[slim] Browse Music Folder 'problems'
Hi, I’ve been using my sb2 and ss v6.0 for a few days now and there are 2 little problems that are bugging me. They are: 1) I have 2 hard drives on my music server one with songs from a-g and the other with h-z there are links to these drives (or directories on them in actual fact) in my ‘my music’ directory which is set as slim servers music directory. I’ve set things up so that ‘The Doors’ will go on the a-g drive however even though I have the ‘articles to ignore’ setting set to ignore ‘The’ and ‘the’, ‘The Doors’ still appears at the end of the directory list rather than with the ‘D’s’ as it did on 5.4 2) My music library is reasonably large (900 ish albums 10k+ songs) the thing is when I browse into either main directory (a-g or h-z) there is an annoying pause (presumably while some stuff is cached) I’ve tried the options I could see in the server controls that may have a bearing on this but can’t rid myself of the pause. This didn’t happen with 5.4 either. Any ideas what I can do to fix these minor problems or is it a ‘wait for an update’ jobbie? Cheers Julian. ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Squeezebox 2 audio quality.
I haven't specifically tried wavs but I can't tell any difference between server and client side flac processing (server side involves transcoding to wav's I believe). Personally I'd go (and have gone with) with flacs ripped as tracks and make sure they are properly tagged. If you truly have 'infinite' storage then either method is as valid as the other imho. Cheers Julian -Original Message- From: Mike Reeve [mailto:[EMAIL PROTECTED] Sent: 30 March 2005 23:53 To: discuss@lists.slimdevices.com Subject: Re: [slim] Squeezebox 2 audio quality. Natan & Nicki Tiefenbrun <[EMAIL PROTECTED]> writes: > I'm hooking my SB up to a Linn > DAC, and even with raw WAVs it doesn't sound half as good as the CD > player with digital out, so I assume it's to do with the clocking. Hi Just in case you are running a wireless set up ... ... have you checked (and I apologise if you have already done so) that you have set Player Settings > Bitrate Limiting to No Limit? [Recall that it defaults to transcoding to 320Kbps MP3 ... ... I forgot this once after changing my set up and had a moment of shock/panic until I remembered :-O] Mike ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] kinks in the SB2?
Title: Re: [slim] kinks in the SB2? Hi, When I was setting up my squeezebox2 I had to remove my behringer ultramatch (mk1) from between the sb2 and my dac. With the ultramatch (a jitter buster amongst other things) in the loop I got no sound and the emphasis light on both the dac and the behringer flashed randomly. Removing it got everything working correctly, not that it was a big problem for me as the sb2’s output is clean enough already. This was using a coaxial digital cable (shark wire terminated with bullet plugs). I use a Linksys wrt54g which has never crashed. The great thing is that now Sean and the guys have total control over the dsping and spdif output it should be fixable via a patch once they’ve worked out what’s going on. Cheers Julian. From: Mike Hartley [mailto:[EMAIL PROTECTED] Sent: 30 March 2005 04:23 To: Slim Devices Discussion Subject: RE: [slim] kinks in the SB2? Sean, I'm about to buy my wireless set up for SB2 now. Probably FLAC, definitely lossless, via Xandros (Debian based) Linux distro. Any other hardware to stay away from, or more importantly that is really stable? Mike -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: Tue 3/29/2005 9:49 PM To: Slim Devices Discussion Cc: Subject: Re: [slim] kinks in the SB2? > First, mine goes blank for no apparent reason sometimes. Blank screen. Does it keep playing audio? Can you control it from the web interface? > 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). That shouldn't be. Which receiver? > Plus, it won't connect to my wireless network--though the tech support > seems > to think it's my access point. Which access point? We tested numerous configs with several dozen access points. So far we've had only one issue that I know of, with a specific recent hardware rev of the Linksys BEFW11S4 - we're looking into it. ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
[slim] Squeezebox 2 audio quality.
Hi, Some of you may remember that I posed a few questions a while ago about the internal clock and digital output quality of the original squeezebox. I was running my mk1 via an audio synthesis dax-2 and lately a dax decade. Both of these dac’s have a digital signal quality indicator showing the dac’s ability to lock onto the incoming signal. There are 2 levels, a wide bandwidth level which allows pretty much any signal to be locked onto and a ‘crystal’ or ‘x’ lock which achieves a ‘quieter’ lock with higher quality digital signals. Unfortunately my mk1 playing flacs/wavs was unable to output an ‘x’lockable signal – although flac’s converted to mp3 did generate a digital signal that could be x-locked it seemed a bit pointless to convert flac to mp3 for ‘better quality’. I tried various tweaks including a monarchy dip, a behringer ultramatch and a slightly upgraded psu all of which failed to light the x-lock indicator. To further rub salt in the would a friend brought round his dpa cd transport which comprehensively annihilated the squeezebox in the quality stakes – although the ‘box was wy ahead in the convenience stakes. This morning the nice man from UPS dropped off my squeezebox 2 and after I’d paid the chancellor of the exchequers ransom I connected it up and away we went. After a small delay music issued forth and then click the ‘xlock’ lights up. Boy the grin nearly split my head in two. Over the day I’ve been listening to a lot of my music collection and must say that it’s a no brainer to say that the new squeezebox 2 is miles better than the mk1 from an ‘audiophile’ quality point of view. I would say that, from my admittedly fallible audio memory, the dpa transport could be considered still slightly ‘better’ but now they are in the same ballpark and one could conceivably prefer the squeezebox’s more relaxed presentation to the dpa’s more in your face manners – it would depend on the listener. So is that ‘better’ or just ‘different’ then? I’m going to try to arrange a rematch with the dpa to see how it really measures up. Still given that the squeezebox cost £200 and the dpa was over £1k new the squeezebox is ahead in my book. Cheers Julian. ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Squeezebox2 on the 31st?
JJ, I placed my order the day they announced and mine arrived this morning. Cheers Julian. -Original Message- From: JJ [mailto:[EMAIL PROTECTED] Sent: 29 March 2005 20:04 To: Slim Devices Discussion Subject: [slim] Squeezebox2 on the 31st? Are you guys still on track to ship the Squeezebox2 on March 31st? I'm placing an order for a Wireless Platinum and I'm pretty sure I'd just bust if it were delayed. :-) ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Modifying squeezebox clock
Hi, This is the stuff I found on audio synthesis' website... Following input selection two PLLs are used to extract and purify the word clock. An electrically and mechanically isolated crystal oscillator forms part of a sophisticated second order PLL, tightly bandpass filtering the recovered clock before applying it to the digital filter and DACs. A third digital loop is used to isolate jitter generated in the digital filter - all DAC loading and sample & hold timing is derived from this third digital PLL. Inputs are reclocked on multiple occasions before reaching the DAC itself. Low frequency jitter is the most difficult to eliminate so the three PLLs have been carefully optimised to boast a sub 1Hz jitter cut off frequency With the aid of multiple PLLs and intelligent muting we have made locking to jitter-ridden or off frequency transports routine, whilst maintaining automatic silent crystal locking for higher quality sources. >From here: http://www.audiosynthesis.co.uk/dax_decade.htm So much phlogiston to me though. I've actually upgraded recently to the decade from a dax-2 from the same company - this previous dac also exhibited similar behaviour with mp3 / pcm/flacs. I take on board the buffer stuff phil mentioned but I don't get stuttering and can't see how an intermittent buffer underrun would cause the dac to fail to xlock (crystal lock from above). Cheers Julian. -Original Message- From: Sean Adams [mailto:[EMAIL PROTECTED] Sent: 02 March 2005 05:09 To: Slim Devices Discussion Subject: Re: [slim] Modifying squeezebox clock On Mar 1, 2005, at 8:08 PM, Phil Karn wrote: > 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. >> There is a "just for your information" bit for clock precision in the s/pdif channel status data - that may be what it is. If so, it has nothing to do with actual clock precision, it's just a bit that says "I think I'm a high precision clock" or not. I'm not sure what we send for this bit but I could check it on my analyzer - at any rate, it does not affect functionality/performance in any way. Does your receiver (maybe in the instruction manual) say exactly what it's reporting? ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Modifying squeezebox clock
Phil, I take on board what you are saying but unfortunately don't know enough about the subject to say whether I agree or not. However there are 2 readily observed phenomena (one purely objective and one arguably subjective) which say there is something not right about the squeezebox's digital output. 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. 2) The 'subjective fact' that the squeezebox sounded worse than a cd transport. If jitter is not the cause of these observations they I'd ask you if you can propose an alternate reason for them. 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. 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. Cheers Julian. -Original Message- From: Phil Karn [mailto:[EMAIL PROTECTED] Sent: 01 March 2005 12:38 To: Slim Devices Discussion Subject: Re: [slim] Modifying squeezebox clock Triode wrote: > For some science, try the following: > > http://www.essex.ac.uk/ese/research/audio_lab/malcolmspubdocs/C41%20SPDIF%20 interface%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
RE: [slim] Modifying squeezebox clock
Robin, Unfortunately, I'm pretty new to the world of modifying components - until recently I was blindly climbing the Naim upgrade ladder. This means I don't have any way of measuring jitter at my disposal. Also the transports were brought round to my place where I hosted what is commonly known as a 'bakeoff' where we all sit down and give other peoples kit a listen at one place. I would love to know what is going on between flac and mp3 - I've e-mailed slim devices about this a couple of times and they've given me some suggestions but nothing has helped. I doubt it's a high priority for them as, as someone else pointed out, only about 1% of squeezebox users are going to be fussed. As I said I'm going to be trying a lightly modded roku in the near future so I'll post the results of that one. The other thing I was wondering was whether the pc used as the server would have any impact on things - I was going to try switching to linux to see if that made a difference. If none of the above make any difference then I'll be looking for an appropriate case to re-house the squeezebox in, in order to re-clock it. I know there is a high end transport hidden within the squeezebox dying to get out. Cheers Julian. -Original Message- From: Robin Bowes [mailto:[EMAIL PROTECTED] Sent: 01 March 2005 09:53 To: discuss@lists.slimdevices.com Subject: Re: [slim] Modifying squeezebox clock Julian Alden-Salter wrote: > Robin, > > >>What difference do you hear? > > > Without launching off into audiophile rambling about 'inner detail' and > 'timbre'. The dpa just sounded more realistic, detail was easier to hear and > transients (i.e. drum strikes) had more impact whilst the squeezebox sounded > muffled in comparison. That sounds like classic symptoms of jitter - smearing of the transients. > I'd have no qualms about trying to discern the > differences in a double blind test and I'd be pretty damn confident that I'd > be able to get a statistically significant result telling the two apart. To > give myself a bit of leeway with hyperbolae - the difference was like night > and day. The fact that 4 of us were in total agreement (even though the > tests were sighted and we all have wildly different tastes in music and > systems at home) should give some clue as to the magnitude of the > difference. Have you got anyway to measure jitter? If so, you may be in a position to perform an experiment to keep Phil happy :) Something like this: 1. Measure jitter from Squeezebox 2. Measure jitter from the dpa 3. Perform double-blind testing between the two sources I'd be very interested to hear about the results of such an experiment. >>Erm, mp3s are lossy so they will inevitably sound "worse". How are you >>comparing? > > > I'm not actually talking about sound quality here - flac/pcm still has the > legs on mp3 (just at 320 but it's a tough call at that bit rate). As I said > before my dac has low and high quality locks - one for high jitter signals > which isn't very picky and can lock onto pretty much any standard digital > signal you throw at it. There is also a higher quality lock which will only > 'XLOCK' if the signal received is low enough jitter / high enough quality. > Mp3's and the transports tested all 'XLOCK'ed. Flac / pcm's do not. This is > 100% verifiable and consistent and not subject to any subjectivity - I can > post pics somewhere if you like. So, you're saying that the SB digital out is different for flac vs. mp3, and that your dac will only XLOCK onto mp3s (and the output from your other transports)? That is certainly interesting information. Can you measure the jitter in the output when playing mp3s vs playing flac/pcm? >>I've got to say, the Tent clock chip upgrade looks very appealing. I'm >>planning an upgrade to my Art DI/O (4 x ALWSR PSUs: +/-15V, +5V digital >>and +5V analogue) and I might slip a clock chip upgrade in too. > > >>Let me know how you get on. > > > I'm going to be borrowing a prototype alwsr psu specifically designed for > the squeezebox in the near future. I'm not sure how big a difference it will > make as I feel a psu will mostly benefit the analogue output stage. I tried > a monarchy dip reclocking device but this made little or no difference so it > looks like the tent clock will be my next thing to try but the casing issue > makes things difficult. Yeah, you need to pull the guts out of the SB and mount it in another case if you're doing this level of modification. I haven't decided exactly what I'll be doing, but one option is to build the SB and the digital stage of my Art DI/O into a new chassis, along with several
RE: [slim] Modifying squeezebox clock
Robin, >What difference do you hear? Without launching off into audiophile rambling about 'inner detail' and 'timbre'. The dpa just sounded more realistic, detail was easier to hear and transients (i.e. drum strikes) had more impact whilst the squeezebox sounded muffled in comparison. I'd have no qualms about trying to discern the differences in a double blind test and I'd be pretty damn confident that I'd be able to get a statistically significant result telling the two apart. To give myself a bit of leeway with hyperbolae - the difference was like night and day. The fact that 4 of us were in total agreement (even though the tests were sighted and we all have wildly different tastes in music and systems at home) should give some clue as to the magnitude of the difference. >Erm, mp3s are lossy so they will inevitably sound "worse". How are you >comparing? I'm not actually talking about sound quality here - flac/pcm still has the legs on mp3 (just at 320 but it's a tough call at that bit rate). As I said before my dac has low and high quality locks - one for high jitter signals which isn't very picky and can lock onto pretty much any standard digital signal you throw at it. There is also a higher quality lock which will only 'XLOCK' if the signal received is low enough jitter / high enough quality. Mp3's and the transports tested all 'XLOCK'ed. Flac / pcm's do not. This is 100% verifiable and consistent and not subject to any subjectivity - I can post pics somewhere if you like. >I've got to say, the Tent clock chip upgrade looks very appealing. I'm >planning an upgrade to my Art DI/O (4 x ALWSR PSUs: +/-15V, +5V digital >and +5V analogue) and I might slip a clock chip upgrade in too. >Let me know how you get on. I'm going to be borrowing a prototype alwsr psu specifically designed for the squeezebox in the near future. I'm not sure how big a difference it will make as I feel a psu will mostly benefit the analogue output stage. I tried a monarchy dip reclocking device but this made little or no difference so it looks like the tent clock will be my next thing to try but the casing issue makes things difficult. R. -- http://robinbowes.com ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Modifying squeezebox clock
Having done a number of comparisons between various cd transports (dpa, arcam) and the squeezebox into my dac using the same digital interconnect I'd be interested to know peoples thoughts on what is causing the clear differences heard if not jitter. Power supplies have been mentioned and to an extent I would agree that they make a difference. Another theory I've heard is that somehow interference / noise is being transferred from some transports to the dac and thus causing degradation to the sound. Anyone got any alternatives? The thing that doesn't really make sense is that flacs (pcm / wavs) have a worse digital output than mp3's. I'm wondering if this is an internal rf noise issue as something will have to be working harder to process more data than with mp3's - or perhaps not. The thing I like about the tent xo3 is that it doesn't just feed a high quality clock to the decoder but it also corrects the spdif output after decoding. There is another board by a company called trichord which does a similar thing but it's more expensive. The biggest problem that I've got at the moment is how to fit the extra gubbins inside the squeezebox. I guess I'd have to re-box it but finding something that's a suitable size is difficult. Cheers Julian. ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Modifying squeezebox clock
There is some discussion on the subject here: http://www.pinkfishmedia.net/forum/showthread.php?s=&threadid=10685 I'm following this thread pretty closely as at some point I'll probably be adding the tent clock xo3 and the psu. This reclocks the decoder and the spdif output. Afaict the ram is only used for network latency not to de-jitter the output from the decoder chip. A friend is going to be bringing his moderately tweaked roku over for a shootout on my dac in a week or two so we'll see how things go there too. Cheers Julian. -Original Message- From: Robin Bowes [mailto:[EMAIL PROTECTED] Sent: 28 February 2005 15:10 To: discuss@lists.slimdevices.com Subject: Re: [slim] Modifying squeezebox clock Arthur Cheng wrote: > I have some idea of how to do it, but I am not too sure what the max > voltage that the chipMAS3539F can tolerate? It should be listed in here: http://www.rockbox.org/twiki/pub/Main/DataSheets/mas35x9f_1ds.pdf R. -- http://robinbowes.com ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
RE: [slim] Bit-correct digital out on Squeezebox?
Hi, I use an audio synthesis dax decade with my squeezebox. I must first point out that I'm extremely happy with the replay however a few weeks ago I invited some friends round to have a listen to the squeezebox and compare it to a couple of dedicated transports. The other transports were from dpa and Arcam both of which were judged (subjectively) to be better than the squeezebox by all present. Not a very scientific test I know but I agreed with the consensus and to my ears it wasn't a small margin. Now in the ultimate scheme of things a squeezebox / dac combo will blow a lot cd players away even some seriously expensive ones however I believe that it's jitter performance - especially with flac's is less than great. I say this because my dac (the aforementioned dax decade) has 2 methods of locking onto the digital signal - one which can lock onto low quality / high jitter signals and one that will lock onto better quality / low jitter signals. This is denoted by the display of lock or xlock in the dac's display. Without fail if I play flacs I only get a lock however if I play mp3's either native or transcoded from flacs I get the higher quality xlock. What's going on, I'm not sure but it seems to me that there is more performance within the squeezebox than is currently accessible. I'm currently looking into the possibility of improving the clock within the squeezebox which may help things. Of course for 200 quid it's a fantastic bit of kit but I don't think it's going to replace a true high end transport just yet. For me however the convenience of having my entire music library on hand far outweigh what at the end of the day is a 30%ish difference in quality. Cheers Julian. -Original Message- From: Michael Peters [mailto:[EMAIL PROTECTED] Sent: 25 February 2005 19:14 To: Slim Devices Discussion Subject: Re: [slim] Bit-correct digital out on Squeezebox? On Fri, 25 Feb 2005 16:34:03 +0100, Steinar Bjaerum <[EMAIL PROTECTED]> > > I am thinking about playing an uncompressed WAV, recording the signal at the > digital output and comparing the recording with the original WAV. > > Be careful when you are doing so - a lot of sound cards will resample digital input. AC97 cards do, for example (or so I've been told) I believe M-Audio Audiophile 2496 does not, I don't know about their other models. Again, that is what I've been told. I have not done any testing myself, mp3 at 192VBR is good enough for me (though I do archive lossless in flac). -- http://mpeters.us/ ___ Discuss mailing list Discuss@lists.slimdevices.com http://lists.slimdevices.com/lists/listinfo/discuss
