Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
JohnSwenson wrote: > I hope I can try and do this without pictures, I don't have time right > now to draw some nice pictures, so I'll try and be clear with the words. John, thanks for your additional explanation. Let me summarise what I think you're saying and I hope you can tell me if it's correct. Here are the stages involved when using oversampling: 1. Stuff zero samples between the actual samples. (Actually, any random values will do, zero just happens to be convenient, right?) 2. Use a digital filter to remove the artefacts above Fs/2. 3. Feed the oversampled signal (which now has a much higher Fs) into the D/A stage. 4. Resulting analogue signal has aliasing artefacts much higher up the frequency spectrum and so a much gentler analogue reconstruction filter may be used. Is that about right? Until now, I hadn't appreciated that step 2 was required. I must say that it had always seemed unintuitive that just stuffing extra samples into the data stream would shift the artefacts up, but I have never fully understood the mathematics and just accepted it based on the explanations of other authorities - who presumably were oversimplifying. Now you've explained it in greater detail, it actually makes sense. cliveb's Profile: http://forums.slimdevices.com/member.php?userid=348 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
cliveb wrote: > John, thanks for trying to explain, but I'm still confused. Your very > first paragraph is completely at odds with what I have always understood > to be the purpose of oversampling in a DAC: > > > I thought that the effect of oversampling (ie. zero-stuffing) was that > it moves the aliasing artefacts up the frequency spectrum. If you 4x > oversample a 44.1kHz signal, then the aliasing artefacts will begin at > 88.2kHz instead of 22.05kHz. Hence you can use a much gentler > reconstruction filter - indeed, you can in this case use EXACTLY the > same filter that you would use on a non-oversampled 176.4kHz signal. > > Are you saying that I've been misunderstanding the purpose of playback > oversampling all this time, and that when you oversample the aliasing > artefacts are NOT moved up the frequency spectrum? If that is the case, > then what IS the purpose of oversampling? I hope I can try and do this without pictures, I don't have time right now to draw some nice pictures, so I'll try and be clear with the words. So what causes the aliases in the first place? Think of a spectrum of the output of a good old fashioned ladder DAC chip running at 44.1, each time a new sample comes along the output (almost) instantaneously changes to a new value, the infamous "stair step" output. What does the spectrum of this look like? Each of those sharp edges going from one value to the next requires a series of high frequency harmonics to implement the sharp edge. These high high frequency harmonics beating with the audio signal frequencies are what create the aliases. It is purely a byproduct of the "sharp edges" in the stair step. Now lets try a 4X oversampling, we create 4 times as many samples, every fourth one being an original value and the rest being zero. Now take the spectrum of that, the sample rate is now 4 times higher, but the data only changes every 4 samples, the harmonics are at exactly the same frequencies, the amplitudes are MUCH greater because the sharp edges are now going from zero to the full sample value, not just the difference between sample values. So this step by itself has made things WAY WAY worse. The magic is what happens when you run this through the FIR filter. The filter in a nutshell puts values in those zero slots to produce a smoothly varying curve between the original samples. So what does this look like in the frequency domain? Well look at the spectrum of the zero stuffed signal, audio data up to 20KHZ and aliases and harmonics above 22.05KHz, what do you have to do? Get rid of all that stuff above 20KHz. of course! You need a filter that passes everything up to20KHz but blocks everything above 22.05KHz, this is the infamous brick wall filter. This process fills in the spaces between the original samples, it still has stair steps, but those staps are now coming at 176.4 and the height of the steps is much smaller. The spectrum of this shows audio up to 20KHz, then nothing up to 88KHz then harmonics of the sampling frequency going up from there, but these are now much lower in amplitude than the harmonics from the original 44.1 signal. The oversampling and zero stuffing BY ITSELF does not fix the situation, it just allows you to implement a filter which can filter out most of the above audio band stuff caused by the "stair step" in the original data. Again I hope this makes sense, it's a lot easier to grasp with the right pictures. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
John, thanks for trying to explain, but I'm still confused. Your very first paragraph is completely at odds with what I have always understood to be the purpose of oversampling in a DAC: JohnSwenson wrote: > A 44.1 stream still needs to get filtered with a sharp cutoff filter > close to 20KHz in order to properly supress aliases. This is because we > know that there is going to be real input data all the way up to 20KHz. > A filter for 176.4 can get away with a less steep filter because we > assume that there will be no (or VERY little) actual data close to > 88KHz. If you take that 44.1 stream and zero stuff it and feed it into a > 176.4 filter with a shallow filter function it's not not going to get > properly filtered. I thought that the effect of oversampling (ie. zero-stuffing) was that it moves the aliasing artefacts up the frequency spectrum. If you 4x oversample a 44.1kHz signal, then the aliasing artefacts will begin at 88.2kHz instead of 22.05kHz. Hence you can use a much gentler reconstruction filter - indeed, you can in this case use EXACTLY the same filter that you would use on a non-oversampled 176.4kHz signal. Are you saying that I've been misunderstanding the purpose of playback oversampling all this time, and that when you oversample the aliasing artefacts are NOT moved up the frequency spectrum? If that is the case, then what IS the purpose of oversampling? cliveb's Profile: http://forums.slimdevices.com/member.php?userid=348 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
dsdreamer;699293 Wrote: > I have a theory that any filter steep enough to get from 0dB at 20kHz to > -80dB at 22kHz will somehow be detectable to the human ear/brain. It > needs a lot of FIR taps to make such a filter, and the ultrasonic > pre-ringing that is likely to be associated with linear-phase design > may be somehow detectable. (See attachments.) Such a filter is needed > twice: firstly to allow the track to be down-sampled to 44.1kHz from > the studio format for distribution; secondly, a similar filter is > needed as part of the up-sampling process in the DAC as described by > John S. > > Intuitively, I expect the convolution of two such filters to be > harmful, not due to removal of high frequency information that we > cannot detect, but in the way the brain interprets wavefronts and > timing cues. This is partly the theory behind so-called apodizing > filters. > > I'd be interested in any justified assertions as to why this could or > could not be an impairment worth addressing. Try for your self , in the 192k is harmful tread I managed to get the SoX resampler installed at my desktop and took a couple a what it bought is good sounding 24/96 files and converted to a number of rates including 16/44.1 ( I used files of true digital PCM recordings at >= 24/96 all else would obscure the experiment, the fact that a large number of hirez recordings are "fakes" would not help ). I'm not finished with that experiment yet . My theory is that there may be at least some recording where it matter, but I have not been able to tell a conclusive difference yet . It's not impeccable science it is sort of semi blind I just shuffle a bunch on copies of converted and non converted files and just guess what I'm listen to. One have listen repeatedly ,hearing throw curve balls you sometimes think the same file sound different ,but that's just how the brain works. But you need not to look further than Boston audiophile societies famous inline resampling experiment for evidence on how hard this is to detect this in practice even if the ringing looks like it does. ( the put an 16/44 AD-DA device in a hirez playback chain and besides some rare case of the higher noise of the equipment being detachable in someone's rig no one heard anything ) -- Mnyb Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3 sub. Bedroom/Office: Boom Kitchen: Touch + powered Fostex PM0.4 Misc use: Radio (with battery) iPad1 with iPengHD & SqueezePad (in storage SB3, reciever ,controller ) http://people.xiph.org/~xiphmont/demo/neil-young.html Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
> This by the way is why I think high sample rate files are popular with > some people, its not that the file has more high frequency information, > but that the DAC is using a better sounding filter at that sample rate. I have a theory that any filter steep enough to get from 0dB at 20kHz to -80dB at 22kHz will somehow be detectable to the human ear/brain. It needs a lot of FIR taps to make such a filter, and the ultrasonic pre-ringing that is likely to be associated with linear-phase design may be somehow detectable. (See attachments.) Such a filter is needed twice: firstly to allow the track to be down-sampled to 44.1kHz from the studio format for distribution; secondly, a similar filter is needed as part of the up-sampling process in the DAC as described by John S. Intuitively, I expect the convolution of two such filters to be harmful, not due to removal of high frequency information that we cannot detect, but in the way the brain interprets wavefronts and timing cues. This is partly the theory behind so-called apodizing filters. I'd be interested in any justified assertions as to why this could or could not be an impairment worth addressing. +---+ |Filename: mag_plot.jpg | |Download: http://forums.slimdevices.com/attachment.php?attachmentid=13212| +---+ -- dsdreamer -- "Dreamer, easy in the chair that really fits you..." dsdreamer's Profile: http://forums.slimdevices.com/member.php?userid=12588 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
John, do you mind name a few DACs on the market that you think have a decent implementation? -- touchporter touchporter's Profile: http://forums.slimdevices.com/member.php?userid=54882 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
adamdea;699192 Wrote: > The conclusion so far is that john Swenson considers that digital > filters don't sound good unless they are on separate FPGAs -he doesn't > like the ones on the dac ships themselves (also IIRC some time ago he > said that the ones on the Sabre Dac are OK.) > > AS I understand it conventional engineering thinking is that a well > designed DAC is not source dependent. Well designed does not mean > expensive. > > Curiously the ones which might be source dependent are those designed > for/by audiophiles. At that point, I suggest that you draw your own > conclusions. > > [by way of a side turning i have been meaning for a while to see > whether I could test JS's hypothesis by implementing a sinc like > (linear phase, at least 80 db attenuation at nyquist) filter in Sox > which would mean that the half band filter in my MF DAC did very little > (there being no frequencies for it to attenuate- although I don;t > suppose i could really cut it off entirely without having -80db at > 20kHz; but presumably a transition band 20-22kHz would mainly take it > out of the equation). I assume that filtering in Sox should work at > least as well as the separate FPGA. > > I have never got round to working out how to do this though especially > given the upsampling to 96Khz which sox does at present courtesy of > phil's setting for inguz.] I would not say that the only DACs that are source sensitive are ones designed by audiophiles. Any DAC that uses one of the off the shelf S/PDIF receivers and directly feeds the data and clock into a DAC chip is going to be quite sensitive to the outside world. There are a LOT of those out there and many are still being built that way. A rough estimate is that maybe half of the DACS being made today are using ASRC chips, this helps significantly but does not completely eliminate outside influence. There are other factors involved such as ground plane noise and the local clock itself. (the groundplane noise frequently modulates the local clock) The sabre chips are interesting, they seem to be the only ones that have fairly decent digital filters (not perfect but quite good), BUT they have a builtin ASRC which causes the same sort of issue I have been discussing with ASRCs. There is one DAC maker I know of who has figured out how to bypass the internal ASRC and uses his own very low jitter circuit so the ASRC is not needed, this DAC sounds really good but is very expensive. Using external software (SOX etc) to perform the filtering is a very viable alternative, as long as the software uses enough precision in its internal computations. A 32 bit float is NOT enough precision, a double (64 bit float) IS sufficient. But after you do this you don't want it going into through the same digtial filter built into the DAC chip. Fortunately many chips use different filtering at 176/192 than they do at lower sample rates, some don't do any filtering at the highest sample rates. So if you use software to upsample to 176/192 you might actually have a good chance of bypassing many of the problems with the 44.1 filters in most DAC chips. This by the way is why I think high sample rate files are popular with some people, its not that the file has more high frequency information, but that the DAC is using a better sounding filter at that sample rate. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
adamdea;699206 Wrote: > Audiophiles tend to like things which sound different and can be tweaked > like turntables, valves and nos dacs. this is because the hobby consists > of fiddling and listening to differences in kit while pretending to > listen to music. No fiddly-listen-difference, no hobby. > > Of course it doesn't actually have to be a real difference. If that's the case, and if we were to take your definition as gospel, then I'm not an audiophile. I'm only interested in achieving the quality playback that will bring real differences in how I experience music. Cynics insist that there is no real difference, but it's like insisting that there is no real difference between blue skies and grey skies. It's all in the mind, you know. -- magiccarpetride magiccarpetride's Profile: http://forums.slimdevices.com/member.php?userid=37863 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
adamdea;699206 Wrote: > Audiophiles tend to like things which sound different and can be tweaked > like turntables, valves and nos dacs. this is because the hobby consists > of fiddling and listening to differences in kit while pretending to > listen to music. No fiddly-listen-difference, no hobby. > > Of course it doesn't actually have to be a real difference. :-). So so true... -- Phil Leigh You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal... Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables Stax4070+SRM7/II phones Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything. Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Audiophiles tend to like things which sound different and can be tweaked like turntables, valves and nos dacs. this is because the hobby consists of fiddling and listening to differences in kit while pretending to listen to music. No fiddly-listen-difference, no hobby. Of course it doesn't actually have to be a real difference. -- adamdea adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
adamdea;699192 Wrote: > > Curiously the ones which might be source dependent are those designed > for/by audiophiles. At that point, I suggest that you draw your own > conclusions. > LOL! A real audiophile bargain. "Regarding the manufacturing defect suffered by our original review sample of the Zanden 5000S, the second sample was free from this problem, and offered very much better low-frequency linearity and a much lower output impedance. Other than the slight difference in hum, its performance was basically the same in both inverting and noninverting polarity conditions. However, its digital-domain performance is still rather old-fashioned, particularly regarding word-clock jitter rejection and low-level linearity error. An enigma.John Atkinson" http://www.stereophile.com/content/zanden-5000-mkivsignature-da-converter-2000-premium-cd-transport-second-sample-measurements -- NoRoDa SBT | Teddy Pardo TTouch | Rega DAC | Audionet SAM V2 | Sonus Faber Domus Grand Piano | NoRoDa's Profile: http://forums.slimdevices.com/member.php?userid=49139 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
The conclusion so far is that john Swenson considers that digital filters don't sound good unless they are on separate FPGAs -he doesn't like the ones on the dac ships themselves (also IIRC some time ago he said that the ones on the Sabre Dac are OK.) AS I understand it conventional engineering thinking is that a well designed DAC is not source dependent. Well designed does not mean expensive. Curiously the ones which might be source dependent are those designed for/by audiophiles. At that point, I suggest that you draw your own conclusions. [by way of a side turning i have been meaning for a while to see whether I could test JS's hypothesis by implementing a sinc like (linear phase, at least 80 db attenuation at nyquist) filter in Sox which would mean that the half band filter in my MF DAC did very little (there being no frequencies for it to attenuate- although I don;t suppose i could really cut it off entirely without having -80db at 20kHz; but presumably a transition band 20-22kHz would mainly take it out of the equation). I assume that filtering in Sox should work at least as well as the separate FPGA. I have never got round to working out how to do this though especially given the upsampling to 96Khz which sox does at present courtesy of phil's setting for inguz.] -- adamdea adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
What is the conclusion so far? There may be DACs that are so dependant on the incoming datastream beeing perfect that you may hear differences between settings in the SBT? Even with your eyes closed? Few DACs that incorporate the near impossible task of getting it perfect, it's hard labor and not for everyone to make it happen and no way your neighbour can buy it at the local store. It's a tough thought, that a SBT can be bought anywhere and is a good digital source right out of the box. Go analog guys, lots of secrets for the audiophiles hidden in the vinyl grooves. Regards -- NoRoDa NoRoDa's Profile: http://forums.slimdevices.com/member.php?userid=49139 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
JohnSwenson;699117 Wrote: > My objection to ASRC is that the current implementations out there don't > seem to sound good to me. I don't have an objection to the concept, but > I haven't heard an actual implementation that I like. My guess is that > it's the issue I have with almost all digital filters in commercial > audio chips in general: they seem to have compromised implementations, > my guess is in order to cut costs in the chips. > > So far I have not heard a delta-sigma DAC chip that can beat a 1704, > BUT if done properly they can come quite close. The big issue is to > bypass the internal digital filter and instead use a properly > implemented digital filter. > > For example I have taken a 1794 (or was it the 1792, I don't remember > right now) and fed it from an FPGA, I could either send it I2S and use > its own digital filter or put my own digital filter in the FPGA and > bypass the internal filter. Note this filter was a straight forward > SINC brickwall filter, nothing fancy. The FPGA filter sounded much > better. I did this listening with a bunch of people under blind > conditions and everyone prefered the FPGA filter. So even though both > filters were supposed to be doing the same thing something was > different. > > Two obvious possibilities for the difference: the internal filter is > somehow not doing what it should, or the extra processing of the > internal filter is generating more noise on the internal PS and ground > traces causing noise which is messing up the sound. > > I tested the second possibility by putting the DAC chip in external > mode and using a DF1704 digital filter instead of the FPGA filter, it > sounded almost identical to the internal filter. This tends to rule out > the extra processing going on in the DAC chip as the culprit. This > leaves the implementation of the digital filter itself as a prime > suspect. > > I am guessing that the designers of these chips cut corners in the > implementation of the digital filters. An implementation of the proper > filter for 44.1 sample rate takes a fair amount of horsepower, which > can lead to a chip costing more than the management prefers. Thus the > designers are under preasure to figure out how to cut corners to > decrease cost, but still meet the specifications. I know from > experience that this is quite common in the industry, I have personally > worked on quite a few DSP chips (mostly for image processing) and there > has always been preasure from management to figure out how to make > things "good enough" to keep costs down. > > Another telltale sign is the graphs that the manufacturers publish for > their filter functions: they don't look anything like what a simple > SINC filter should look like. They are doing something else, I'm not > exactly sure what, I'm not good enough at DSP theory to reverse > engineer the hardware from the graphs, but I do know its not a simple > SINC. > > When I do a simple SINC in an FPGA it sounds much better, but does take > a lot of resources, so my guess is that the designers of these chips are > getting fancy, using something other than a simple SINC in order to get > the spec sheet numbers they are after and still fall within the cost > parameters the management wants, and somehow this does not sound as > good. > > And no I have not done this with every DAC chip on the planet, but I > have done it with several. In order to do my test it has to be a chip > that allows you to turn off the internal filter and use an external > one. Many DAC chips do not allow this. But I have done enough of this > and heard the difference between my simple filter and what is inside > the chips to get a good feeling for difference in sound. I have > listened to quite a few of the chips that don't let you disable the > internal filter and I hear a similar sound to what I was getting from > the chips where I could use an external filter. > > So my conclusion is that this compromise is pretty pervasive in the > industry. > > I guess all this is a very long winded way of saying that yes a I think > you can get very good results out of a delta sigma DAC chip, you just > need one that lets you use an external digital filter. > > And BTW I think this is also a major reason for the NOS movement. > Going back to old DAC chips without digital filters seems to let > parts of the music "through" which normally somehow get messed up by > the internal digital filters. Of course then you have aliasing all over > the place. For some people its worth the tradeoff. > > Of course the proper solution is to use proper digital filters so you > can have the best of both. > > John S. Thanks John -- adamdea adamdea's Profile: http://forums.slimdevices.com/member.php?userid=37603 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list au
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
adamdea;699021 Wrote: > John, I have two questions > 1) given your objection to ASRC, does it follow that not only is it > necessary to use a different method of cleaning up the clock, it is > necessary to use a multibit dac chip- won't a delta sigma chip require > rather more drastic sample rate conversion? Is it just the > asynchronousness (asynchronicity?)which you object to? I appreciate > that ASRC may end up slightly changing the information in extreme > cases, whereas all SRC will alter the nominal data values. > 2) assuming that we use the buffer and slowly adjustable clock option, > why now should jitter and/or noise in the S/PDIF signal make any > difference to the output- is it just the ground plane noise? presumably > you agree that it is possible to read the data perfectly and to have a > clock which is not derived from the S/PDIF stream. My objection to ASRC is that the current implementations out there don't seem to sound good to me. I don't have an objection to the concept, but I haven't heard an actual implementation that I like. My guess is that it's the issue I have with almost all digital filters in commercial audio chips in general: they seem to have compromised implementations, my guess is in order to cut costs in the chips. So far I have not heard a delta-sigma DAC chip that can beat a 1704, BUT if done properly they can come quite close. The big issue is to bypass the internal digital filter and instead use a properly implemented digital filter. For example I have taken a 1794 (or was it the 1792, I don't remember right now) and fed it from an FPGA, I could either send it I2S and use its own digital filter or put my own digital filter in the FPGA and bypass the internal filter. Note this filter was a straight forward SINC brickwall filter, nothing fancy. The FPGA filter sounded much better. I did this listening with a bunch of people under blind conditions and everyone prefered the FPGA filter. So even though both filters were supposed to be doing the same thing something was different. Two obvious possibilities for the difference: the internal filter is somehow not doing what it should, or the extra processing of the internal filter is generating more noise on the internal PS and ground traces causing noise which is messing up the sound. I tested the second possibility by putting the DAC chip in external mode and using a DF1704 digital filter instead of the FPGA filter, it sounded almost identical to the internal filter. This tends to rule out the extra processing going on in the DAC chip as the culprit. This leaves the implementation of the digital filter itself as a prime suspect. I am guessing that the designers of these chips cut corners in the implementation of the digital filters. An implementation of the proper filter for 44.1 sample rate takes a fair amount of horsepower, which can lead to a chip costing more than the management prefers. Thus the designers are under preasure to figure out how to cut corners to decrease cost, but still meet the specifications. I know from experience that this is quite common in the industry, I have personally worked on quite a few DSP chips (mostly for image processing) and there has always been preasure from management to figure out how to make things "good enough" to keep costs down. Another telltale sign is the graphs that the manufacturers publish for their filter functions: they don't look anything like what a simple SINC filter should look like. They are doing something else, I'm not exactly sure what, I'm not good enough at DSP theory to reverse engineer the hardware from the graphs, but I do know its not a simple SINC. When I do a simple SINC in an FPGA it sounds much better, but does take a lot of resources, so my guess is that the designers of these chips are getting fancy, using something other than a simple SINC in order to get the spec sheet numbers they are after and still fall within the cost parameters the management wants, and somehow this does not sound as good. And no I have not done this with every DAC chip on the planet, but I have done it with several. In order to do my test it has to be a chip that allows you to turn off the internal filter and use an external one. Many DAC chips do not allow this. But I have done enough of this and heard the difference between my simple filter and what is inside the chips to get a good feeling for difference in sound. I have listened to quite a few of the chips that don't let you disable the internal filter and I hear a similar sound to what I was getting from the chips where I could use an external filter. So my conclusion is that this compromise is pretty pervasive in the industry. I guess all this is a very long winded way of saying that yes a I think you can get very good results out of a delta sigma DAC chip, you just need one that lets you use an external digital filter. And BTW I think this is also a major reason for the NOS movement. Going back to old DA
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
The music will have passed through a few ASRC's on its way to the CD or download that you purchased... Don't feel too bad about this though ... It's just another part of the enigma -- Phil Leigh You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal... Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables Stax4070+SRM7/II phones Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything. Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
JohnSwenson;698661 Wrote: > The recent questions in this thread are actually very important and are > at the crux of a LOT of discussion in audiophiledom these days. I'll > try and take a stab at answering some of these based on what I have > uncovered by designing my own DAC over the last 10 years or so and > listening to and trying to analyze many others "out there". > > There are really only three external world influences that can affect > the sound coming out of the DAC: > > 1) the bits! > 2) timing and noise on the input signal > 3) noise on the power supply (this includes ground noise) > > There can also be EMI from outside sources getting into the box, but > for most reasonable designs this is such a low level I'm not going to > include it. (primarily because the measures used to screen a device so > it meets regulatory emission limits, means it also screens out external > influences) > > We cannot completely ignore #1 at this point since some of the most > popular methods for decreasing jitter DO change the bits, I'll get into > that later. > > #2 is where most of the discussion centers around. #3 is usually > ignored but can be a significant contributor. > > So I'll talk about #2 first. Lets look at coax first. The signal is an > analog voltage and current, ones and zeros are a way of interpreting the > different voltage levels. The voltage on the wire takes time to change > from a zero to a one and vice-versa. When its at a high or low voltage > level there can be noise "riding on top" of the desired signal. Higher > frequency noise shows up as voltage variations (wiggles) in the "steady > state" areas between the transitions. In addition you can have low > frequency noise (usually power supply related) which slowly moves the > whole signal up and down. Then there are what are called "reflections", > when these sharp changes in voltage hit an impedance difference > (connectors, cables, components on a board) part of the signal gets > reflected with an inverted sense back down the other direction of the > wire. If there is also a discontinuity at the transmitter end the > signal gets reflected back AGAIN, so now its sitting on top of whatever > the transmitter is NOW sending. A pulse can go back and forth several > times before it damps out. > > The result of all this is that the signal at the receiver chip is far > from perfect. The total result of all of the above makes it very > difficult for the receiver chip to determine exactly when that signal > changes from a zero to a one (and vice-versa). Figuring this out is a > very important part of a receiver. There are several different methods > used to try and do this well even with all this junk on the signal > trying to make it difficult to do. Some are more successful than > others. Some cost more money than others. Some have drawbacks which > make then not very popular (like it takes a couple seconds to lock onto > a stream) > > The engineering comunity has pretty much settled on one way of doing > this which is used for at least 99% of inputs. This approach gives > somewhere between 200ps and 50ps of jitetr on the recovered clock. (ALL > methods are bit perfect) As is these receivers are fairly susceptible to > changes in the crud on the input signal. The jitter on the output clock > (and its spectrum) will vary significantly with changing conditions of > the input signal. > > There have been a number of techniques used in different DACs to try > and improve this by employing different techniques AFTER the primary > input receiver. BUT there is another process going on which has been > almost completely ignored, the amount and timing of current drawn by > that receiver chip varies significantly with whats going on with the > input signal. These current changes cause noise in the power supply > rails AND in the ground plane of the DAC. Some designers will use > separfate regulators for the receiver chip and for the rest of the DAC, > but almost everybody ignores the noise on the groundplane. This noise > can cause jitter on the clock and can get into the analog signal > directly. So even if the designers spend a ton of effort to "clean up" > the signal coming out of the receiver chip, the input signal condition > can still affect the audio output. > > There are three primary methods used to clean up the clock from the > receiver chip: > 1) separate narrow band PLL > 2) buffer with slowly adjustable low jitter clock > 3) ASRC > > #1 used to be fairly popular and a number of "reclocking boxes" of the > past were built using this technology. The results can be quite good, > but there is a tradeoff, the lower the jitter the longer it takes to > lock on to a signal. > > #2 can produce extremely good results, but is much more costly to > implement and usually takes a LOT of development time. Because of this > very few have ever been actually built. > > #3 is the ruling darling of the industry. It seems like a slam dunk
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
hmm. Paul Frindle (DSP Expert and Digital Audio Guru had this to say about DACs) (posted with his permission) I trust his word over most others(well, he and Bruno Putzeys- who was the head of Engineering at Phillips at one time) nothing at all about "noise" There are only 2 things a DAC responds to; 1) the data we feed to it , 2) the timing information it gets. For the data; data is data - there is no possibility that identical data sets could ever produce a different sound, regardless of their origins. In short, numbers are numbers - and if they are the same, they are the same - period. For timing it's a slightly different matter, because it is essentially and analogue signal - it's properties (i.e. rate) are analogue in concept. So there is a slight possibility that interference on the clock signal can affect the DACs performance, if the timing is modulated in some way, by slightly changing the rate of plyout with time. Of course a good DAC system will circumvent this possibility by using it's own internal clock and some buffering of the data - so that the DAC's timing follows the filtered average rate of the input timing - such that short term timing rate variations (due to interference) do not make it through. I.e. it will synchronise to the input clock, rather than simply passing the input clock straight into the DAC.. You can almost think of it as stacking up the data as it ccomes in and playing it out using it's OWN high quality clock set to the same rate. However - as you can imagine, price competition tends to rule out anything that might increase the cost of the product - and so you will have to spend extra money on your DAC system to get this.. So where does the timing modulation of the clock rate come from (sometimes called jitter)? First on the list are wires and connections. Line frequency hum, RF and interference getting into clock cables will modulate the timing at the recieving end. If you DAC doesn't reject them (as above) performance may suffer. Next on the list is bad design. For instance, in consumer players where the DAC is within the player box, power supply modulation from bad design may cause internal circuits to interfere with each other. For instance if the motor servo is being varied by slightly eccentric discs and/or wobbly ones that stress the focus servo, the cyclic changes to current draw my affect the clock oscillator if the power supply is badly desogned etc.. So although the data is correctly read, the DAC itself may perform less well than it might, due to internal design flaws.. And of course this is true of any electronic system... So there you go - it's a simple as that - no PHD required :-) In summary: Data itself coming from different systems cannot cause a change in sound - if the data values are both identical. But hardware may perform slightly differently if the timing integrity is compromised. Having an external DAC does not avoid the issue - unless it's a very good one with internal timing re-clocking. In which case it will be effectively immune from timing errors and will sound the same whatever way it gets connected, providing the data is correct. So instead of worrying about data coming from Macs or PCs and/or a whole load of hot air from HiFi zealots filling pages with 'waffle' - get yourself a high quality DAC and say bye-bye to the whole discussion :-) I hope this helps.. Paul -- TheOctavist Vortexbox>SBT(stock)>>Forssell MDAC-2>>>Klein and Hummell 0300D Sota Sapphire/Lyra Kleos>>Bespoke Valve Phono Stage>>Mastersound Due Venti>>Link Audio K100 TheOctavist's Profile: http://forums.slimdevices.com/member.php?userid=52700 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Soulkeeper;698684 Wrote: > Tuning the server, OTOH, is an entirely different question. > > Which I will not even begin to discuss right now; I just wanted to > point out that those are two wildly, insanely, flabbergastingly > different things. While I don't really consider myself technically qualified to answer. intuitively, I'd agree with you. -- rgro Rg System information Main: PS Audio Quintet > Vortexbox > Teddy Pardo PS, Touch (wired) > Toslink > Rega DAC > LFD LE IV Signature amp > VA Mozart Grands > REL Acoustics R305. Home Theatre: Duet/SBR (Wired) > Pioneer VSX 919 > Energy Take 5 Classic 5.1. SBS 7.7.2 r33908 on a Vortexbox Appliance, V 2.0, Touch: FW 7.7.1 r9558. Duet: FW 77. rgro's Profile: http://forums.slimdevices.com/member.php?userid=34348 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
rgro;698683 Wrote: > This is the crux of the disagreements Well, I don't know about that. Tuning the Touch in any way imaginably may conceivably affect the sound. I don't think anyone will dismiss that possibility 100%. Tuning the server, OTOH, is an entirely different question. Which I will not even begin to discuss right now; I just wanted to point out that those are two wildly, insanely, flabbergastingly different things. -- Soulkeeper Noise < Music < Silence Soulkeeper's Profile: http://forums.slimdevices.com/member.php?userid=35297 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
John---thank you. This was very informative. Over on Pinkfish there's a fellow, Brian, with the handle "Turtle", that's either your twin brother, or your alter ego! I'm trying to wrap my brain around all this and it's not easy...so forgive me. Is it accurate to say the following: * The argument that 1's and 0's cannot be altered is true in it's most basic form, but there can actually be measurable quality differences in how those 0's and 1's are both sent and received from a clocking perspective. * If the above is accurate, the quality of those differences in 0's and 1's can potentially lead to audible differences in sound. How much of this variablility is needed to produce audible changes is it not, the same argument as to how much jitter is actually audible and what audible form does that take??? * The question I've been trying to get answered is: can the "tuning" (for lack of a better word) of the priority settings and buffer size in the Touch actually effect the quality of the digital output signal of the Touch? In other words, will changing those values potentially alter the signal enough that it will affect how the dac will both process that signal and output it in its analog stage (thus leading to sound quality differences)? It certainly seems like altering those values can produce unambiguously audible negative changes i.e, distortions, clicks, etc. But, would it be possible that different combinations of priority settings and buffer sizes would produce different positive sound qualities---much like some of the digital filters that are implemented in many of the dac chips? Changes that may be on the more subtle side, which are nonetheless real changes, but for which the preference of one or the other combination would simply be a matter of personal taste? This is the crux of the disagreements (to put it mildly) between Soundcheck, SBGK (and others), and numerous other folks that say that priority/buffer tuning is a complete crock. -- rgro Rg System information Main: PS Audio Quintet > Vortexbox > Teddy Pardo PS, Touch (wired) > Toslink > Rega DAC > LFD LE IV Signature amp > VA Mozart Grands > REL Acoustics R305. Home Theatre: Duet/SBR (Wired) > Pioneer VSX 919 > Energy Take 5 Classic 5.1. SBS 7.7.2 r33908 on a Vortexbox Appliance, V 2.0, Touch: FW 7.7.1 r9558. Duet: FW 77. rgro's Profile: http://forums.slimdevices.com/member.php?userid=34348 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Soulkeeper;698674 Wrote: > Can you please elaborate on ASRC? What is it? (async clock recovery? > adaptive clock recovery something? if so, what is that and how does it > work, can you point me to some resources?) > Asynchronous Sample Rate Converter - try the thread on diyaudio: http://www.diyaudio.com/forums/digital-source/28814-asynchronous-sample-rate-conversion.html -- Triode Triode's Profile: http://forums.slimdevices.com/member.php?userid=17 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Can you please elaborate on ASRC? What is it? (async clock recovery? adaptive clock recovery something? if so, what is that and how does it work, can you point me to some resources?) You're one of the very few people on this forum that I consider knowledgable in the field of DAC, so I hope this question doesn't bore you too much or insult you ... -- Soulkeeper Noise < Music < Silence Soulkeeper's Profile: http://forums.slimdevices.com/member.php?userid=35297 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
The recent questions in this thread are actually very important and are at the crux of a LOT of discussion in audiophiledom these days. I'll try and take a stab at answering some of these based on what I have uncovered by designing my own DAC over the last 10 years or so and listening to and trying to analyze many others "out there". There are really only three external world influences that can affect the sound coming out of the DAC: 1) the bits! 2) timing and noise on the input signal 3) noise on the power supply (this includes ground noise) There can also be EMI from outside sources getting into the box, but for most reasonable designs this is such a low level I'm not going to include it. (primarily because the measures used to screen a device so it meets regulatory emission limits, means it also screens out external influences) We cannot completely ignore #1 at this point since some of the most popular methods for decreasing jitter DO change the bits, I'll get into that later. #2 is where most of the discussion centers around. #3 is usually ignored but can be a significant contributor. So I'll talk about #2 first. Lets look at coax first. The signal is an analog voltage and current, ones and zeros are a way of interpreting the different voltage levels. The voltage on the wire takes time to change from a zero to a one and vice-versa. When its at a high or low voltage level there can be noise "riding on top" of the desired signal. Higher frequency noise shows up as voltage variations (wiggles) in the "steady state" areas between the transitions. In addition you can have low frequency noise (usually power supply related) which slowly moves the whole signal up and down. Then there are what are called "reflections", when these sharp changes in voltage hit an impedance difference (connectors, cables, components on a board) part of the signal gets reflected with an inverted sense back down the other direction of the wire. If there is also a discontinuity at the transmitter end the signal gets reflected back AGAIN, so now its sitting on top of whatever the transmitter is NOW sending. A pulse can go back and forth several times before it damps out. The result of all this is that the signal at the receiver chip is far from perfect. The total result of all of the above makes it very difficult for the receiver chip to determine exactly when that signal changes from a zero to a one (and vice-versa). Figuring this out is a very important part of a receiver. There are several different methods used to try and do this well even with all this junk on the signal trying to make it difficult to do. Some are more successful than others. Some cost more money than others. Some have drawbacks which make then not very popular (like it takes a couple seconds to lock onto a stream) The engineering comunity has pretty much settled on one way of doing this which is used for at least 99% of inputs. This approach gives somewhere between 200ps and 50ps of jitetr on the recovered clock. (ALL methods are bit perfect) As is these receivers are fairly susceptible to changes in the crud on the input signal. The jitter on the output clock (and its spectrum) will vary significantly with changing conditions of the input signal. There have been a number of techniques used in different DACs to try and improve this by employing different techniques AFTER the primary input receiver. BUT there is another process going on which has been almost completely ignored, the amount and timing of current drawn by that receiver chip varies significantly with whats going on with the input signal. These current changes cause noise in the power supply rails AND in the ground plane of the DAC. Some designers will use separfate regulators for the receiver chip and for the rest of the DAC, but almost everybody ignores the noise on the groundplane. This noise can cause jitter on the clock and can get into the analog signal directly. So even if the designers spend a ton of effort to "clean up" the signal coming out of the receiver chip, the input signal condition can still affect the audio output. There are three primary methods used to clean up the clock from the receiver chip: 1) separate narrow band PLL 2) buffer with slowly adjustable low jitter clock 3) ASRC #1 used to be fairly popular and a number of "reclocking boxes" of the past were built using this technology. The results can be quite good, but there is a tradeoff, the lower the jitter the longer it takes to lock on to a signal. #2 can produce extremely good results, but is much more costly to implement and usually takes a LOT of development time. Because of this very few have ever been actually built. #3 is the ruling darling of the industry. It seems like a slam dunk, you have a local very low jitter clock, feed it to the ASRC and out comes a stream that is synchronized to that clock. What could be better? Unfortuntely this process DOES change the bits! That is it's whole p
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Phil Leigh;698496 Wrote: > It can be, but is highly system dependent - I've always suggested people > try toslink (assuming they haven't removed the connector!) vs. spdif to > see which they prefer in THEIR system. Yes it's a trade off some TOS components are not that fast theoretically more jitter BUT galvanically isolated which means less electrical noise creeping in. And if the TOS implementation is not that well enginereed some noise could actually be transformed to ligth to, but now we are at the usual nitpicking about details again... -- Mnyb Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3 sub. Bedroom/Office: Boom Kitchen: Touch + powered Fostex PM0.4 Misc use: Radio (with battery) iPad1 with iPengHD & SqueezePad (in storage SB3, reciever ,controller ) http://people.xiph.org/~xiphmont/demo/neil-young.html Mnyb's Profile: http://forums.slimdevices.com/member.php?userid=4143 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
magiccarpetride;698444 Wrote: > Gotcha. Thanks for the clarification. > > That being the case, would switching to the optical out be a cleaner > solution? It can be, but is highly system dependent - I've always suggested people try toslink (assuming they haven't removed the connector!) vs. spdif to see which they prefer in THEIR system. -- Phil Leigh You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal... Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables Stax4070+SRM7/II phones Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything. Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Bytec;698352 Wrote: > In case of digital transport Touch Toolbox mods will not have any > positive effects, but there is always a possibility to brake things and > ruin your Logitech SB user experience (turning off display, IR, etc). True. My iPhone ran out of battery juice, and because I had the SBT screen turned off, I was left with no other choice and was forced to grab my iPad in order to select what to play next. Terrible, terrible experience, I blame Soundcheck for tricking me into this. Is that what kids nowadays call 'the first world problem meme'? -- magiccarpetride magiccarpetride's Profile: http://forums.slimdevices.com/member.php?userid=37863 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Phil Leigh;698438 Wrote: > (in very simple terms) SPDIF is an analogue signal that represents the > data (the bitstream) in an encoded format including an embedded clock > signal. like all signals it can be clean or dirty (contain noise and > distortion). The dirtiier that signal is, the harder the spdif receiver > circuit/chip in the DAC has to work to accurately recover the clock (the > music data is not a problem). Gotcha. Thanks for the clarification. That being the case, would switching to the optical out be a cleaner solution? -- magiccarpetride magiccarpetride's Profile: http://forums.slimdevices.com/member.php?userid=37863 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
magiccarpetride;698392 Wrote: > Now I'm even more confused. What do you mean by 'cleaner signal'? > Cleaner than what? (in very simple terms) SPDIF is an analogue signal that represents the data (the bitstream) in an encoded format including an embedded clock signal. like all signals it can be clean or dirty (contain noise and distortion). The dirtiier that signal is, the harder the spdif receiver circuit/chip in the DAC has to work to accurately recover the clock (the music data is not a problem). -- Phil Leigh You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal... Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables Stax4070+SRM7/II phones Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything. Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Bytec;698352 Wrote: > In case of digital transport Touch Toolbox mods will not have any > positive effects, but there is always a possibility to brake things and > ruin your Logitech SB user experience (turning off display, IR, etc. Thank you, Bytec. Would this, then be the general consensus of other folks llike Phil, John Swenson, Mynb, etc.? I have what I believe would be considered to be modern dac, so what I'm hearing is that, at least in my case, clock and jitter issues really should not be of concern. -- rgro Rg System information Main: PS Audio Quintet > Vortexbox > Teddy Pardo PS, Touch (wired) > Toslink > Rega DAC > LFD LE IV Signature amp > VA Mozart Grands > REL Acoustics R305. Home Theatre: Duet/SBR (Wired) > Pioneer VSX 919 > Energy Take 5 Classic 5.1. SBS 7.7.2 r33908 on a Vortexbox Appliance, V 2.0, Touch: FW 7.7.1 r9558. Duet: FW 77. rgro's Profile: http://forums.slimdevices.com/member.php?userid=34348 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
Phil Leigh;698379 Wrote: > The Touch has a very different spdif circuit to the Duet and provides a > cleaner signal to a connected DAC. Now I'm even more confused. What do you mean by 'cleaner signal'? Cleaner than what? -- magiccarpetride magiccarpetride's Profile: http://forums.slimdevices.com/member.php?userid=37863 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
magiccarpetride;698373 Wrote: > I don't know the answer to your question. But I'd like to, if I may, add > another question, which may tie in nicely with what you're asking: > > Is there a rationally explainable reason why Squeezebox Duet, when > feeding digital signal via digital coax into my DAC, sounds noticeably > inferior when compared to Squeezebox Touch feeding the exact same > digital signal via digital coax into the same DAC? Or is it the case > here that even those 'noticeable differences' between the Duet and the > Touch are mere fabrications coming out of the cargo cultists' deluded > pathetic minds? > > And finally, is the above question providing sufficient grounds for > banning me from this forum? The Touch has a very different spdif circuit to the Duet and provides a cleaner signal to a connected DAC. This MAY make it easier for some DACs to perform better. There's nothing wrong with your question! -- Phil Leigh You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal... Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables Stax4070+SRM7/II phones Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything. Phil Leigh's Profile: http://forums.slimdevices.com/member.php?userid=85 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
rgro;698343 Wrote: > If people are merely using the Touch as a digital transport---connected > to a dac via coax or toslinkis there an explanation, other than the > much-discussed jitter question, for why there are such a number of > people here that seem to hear differences with the TT 3.0 modifications > and, more specifically, with changes to the priorities and buffer sizes. > Is there any reason---real or even theoretically---that, for lack of a > better word---something like noise would be transmitted to the dac > along with the bitstream and somehow find it's way from the Touch, > through the dac's processes, into to the dac's analog output, and on to > be manifested in audible changes? I don't know the answer to your question. But I'd like to, if I may, add another question, which may tie in nicely with what you're asking: Is there a rationally explainable reason why Squeezebox Duet, when feeding digital signal via digital coax into my DAC, sounds noticeably inferior when compared to Squeezebox Touch feeding the exact same digital signal via digital coax into the same DAC? Or is it the case here that even those 'noticeable differences' between the Duet and the Touch are mere fabrications coming out of the cargo cultists' deluded pathetic minds? And finally, is the above question providing sufficient grounds for banning me from this forum? -- magiccarpetride magiccarpetride's Profile: http://forums.slimdevices.com/member.php?userid=37863 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Digital transport/TT 3.0 explanation revisited, please
SPDIF interface transmits data AND clock. The question is what clock is used in the DAC when received data from SPDIF is playing. If DAC is in the slave mode and follows clock that is received via SPDIF then there is some space for discussion about jitter (source clock precision, cable quality, length ...). In the "old days" there were SPDIF clock injectors (re-clocking data with better clock) that you can connect between source and DAC and get very low jitter. Modern DACs has their own internal clocks and they re-clock SPDIF data by themselves and this is how jitter is reduced to very low levels. Re-clocking is just a buffering mechanism (few samples). So SPDIF is not that bad with modern DACs. :) I use SB Touch with TOSLINK. -- Bytec Bytec's Profile: http://forums.slimdevices.com/member.php?userid=17676 View this thread: http://forums.slimdevices.com/showthread.php?t=94352 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
