacing the timebase oscillator in a DAC in an 
attempt to reduce it. The timebase simply isn't the problem; the 
narrowband channel is the problem. Even the cheapest and worst crystal 
oscillators have very low phase noise; when you buy a more expensive 
crystal, you're mainly buying improved frequency accuracy and long term 
frequency stability, not lower phase noise. If the incoming S/P-DIF 
signal has a lot of jitter due to tight filtering, then even an 
absolutely noise-free local oscillator would be forced, by the error 
feedback signal in the clock recovery PLL, to reproduce this jitter at 
its output.

This reinforces the comment I made yesterday that if you're truly 
concerned about jitter, then the very last thing you want to do is to 
attach an external DAC to your Squeezebox. Just use its internal DAC and 
you'll get an analog signal with virtually no jitter because there's no 
S/P-DIF link and no PLL anywhere in the path. Just a DAC clocked 
directly by a crystal, playing out audio data at its own rate. You can't 
get better than that.

But back to S/P-DIF. The obvious and preferred solution to the S/P-DIF 
jitter problem -- if it's even a real problem -- is to simply avoid 
transmitting S/P-DIF signals over bandwidth-constrained channels in the 
first place. While this may be difficult in some professional 
applications where you have to go long distances, e.g., hundreds of 
meters, it ought to be easy in most consumer applications where you're 
only going a meter or so. Especially if the link is optical, as it often is.

If that's not possible, then I agree with the recommendations in the 
paper: tighten the loop bandwidth of the clock recovery PLL in the 
receiver so while it will continue to track the incoming clock, it won't 
attempt to track the faster jitter components. (Another way to look at 
this is that the local reference oscillator won't be forced to follow 
the higher frequency components in the incoming jitter.) As the paper 
correctly points out, this may slow lock-up or prevent it entirely if 
there is an excessive frequency offset between the incoming and local 
reference clocks, but a two-step acquire/track PLL can fix this.

But I'm still totally unconvinced there even *is* a jitter problem in 
the first place. His analyzes tend to assume absolute worst case, and 
even so the effect of jitter tends to be right at the level of the 
quantization noise in a 16-bit system. Considering that the vast 
majority of real-world audio sources, even the very best ones, have a 
dynamic range considerably worse than 16 bits, most people would 
conclude that jitter is simply not a problem worth fixing with hundreds 
of dollars of fancy accessories. And if anyone believes otherwise, all 
they have to do is to prove it with carefully controlled studies. 
Glowing testimonials and anecdotes from audiophiles won't do.

--Phil
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Re: [slim] Modifying squeezebox clock

2005-03-01 Thread Phil Karn
Julian Alden-Salter wrote:
1) The fact that my dac locks on with different qualities of lock when mp3
and flacs are played back. Suggesting that there is indeed some difference
in the spdif data stream between the two formats.
What do you mean by different qualities of lock? I've had no trouble 
with my squeezebox connected optically to a Sony V333ES receiver. Works 
fine in both MP3 and raw PCM mode.

There is an important difference between these two modes, though. A 
fixed-size buffer in the Squeezebox provides several seconds of network 
buffering at MP3 rates, but far less time at the much higher raw PCM 
speed. MP3 streams play fine, but I often hear music interruptions in 
raw PCM mode when the slimserver is doing other things (e.g., rebuilding 
the database.) Since most of my database is in FLAC format and is 
streamed in raw PCM format to the Squeezebox, this is a major concern 
for me.

Renicing all slimserver-related tasks to a higher priority usually stops 
the interruptions, but this is clearly not a very desirable fix 
especially since the slimserver sometimes likes to gobble up all 
available CPU, e.g., when it's rebuilding the database or when it just 
goes out to lunch for no reason.

It's clear that some attention should be given to real-time issues and 
thread scheduling priorities in the slimserver. Threads and processes 
involved in real-time playback activities should be given higher 
scheduling priority, whie background activities like rebuilding the 
database should be given a *lower* than normal priority. The 
UI/webserver should run at normal priority.

Perhaps you're simply having similar problems with buffer underruns when 
operating in raw PCM mode?

2) The 'subjective fact' that the squeezebox sounded worse than a cd
transport.
They sound *exactly* the same to me, assuming of course there are no 
buffer underruns.

Secondly we aren't talking about reclocking the dac. We are talking about
reclocking the squeezebox - specifically taking the spdif output, stripping
the old clock data out and adding a more accurate one to the data stream. If
I understand correctly this is what both the trichord and tent xo3 boards
do.
*Any* external DAC with a S/P-DIF input should do a good job of cleaning 
up its input clock. That's the entire purpose of a clock recovery PLL. 
If you think that the PLL in a particular external DAC is badly designed 
and has too much jitter, then try tightening the bandwidth of its loop 
recovery filter. That should work just as well -- if not better-- as 
buying an expensive standalone clock regenerator box, and at the cost of 
just a few component value changes.

The reason I say this should work even better than the external box is 
very simple. If, as that AES paper suggests, tight filtering of the 
composite S/P-DIF signal by the transmission line is the root cause of 
jitter, then it won't do any good to regenerate the clock in a separate 
box only to re-encode it as S/P-DIF and send it over another cable into 
the DAC; you'll just reintroduce the jitter! And if you use a high 
quality, wideband cable between the clock reconstructor and the DAC to 
minimize jitter, why not just use one directly between the Squeezebox 
and the DAC and omit the clock reconstructor box altogether?

As for not using a dac, well the analogue output stages and filters inside
the squeezebox are poor compared to that in a decent off board dac. Even
with the jitter / unknown problems effecting the spdif output it still
sounds better than the analogue outs from the squeezebox.
Really? They sound just fine to me. I just can't get excited over minor 
differences in filter design that are highly unlikely to be audible. I'm 
old enough to remember all the nonsense about linear phase lowpass 
filters when the CD first came out, in 1984. A bunch of us tried 
carefully controlled listening tests and *none* of us could tell *any* 
difference among players in a blinded test with carefully matched audio 
levels -- with one exception. The only one player that we could easily 
tell apart from the others was the Sony D-5, the very first Walkman 
mini-player. It had very noisy analog electronics after the DACs, and 
the hiss was easily heard.

A much more valid reason to use an external DAC is when the device whose 
DAC you're bypassing is in a system with a lot of digital noise running 
around. This can sometimes be true in your garden-variety PC sound card 
given that it's plugged into a motherboard with ~100A of supply current 
flowing through a high-end Pentium 4 only 10 cm away. But I sincerely 
doubt it's true in a small, low power box like the Squeezebox with its 
own power supply.

Now it's still possible that there's a ground loop or similar design 
flaw in the Squeezebox that introduces noise into the analog outputs, 
but I haven't heard anyone complain about that yet and I haven't noticed 
any such noise myself. This is in contrast to some older network audio 
players, such as the 

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