Re: [SlimDevices: Audiophiles] Audiophile Buzzwords, Fads, Crazes, Hypes and other Quantum Matters
ralphpnj wrote: > Nor did Mr. Swenson give any numbers or measurements regarding the types > of errors these green markers are supposed to correct. His response was > all very nicely written and with a seemingly clear logical and > scientific foundation but all beyond the prose and everything quickly > falls apart. > > So let's parse a little, shall we? > > > > Sound reasonable, right? So the electrical noise generated by the > tracking circuits is that noise audible and does this noise get > transmitted or carried through the DAC circuits and into the analog > output? And what would this "noise" sound like and how would one measure > it? Or is it all quantum mechanics and beyond my understanding? First off I never said it created errors (as in actual bit errors). What I found was noise on the power and ground lines, which was decreased when playing the treated CD. A good old fashioned spectrum analyzer showed it nicely, no special quantum micro analyzer needed. I didn't take any pictures or such I was just testing things out for my own interest, I wasn't trying to put together a "proof" for 20 years later. I can't redo any such test because the player I was using has long ceased to function and I don't have the one treated CD anymore. (or maybe I do, but I can't find it now). The sonic affect was small so I did not consider it worthwhile to spend the time to paint all of my CDs, so I dropped the whole thing. Several years later when that CD player died I switched to a DVD player, and noticed some significant differences in sound with the DVD player, which led to another round of testing players. In this case the culprit was vibrations exciting mechanical resonances of the magnetic suspension system of the DVD lens assembly. The servo loop has to work hard to keep these under control. There is a lot of EMI floating around the player from these tracking adjustments. This did wind up on the audio out. Again I didn't bother to take pictures of this, I wasn't trying to convince anybody or writes papers on this. Various vibration control schemes did make significant changes in this noise. I did try try the "green" CD with this and found there was some slight change in the noise, but it was difficult to distinguish due to the vibrationally induced tracking noise. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=101032 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Audiophile Buzzwords, Fads, Crazes, Hypes and other Quantum Matters
Archimago wrote: > Greening the edge of CD's was all the rage around the time I first got > some $$$ to buy audio stuff. > > Start date: ~1990 > > Reason for the buzz: Some vague theory about stray red laser light > bouncing around the polycarbonate. An article in ICE (International CD > Exchange) apparently. Green supposed to "absorb" it. > > Buzz kill: These things never die, right? Crazy SACD girl Theresa > Goodwin even wrote something around this in 2009 for "Positive > Feedback" (http://www.positive-feedback.com/Issue43/green_pen.htm). > > End result: CD rot? Delusional disorder? I was actually IR laser light, this is a very important distinction. CD players (NOT DVD players) used IR light to read the pits, the particular green ink had a strong absorption to the IR wavelength used by the laser, which did cut down on some leakage coming out of the edge of the CD. That DID allow some CD mechanisms to track better, cutting down on the electrical noise generated by the tracking circuits. Note that in order for this to work it had to be the right ink that had good absorption to IR, not just any old marker pen. Many tests were done with the wrong ink. Now fast forward to DVD players used to play CDs (which is now predominantly the case). The light is now red, not IR, the absorption characteristic of the ink is different to red than IR and the whole optical system and tracking mechanism is VERY different. I won't going into all the gory details but the upshot is that most DVD mechanisms have a very different response to "edge painting" than the original CD mechanisms did. Not that there is NO response, but it is different. The spectrum of the noise is quite a bit different due to the very different tracking systems. The upshot is that edge painting can still cut down on noise with DVD players, but how much is fairly variable depending on the specifics of the mechanism. In addition the noise that is being affected is quite different, which means it will most likely be detectable by different people in different ways than was the noise from the CD only player. So trying the same treated CD in a modern DVD based player is going to give very different results than the same CD in an older CD only player. Because of the different tracking mechanism for DVD players, there are others things that make a much bigger difference than edge painting, primarily vibrations in certain frequency ranges, if you want to tweak DVD players playing CDs, THIS is the area to (pardon the pun) focus on. I don't know whether Blue-Ray players have the same issues or not, I haven't studied them in this regard. (I just got my first Blue-Ray player a couple months ago) John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=101032 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Does an audiophile Squeezebox (digital out only) exist now?
Archimago wrote: > Fantastic info. Thanks John. > > BTW: Are you running room correction software on these? Curious what > plugin... I am not running room correction software, but I am running upsampling in Squeezelite using a filter spec that a friend and I developed for the DAC chip we are using in CSP. The room correction software I WANT to use is Dirac, but we have to figure out a way to get it into the flow on the wandboard. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=100850 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Does an audiophile Squeezebox (digital out only) exist now?
Archimago wrote: > John, which Wandboard are you guys basing the CSP design off of? Duo, > quad? > > Thanks... The software is written for both the dual and the quad. Everything runs fine on the dual, I've had a dual running LMS, playing 192k files, filtering in software, outputting to USB and it is using less than 10% of the resources to do that. What the quad gives you is a working SATA port. The only thing I can think of that might need the extra processing power is if you are running some compute intensive room correction software as a plugin to LMS. Since an off the shelf version of CSP will not be coming with a SATA drive, there is no reason to be using it and these will most likely come with a dual. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=100850 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Does an audiophile Squeezebox (digital out only) exist now?
I have written a series of articles on AudioStream that go into technical details on how differences can get through reclocking. For use with a USB input you might want to try the Community Squeeze project. ( www.CommunitySqueeze.com ) runs on a Wandboard (small Arm computer) which does VERY well at USB. I think it is somewhat better than a Touch. The Wandboard DOES have a TOSLINK out, but it is not particularly good, the S/PDIF on the Touch is better. We are also coming out with our own hardware player which has an exceptionally good sounding DAC, USB out that is very good and an exceptionally good coax S/PDIF. We are not ready with it yet, it should be coming out later this year (note I am NOT saying exactly HOW much later!) It uses the core guts of the wandboard for its processor so if you buy a wandboard now you can use it with the CSP (community Squeeze Player) John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=100850 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
marcoc1712 wrote: > John, > > > could I ask You witch USB DAC are you using? If is kind of a KIt or a > commercial product, could you point me to the source? > > is a kind of the one you'll be using in CSP1? > > thx a lot. > > p.s. > > I could not understand why if you upsample at i.e. 352.8 you still need > a software filter, is not assured this way that aliasing is far away > from the audible spectrum (reason why the hardware filter is disabled, I > suppose)? > > > Marco. The DAC I am using is a custom design I have done so you can't get it anywhere. Speaking of kits, I am designing a somewhat similar DAC for Bottlehead, it will be a kit, but all the hard work is already done on one PCB. It's not available yet, but hopefully will be out sometime early next year. Or just get a CSP! As to upsampling, all (usefull) upsampling has to use a filter, that is just part of the upsampling process. If you just resample the data without applying the filter you haven't actually done anything. This is best with a picture, I don't have one handy (I'm on vacation), maybe someone else can post a picture of this. For the wordy explanation, lets take the output of a basic DAC chip, without any digital or analog filtering you get a stairstep. Upsampling without the filter is just subdividing each of those stairsteps into smaller pieces, but keeping the same values. Lets say you have a sample at .5V level, 8X upsampling gives you 8 shorter samples all at the same .5V, you haven't changed anything. What you want is the upsampler to make a "guess" at filling in the intermediate values between the .5V sample and the next sample. The filter defines exactly how that is done. The human hearing system seems to be quite sensitive to exactly what that filter does, slight differences in that filter can make significant differences in what we hear. All modern DAC chips do this filtering process builtin to the chip, and every one of them does it in such a way that does not sound so good. This thread is all about doing that filtering externally in order to bypass the not so good version in the DAC chip. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Shilling for Dollars
Julf wrote: > So how do you measure what a SET does well? I was wondering if someone would ask that! The part that seems to make the difference is a monotonically decreasing harmonic structure. The third harmonic is a little less than the second, the forth is a little less than the third etc, all the way down to the noise floor. Today this is a very easy measurement, a good sound card and FFT software on a computer can do it. But up until 20 years or so ago it took an expensive spectrum analyzer to resolve the harmonics all the way down. This is VERY different than the "typical" solid state amp. With these the 2nd harmonic will be very low or almost non-existant, then the third will be much higher, then the forth very low, then the fifth a little less than the third. In this sequence the even harmonics are very low or almost absent and the odd harmonics are much greater. And somewhere around the 7th or 9th harmonic the odd ones start getting higher. This is a very different looking harmonic structure. You can't just stop at the 3rd or 4th harmonic, you have to go up to the 13th or 15th to really see what is happening. That means don't just look at 1KHz signals, look at some lower frequency ones as well. For some reason the human perceptual system likes the monotonically decreasing harmonic structure better, even though the total amount of distortion is greater. As I mentioned it IS possible to make both tube and solid state amps that have this harmonic structure and significantly lower overall distortion, but they have not "taken over the market", probably because most of these designs are very inefficient. For example my big tube amp gives 25W per channel, but takes 350 W. It wouldn't exactly pass modern efficiency standards! John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99360 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Shilling for Dollars
I'd like to reply to some comments in this thread, particularly to the ones about SETs and NOS DACs. These are not in the same category as electret cream, they are real life engineering tradeoffs. I'll give some examples of both. I'll start with the SET. I have built MANY amplifiers over the years, both solid state and tube, covering many different architectures. They all have different tradeoffs, doing one aspect better than others. Different people will prefer different implementations, they really like the optimization of one and are willing to live with it's deficiencies. The SET is an interesting case, from a sound perspective it does some things extrordinarily well that some people absolutely love, and does other things poorly, some people are willing to live with the deficiencies to to get what it does well. Other people are not. Part of the issue here is that what it does poorly is very easy to measure, and what it does well is very difficult to measure (not impossible, just difficult). It took a long time for the engineers to figure out how to measure what it was doing well. Now that what an SET does well is mostly known some designers have been able to come up with solid state circuits that sound similar, but they still have a some of the same deficiencies. In my case my current main amp is an SEP, Single Ended Pentode, it has a similar good characteristics to an SET, but it also has vastly lower distortion, less than 0.1% over the whole power range (up to 25 watts) and frequency range. This has been possible by eventually understanding what it is about SETs that make them sound good, and then figuring out how to keep that while improving other aspects. This turns out not to be easy, improving one aspecct very often makes the other worse. It's good old fashioned engineering tradeoffs. Unfortunately it's not cheap, there are about $3000 worth of parts in this thing and it weighs 150 pounds, but boy does it sound good when driving the speakers it was designed to drive. With $3000 worth of parts it would be very expensive if it were commercially produced, and that would NOT be exhorbitant profits, it just plain costs a lot of money to get that level of performance. At least for now. Over time designers might figure out how to do it for a lot less money, but that hasn't happened yet. On the DAC side, many years ago I (and others) noticed an interesting fact, that when you bypassed the internal digital filter in most DAC chips things sounded better in some ways, and also worse in others. Without the filter the sound was more musical, more alive, more realistic, BUT it also sounded "dirtier". Some people are willing to live with the "dirtier" in order to get the "more alive" sound. The dirtier of course comes from the aliases, but nobody yet knows what the digital filters are doing that squashes the "alivenes". Again it's a case of nobody knows how to measure "the goodness" but it is very easy to measure the "badness". I have spent the last 7 years trying to find out what it is that the digital filters are doing that causes the problem. My currrent understanding is that it is NOT the fact that it is a digitalfilter in general that is the problem, but the implementation that is used in ALL DAC chips that have a builtin filter. In order to get extremely good numbers in the spec sheets for certain parameters the designers have taken to using complex filters in the chips. Getting these spec sheet numbers with the traditional mathemetical function you read about in DSP textbooks takes a lot of processing power which would significantly increase the price of the chips. So they have come up with DSP "tricks" to get those numbers, but now the filter function is much more complex, and this "complexity" seems somehow responsible for the sonic degradation. At this point I have no idea WHY this is, just that it is. I have now built several DACs using external digital filters and DAC chips that either don't have digital filters or it can be turned off. When I use the basic simple function and give it enough processing power (either in software on a computer or in an FPGA) the results are amazing, you get the "aliveness" of the NOS DAC, but all the dirt is gone, the result is stunning. Again the NOS DAC is a tradeoff, people willing to live with the negative aspect to get the positive aspect. As we start to understand what is going on it starts to get possible to do away with the tradeoff and get both aspects done well. Neither the SET nor the NOS DAC are an attempt to bilk the public for monitary gain, they are legitimate engineering tradeoffs that some people are willing to take. Hopefully as we understand what is actually going on we will be able to do away with the tradeoffs, but that is not fully there yet. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?useri
Re: [SlimDevices: Audiophiles] Audio-dg (AGD) DAC + SBT + EDO
The Fez wrote: > Hi Guys -- Had a SBT for years now. Really nice product. Just wondering > if anyone else has a Audio-gd DAC connected via USB. I have the EDO > installed, but the SBT doesn't recognise the DAC without a USB hub? I > was of the impression the AGD DAC'S can connect direct to the Touch. I'm > using the USB cable as supplied by AGD that came with the DAC.. > > Thanks for any thoughts... There could be two issues, the USB power management in the Touch strictly enforces the USB power management protocols, max current of 500mA and device has to run under 100mA until it requests higher current. If a device doesn't meet this it won't work. The other is a restriction in the USB hardware that prevents it from properly working with a UAC1 asynchronous DAC. (UAC1 DACs only go up to 96KHz sample rate) Strangely enough the high speed mode of UAC2 works fine. (UAC2 goes up to at least 192) Using a hub with a UAC1 USB interface gets around the hardware linitation of the Touch. In addition if the DAC pulls too much current a powered hub can fix the situation. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99599 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
edwardian wrote: > Cool. Thanks for the info. Do those percentages include FLAC decoding, > or are you feeding it PCM? I'm usually sending flac over the network these days since I don't have a new enough server to handle 176 and 192 pcm. The above numbers were for sending flac at 44.1 over ethernet cable. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
I started this because I was using the upsampling built into Squeezelite(libsoxr), not using SoX in LMS so I can't offer any clues as to how to get that to work properly. The Squeezelite resampling option is not the same as in SoX the program arguments, although the underlying code is the same. The one I'm using right now is mI:::28, which is medium quality, intermediate phase and 28 bit depth. I'm not quite sure exactly how this translates to SoX arguments. The upsampling is done in Squeezelite on the Wanboard and then sent to my own USB DAC, which uses a standard XMOS UAC2 interface, I2S is run through isolators, on isolated side are low jitter clocks, reclocker and the DAC chip is PCM5142 with very low noise regulators on all of this. The low jitter clock is sent back through an isolator to the XMOS interface. The PCM5142 has several builtin filters, some of which are better than others. If you feed it 352.8/384 it turns off the internal filters. So by using resampling in Squeezelite to 352/384 I can bypass the implementation in the chip and just the software. Even if upsampling to a lower rate it can still be advantageous, upsampling to 88.2 with the above paramters is a significant improvement. Going to 176.4 is even better but the best canbe achieved by going high enough that the built in filters are completely bypassed. In all these cases the network traffic stays the same, but the USB data rate goes up since the upsampling is being done in squeezelite. I have not had time to go into depth trying all kinds of different parameters, I spent a few nights trying different things and came up with the above. I've been listening to it for some time now and am still enthralled with what it is doing. BTW the load on the Wandboard processor is about 8% when using this setting. When using the default 20 bit setting it is about 4% and when not doing any upsampling its about 2%. Klaus, to your statement that upsampling should not be necessary, the answer is of course YES. The issue is that as far as I can tell all DAC chips with builtin filters are compromised sonically, the upsampling is an attempt to bypass these filters with an external filter that is more sonically "transparent". So yep it IS a band-aid, but one that is currently necessary for most DACs. If the internal filters are not disabled, exactly how the internal filters interact with the external filter are going to be very DAC specific. In the case of the chip I'm using the filters get simpler as the sample rate goes up so even though they are still there if you upsample to an intermediate rate, the total result sounds better than the builtin filter going from 44.1. The best sounding parameters for going all the wayto 352.8 and 88.2 will almost certainly be different, but I have not explored it yet. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
Hi Pippin, to your underlying question: why do the filtering externally rather than in the DAC chip? The answer is "I don't know". When I bypass the internal filter and do it externally using a basic simple filter it sounds much better. And it's not just me. I've done this in blind fashion with a number of people and they all came to the same conclusion, the external filter sounded much better. Looking at the spec sheets for the DAC chips you can tell that they are using cascaded filters. When I use an external FPGA to implement a filter and program it to be cascaded multiple filters, I hear a similar sound, when I implement a simple filter (no cascading) I get the better sound. The conclusion seems to be that it is somehow the cascading of filters that causes the problem. Again no clue at all why this is so, what the mechanism is etc. They ARE different filter functions, they do output different bits, but they are very close to each other. Whatever it is, its not a big difference in the waveform. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
pippin wrote: > But then you've got all these high frequencies still in there on the > analog side. Isn't the biggest problem with the high frequencies that > power amps (especially ones filtering through intermediate frequencies > like "Class D" amps) tend to create artifacts if you have signal in > multiples of the audible range? > > > > What kind of filters do they use? I do know that kind of stuff for > analog filters where both the reason for their use and the negative > impact are quite obvious. > > But then... like for the windowed sinc filter you mentioned even digital > filters are not perfectly discrete if you need finite impulse response > and linear phase so it makes sense that they will also flatten the > frequencies within your filter/below your cutoff frequency. > But isn't that the same for your interpolation filter in sox? Of course, > the interpolation filter doesn't window so it keeps all the harmonics in > there (and creates some arbitrary new ones) but why is that better than > leaving the simple harmonics of the 44.1 kHz samples in? > What kind of filters does sox use? Hi Pippin, very good questions BTW. It turns out that with high bit depth (24) and high sample rates relative to the bandwidth of the signal the aliases are very low in amplitude. If this was not true a digtal filter would not work for this purpose at all. The amplitude of the alias is determined by the "height" of the "stair step" in the signal. At 16/44.1 those stair steps can get pretty high. Very easy to see in a scope from a NOS DAC. The filter adds steps in between the originals steps, because of the increased samples and the increased bit depth the height of the steps is much less, thus producing much lower amplitude aliases. Another way to think of it is because the bandwidth of the signal has not changed, the maximum amount of change for a given sample is also less, thus the maximum possible height for a stair step is also decreased. The reason to do the digital filtering at all is that for 16/44.1 the stair steps are large, thus producing high amplitude aliases, and those aliases are close to the audio band. There are two methods by which an alias can cause an image in the audio band: intermodulate with signal, or intermodulate with another alias. It turns out that the aliases produced by a given signal are lower in amplitude than the original signal, thus on average the intermodulation of aliases with signals is going to produce higher amplitude images in the audio band that alises intermodulating with other aliases. For example a 25KHz alias intermodulating with a 15KHz signal (thus producing a 10KHz image) will be greater in amplitude than the same 25KHz alias intermodulating with a 35KHz alias. They both exist, but the intermodulation with the signal will usually produce a higher amplitude image. Now let's look at what happens with the signal upsampled to 176.4, the aliases start at 88.2. So lets look at what happens to a 90KHz alias. First that alias is going to be much lower amplitude in the first place (see the above) and the lowest frequency image it can produce from intermodulating with signal is 70KHz, way above the audio band. Now you certainly can get images in the audio band from aliases intermodulating with other aliases, BUT because the aliases are so much lower in amplitude, the image between two is going to be very low in amplitude. The net result is that the images in the audio band are drastically reduced (not completely eliminated) by the upsampling and filtering. A high sample rate high bit depth original recording already has these advantages built in. Now the fun part is to do this filtering in such a way that it preserves the flavor of the original recording. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
pippin wrote: > So what you are saying is that if you use these kind of DACs with "real" > (that is: unfiltered) HD material like actual HD recordings you will > have all the aliasing down there in the audible band? > This means you have to actually run "real" HD material through a filter > to get a useful reproduction with this kind of DAC at all? > > Oh, and I did not fully understand what you mean with a "simple" vs. a > "complex" filter. Interpolation will certainly lead to rather complex > harmonics so from a purely signal theory POV I would find that way more > complex than what you'd usually do in an anti-aliasing filter. Let me see if I understand your first question. As long as the highest frequency of the audio data is small relative to the sample rate (say 40KHz maximum signal frequency for a 192 sample rate) you do not need much if any filtering. There are going to be very little aliasing and its going to be above the audio range. The big problem comes with 44.1 where the audio data is right next to the aliases. By a complex filter I mean multiple filters cascaded together (especially different types of filters) rather than one filter (even if it has a large number of taps). All of the DAC chips with filters have at least three (some 4) different digital filters cascaded together. They do this because the individual filters can be a fairly small number of taps. To implement the same amount of alias suppression with a simple single filter would take a lot more hardware. Again I have no idea why the complex filters cause sonic issues, but it does seem like they do. Every time I have replaced a complex filter with a simple filter the resulting music sounds much better. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Upsampling Impressions
As promised here is post number two on external upsampling. This post is going to focus on the upsampling capability recently included in Squeezelite which is based on the SoX resampling code. These filters can also be used offline with SoX, but the command line arguments for SoX are very different, even if they are implementing the same thing. I'm just going to go over the options for Squeezelite, if you want to try SoX by itself, read the documentation for the filter arguments. All of these adjustments you can make here are essentially "tweaks", they are fine tuning the sound. The biggest improvement by far is just geting out from under the complex filters in the DAC chip. Before getting into details I want to talk about sample rate. The filters in many DAC chips get simpler the higher up the sample rate is. For example the chip I'm using in the CSP player has a very simple filter at 176.4/192 and NO filter at 352.8/384. So in general you want to upsample to the highest sample rate you can get to work reliably, this will probably give you the best advantage from the upsampling. But if at all possible keep it to an integer upsampling, for example upsample 44.1 to 176.4 not 192. If your DAC doesn't do 176.4 but does do 192 and 88.2, you have a choice, upsample to 88.2 or 192, which is best is going to be system dependant, try both and see which you prefer.. The current implementation in Squeezelite does upsampling to the highest interger rate your DAC cupports. Thus if your DAC's maximum rate is 192, it will upsample to 44.1 to 176.4. If your DAC does not support 176.4, you can use the -r option (or max sample rate in the gui) to set the max rate to 96, then squeezlite will upsample to 88.2. Even upsampling to 88.2 will usually make a significant improvement. There has been talk about giving the upsampling more flexibility so you could choose 192 in this case. The SoX resampling gives you several different parameters to adjust, choosing the right one can be a daunting task so they came up with what are usually called "recipes", predefined combinations of parameters. These are good places to start your exploration. When you give Squeezelite a set of paramters you first give a recipe, then you can add modifiers that over ride the parameters in the recipe. The upsampling "argumnt" consists of several fields separated by colons (:). If there is nothing specified for a particular field the default for the chosen recipe is used. There are up to 7 fields. If you are not specifying any of the later fields you do not need to include all the colons. Some examples: abc first field only, all others default abc:xyz first and second field, others default abc:::def first and fourth fields, others default :xyz:::def default first field, fourth field and others default The first field is the basic recipe. It is special, it is made up of three subfields, each specified by a letter. The first letter is the "quality", it can be one of these characters: v h m l qNote the fourth character is the lower case "ell". These stand for "very high", "high", "medium", "low" and "q" A word on these, the differences are not necessarily related to how they sound, they refer to how far the aliases are attenuated (remember the engineering community regards the alias attenuation as the primary figure of merit for a filter). These primarily relate to how long the filters are. All these filters work by performing calculations on a series of samples in the file, the longer filters use more samples, the shorter filters use fewer samples. The greater the alias attenuation the longer the filter has to be. This CAN affect what you hear. Do not just blindly use the "very high" setting, it may not be the best sounding. The second leter is the "phase". You have a choice of: L M I (that is upper case "eye"). L is linear phase, M is minimum phase, I is intermediate phase. There are LOTS of descriptions of what these are out there on the web so I won't go into detail. The third letter is: s It is either included or not. This specifies a very steep filter curve. If the s is not included a slightly shallower filter is used. The deault is hL (no s). If a letter is left off the default is used. Thus 'v' is the same as 'vL'. 's' is the same as 'hLs'. 'Ms' is the same as 'hMs'. All that just for the first field! The second field contains flags. Nobody really knows what these do so its safest to leave this field blank. The third field is attenuation. It is possible that the upsampling can clip on signals that are at or very close to maximum, this attenuates the input so this clipping doesn't happen. The argument is a number, which is the attenuation in db. Thus '6' is 6db of attenuation. The defualt is '1', which is probably good most of the time. The fourth field is "precision", it is the number of bits used in calculations. We know that 20 and 28 are sometimes used depending on o
Re: [SlimDevices: Audiophiles] Upsampling Impressions
Wombat wrote: > Many 44.1 material once was at a higher samplerate and there was already > once a choice what filter to apply. How can you know that a chosen > filter doesn´t only do better because it fits more to the filter that > was applied before? I did read to much nonsense about the sound of magic > filters over the years. The listening tests with different filters were done with a wide variety of music from different labels. The presumption is that not all of these recordings were made with the same filters. We did not find an effect where the sound of a simple or complex filter was influenced by the recording. The simple filter sounded better on everything. I was a bit surprised by this, the effects of the simple filter are things like increase audibility of the recording acoustic environment. I expected that close miced multi trac, overdubbed recordings would show little difference, but this was not the case. They too seemed much fuller, more open, more alive. So while there may very well be an interaction with filters used in the production of the recording, these interactions seem to be much smaller than the difference between complex and simple filters. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
[SlimDevices: Audiophiles] Upsampling Impressions
In recent versions of CSOS Triode and JackOfAll have added upsampling capabilities to Squeezelite (at my request). I wanted to start a thread here for discussing sonic differences people hear between no upsampling and different upsampling parameters. I'm also going to give some history of my involvement with DAC interpolation filters and why I wanted to do this in the first place. First off this is NOT about bit perfectness, upsampling changes bits! It is supposed to! There is a real measurable difference in the waveforms, this is about how you think those changes affect the sound. First a quick refresher on what this is all about. When a DAC outputs a signal there are "aliases" above the half Nyquist frequency, these aliases can inter-modulate with the audio signals to produce spurious signals in the audio band which ARE audible. The reconstruction filter is there to get rid of these aliases. With redbook source this is difficult, the half Nyquist freequency is just barely above the audio band. In early days this was done with anolog filters, but they really messed up the sound. Then chip makers came up with using digital filters to do this. These are called "interpolation" filters because they add samples in between the original samples. Every DAC chip manufacturer has used a different approach to these filters. The metric of filter quality has been "how much does it attenuate the aliases". At first they could get -60db, then it was -90db, then -120db, these days there is big contest between DAC manufacturers to see how far down below -130 they can get. But do you really need to get down that far? The further down you go the more computations need to take place. It takes more silicon and faster clock speeds to do it. Early on in the process the chip manufacturers realized that implementing the filters using straight forward simple filters such as the infamous "windowed sinc" (I'm NOT going to go into details on that here, look it up if you want the details) would take far too much silicon real estate and drive up the costs of their chips. So some massive investigations started looking into how to get these extremely high alias attenuations without using a pile of silicon. The result has been some very complex digital filters where multiple filter types are cascaded together. These do their job, they attenuate the aliases but they seem to have some side affects. I first noticed this when I started building my own DACs many years ago, NOS DACs were coming into vouge so I decided to make my own. I was startled. They sounded way better in many ways than the traditional ones with the builtin digital filters, BUT they had the aliases which caused other problems, I like to call this "dirty sound". So there was a choice, clean, flat univolved sound, or dirty, alive and exciting sound. Many people that liked the the NOS sound were convinced that digital filters were evil and should be expunged from all things audiophile. This then started one of my infamous journeys of exploration. I'm not going to go into all the details here, but the upshot is that it's NOT digital filers per se that are bad, it is those complex digital filters used in the DAC chips that seem to be the culprit. If the filter function is implemented by a simple filter (such as the infamous windowed sinc) you can get the best of both worlds. Note that I really don't have any idea exactly what it is about these complex digital filters that humans don't like. I have built a DAC with DAC chips that don't include digital filters (1704s) and have used an FPGA to implement simple and complex filters that give roughly the same filter response, and the simple filter sound vastly better. And I really do mean vastly, it is NOT something barely distinguishable. Many people are not going to believe this since they have spent their entire digital audio life listening to music through these complex filters. And NOS DACs don't really count since you are trading off one set of sonic issues for another. Even the DAC chips with several filter curves don't really help to hear this since all their filters are done with the complex filters. So what does upsampling have to do with all this? There are quite a few programs out there that can implement the interpolation filter using simple filters, you can either run these real time or upsample the source files so you don't need to run anything at play time. If you have a NOS DAC you can use these to really hear what a simple filter can do, you don't have to choose between the above listed issues. But even if you don't have a NOS DAC, properly done external upsampling can make a big difference. Many DACs use simpler filters (or NONE!) for the higher sample rates so using external upsampling can frequently give you better sound than filters in the chip. Note this is NOT about the asynchronous sample rate converter (ASRC) chips used in some boxes, these do their job using multiple complex filters that are ev
Re: [SlimDevices: Audiophiles] The Next Frontier
There actually IS a technical reason why DSD may sound better than PCM, even though it is technically inferior to hi-res PCM: lack of digital filters. I have done extensive study on this and have come to the conclusion that ALL DAC chips which have internal digital filters have not correctly implemented those filters. Making the filter mathematically correct talkes a fair amount of compute horsepower, particularly for 44.1. None of the chip makers want to include that cost in their chips, so they take shortcuts in the implementations. They are all compromised to some degree. I have tested this with several chips that have the ability to turn off their internal filter and use an external filter. I implemented my own filter in an FPGA, doing it right with proper bit dpeths and no compromises. The results sound quite a bit better than the internal filters. I have had several people do blind tests on this and they all can hear the difference. In regards to DSD, a simple implementation doesn't use a digital filter, just a simple analog filter. All the complex filtering has been done in software in the DSD mastering where the filters can be a much closer match to being mathematically correct. Of course all this is highly implementation dependant. So DSD may actually sound better in some circumstances, but IMO the correct thing to do is implement PCM DACs correctly rather than trying to productize DSD at the consumer level. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=97545 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] This is why I call them clowns
jh901 wrote: > The Transporter's clock is vastly inferior to those found in today's > best DACs. Vastly. > > Anyhow, is asynchronous USB superior to S/PDIF? In any way? I've > pointed out that slaving the Transporter, for example, to a master is > not possible. If the Transporter had asynchronous outputs then it would > be desirable to use the USB inputs on many uber-DACs on the market > today. Mr. Harley is a much better source of advice than anyone on this > forum, btw. He literally wrote the book on hi-end audio. > > http://www.amazon.com/Complete-Guide-High-End-Acoustic-Engineering/dp/0978649311/ref=sr_1_1?s=books&ie=UTF8&qid=1354912394&sr=1-1&keywords=robert+harley But you CAN slave a Transporter to a DAC, it has a wordclock input which can be used to slave its S/PDIF output to a wordclock from a DAC. You just need a DAC with a very good internal clock which outputs a wordclock and reclocks the data from the S/PDIF receiver with its internal clock. With such a setup the clock in the Transporter is ignored, the clock in the DAC is in complete control. It doesn't use USB, but the data from the Transporter is still slaved to the DAC. Admittedly there are not a lot of DACs that can do this, but that doesn't mean the Transporter is incapable of it. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=97489 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Support for 24bit/92/192khz...?
SB3 supports 44.1 and 48. Natively the Touch supports 44.1, 48, 88.2 and 96. There is an applet for the Touch called EDO (Enhanced Digital Output) which will pass 176.4 and 192 on the digital outs, AND allows you to connect a USB DAC to the Touch and will go up to 192 if the DAC supports that over USB. (There are quite a few DACs that support 192 over S/PDIF coax but not over USB) If you try and play a high sample rate file to a player which does not support that rate, the server will resample down to a rate the player does support. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=96438 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Clock in for Transporter
There is only one situation where the word clock in is useful: you have a DAC with a very low jitter internal clock AND it can output a word clock derrived from that internal clock. You connect one of the digital outs from the TP to the DAC and the word clock from the DAC to the TP. This "slaves" the TP to the clock in the DAC. Note that there are problems with this setup, the TP runs at the sample rate of the DAC, so you have to choose the sample rate at the DAC, not the source of the music. For example if your DAC is sending out a 44.1KHz word clock the TP can ONLY play music at 44.1 (because it is slaved to the DAC). If you want to play a 96 KHz file you must either change the frequency of the wordclock coming out of the DAC, or resample the 96 file in the server. You have to be careful with undertsanding what is meant by a word clock and a master clock. A word clock is a stream of pulses at the frequency of the sample rate, 44.1KHz, 48KHz, 96KHz etc. The term "master clock" has two different meanings: 1) a word clock that is connected to multiple devices, thus synching them all together, thus it is the "master". 2) a high frequency clock running at a much higher frequency than the sample rate, common ones are 11.2896MHz, 12.288MHz, 22.5792MHz, 24.576MHz. These numbers sound weird but they are multiples of the common sample rates, 11.2896MHz is 256 X 44.1KHz. The TP only accepts a wordclock, NOT a high frequency master clock. Thus if the DAC just outputs a high frequency master clock you cannot use it with the TP. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=96421 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Pre-ringing, minimal phase filters (and others), and apodizing...
First on the term "apodizing": different people have radically different meanings for the term, so it's very difficult to figure out what any body means. If I remember correctly the original use for the term in regards to audio digital filtering was about a non-brickwall filter applied to higher sample rate signals. Somewhere along the line it got confused with minimum phase filters (I think one box had both apodizing and minimum phase so the press assumed they were the same thing). I have done a fairly extensive set of listening tests both by myself and with others concerning digital filters, both hardware implementations and software implementations. I don't have time to go over all the details right now. The upshot is that the digital filters built into almost all DAC chips are flawed in one way or another. Most of the "digital filters sound bad" sentiment is actually not about pre-ringing etc, the culprit seems to be that the hardware implementations take shortcuts in their implementations in order to both produce good numbers in the spec sheet and meet cost requirements of the chips. Pretty much everybody is assuming these filters are implementing a traditional Sinc function, but in reality they are not. The designers have come up with interesting implementations that that give good numbers but don't take as much chip resources. These shortcuts are what I think are the real issue, not pre-ringing etc. Some results that tend to back this up are: Take a really good NOS DAC feeding something like a 1704, with async USB front end, very low jitter clocks. Listen to it with a DF1704, NOS, digital filter implemented in an FPGA with just basic Sinc response, and software upsampling to 192 again with just basic Sinc. With the DF1704 it sounds flat and uninvolving. Played NOS it sounds way more alive, interesting and musical, BUT it sounds "dirty", you can hear the aliasing going on. With both the FPGA filter and the software filter it keeps the aliveness etc., BUT the "dirtiness" goes away, it's by far the best sounding. This is NOT some just barely noticeable affect, it's actually quite startling when you first hear it. Several DAC chips that contain internal digital filters allow you to bypass them and use your own external filter. I have done the same test with several of these with the same result, internal filter sounds flat, NOS sounds much better but dirty, FPGA or software Sinc sounds wonderful. There is a group of people that are extoling the vitues of software upsampling, but unfortunately most of these people are then feeding this stream into chips that have builtin digital filters implemented with these shortcuts. This severly limits the efectiveness of the software digital filters. Yes some chips support different varients of filters, but they are all implemented with these shortcuts so it makes the inherant differences very difficult to discern. There ARE differences between filters, minimum phase, Sinc etc, but these differences are tiny compared to just getting rid of the hardware filters in the first place. There ARE a few DAC boxes out there that do use FPGA filters instead of the internal filters, but they are few and far between, and usually pretty expensive. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=96098 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Transporter External word clock
When you feed an external wordclock into a Transporter it has to use a PLL to generate the high frequency "master clock" used inside the TP. This PLL is going to have much greater jitter than the crystal oscillators already in the TP. The ONLY reason to use a wordclock with a TP is if you already have a DAC which has very good internal oscillators AND both reclocks the incomming data with those clocks, and outputs a wordclock based on those oscillators. For this and ONLY this case is the external wordclock input useful. It allows the TP to synchronize it's output to exactly match the clock already in the DAC. The data from the TP gets reclocked by the very low jitter clock in the DAC so the extra jitter added by the wordclock PLL in the TP is irrelevant. And for reference the wordclock has to change for every different sample rate, so using an external wordclock can get very tricky since you have to tell the DAC what the sample rate is so it can send the appropriate wordclock to the TP, but the TP runs at whatever rate it gets from the wordclock, sort of a chicken and egg problem. So using an external wordclock JUST into the TP is either going to do nothing or make things worse, so I don't see any point to even try it. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95978 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Dac not compatable with SBT?
What is the firmware version on your Touch? Older firmware had a bug that would rapidly switch to 44.1 on any sample rate change, such as going from 88.2 to 96 it would switch to 44.1 for a few miliseconds which could cause some DACs fits. But you are having problems with 96 to 44.1 so that is probably not the issue. Let me know what firmware you are using and I'll stick a logic analyzer on the Touch and see what the coax output is doing when changing sample rates. Who knows, it might be doing something strange. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94260 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Fidelizer & SBT.. Why should it work?
Note that I specifically said that the tests I had run were done in the same room as the stereo system, usually 6-8 feet away, NOT several rooms away. These tests happened about 5 years ago when I took my SB3 over to a friends house to show it off, how nice it was to use and how good the sound was from such an inexpensive box, and that it could work well with his expensive DAC. I was running this off my laptop with the server on the laptop and using the web interface to control things. After listening for a while he said that he could hear a change in the sound when I was actively doing things with the computer (searching for the next song to play etc). Well that surprised me I had never heard that before. We then started doing some tests to see if it was related to the fact that it was the computer with the server or if it was just the laptop in general. We loaded the server on his Mac mini and sure enough, working with my laptop caused a change in the sound, even if it had nothing to do with the sound flow (server on another computer, playback from SB3). We tried this with one of his laptops and got similar results. Doing things with the Mac mini didn't cause any change, but it was not in the room with the stereo. This got me thinking about how computers can affect things. I started running tests in my system, and other peoples systems. We did find that there was significant difference in sensitivities, some systems more than others. At this point I was not interested in a more rigorous testing to try and figure out correlations (what was it about a system that made it more sensitive). The thing about the mouse was not a wireless mouse, just the increase in activity in the computer when moving the mouse. Interestingly the couple places I have heard this have NOT been on "audiophile" systems, but at peoples houses with "normal" systems. I was at one ladies house, she had the stereo on but the song had stopped playing, she was working at her laptop, we could definitely hear a grinding sound coming from the speakers, changing in concert with the movements of the mouse. I have heard something similar to this at another friend's house, again not an audiophile system. Since this has little to do with the thread topic I'll shut up now. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95644 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Fidelizer & SBT.. Why should it work?
chill wrote: > John > > I can't tell if you're just being mischievous, but if not, are you > seriously suggesting that the mere presence of a powered-up computer in > the same house as the hifi will have an audible effect? That's a bit of > a blow for the whole computer-based audio industry. But moreover, and > returning to the OPs point I suppose, are you suggesting that the > changes in EMI and mains noise between a computer that's running > Fidelizer and one that's not will be audible? > > I realise that your post is describing a theoretical possibility only > (hence the capitalised 'COULD'), but what is the likelihood of these > effects being audible in the real world? How bad would your components > have to be for such minuscule things to have an impact? IMO, the > plug-pull test is already convincing enough. I'm being serious. A computer sitting on the same shelf as your stereo system can have a significant affect on said stereo system from airborne EMI and noise on the power line. If it's three rooms away the effects will be much less. This thread never specified any geometrical arrangement of the components. I know several people who have tried to use their laptop to control an SB, the laptop was across the room from the stereo, they could hear noise on the stero system when they moved the mouse on the laptop. What is going on inside the computer can have an affect on the sound without changing bits. I did some tests on this quite a few years ago testing a bunch of desk tops, laptops, small things such as Mac mini's, embedded devices like FitPCs etc. I did this with my stereo system, and with a few friend's systems. In all these tests it was a computer in the listening room, but NOT right next to the stereo system, usually across the room. Note that the computers being testsed had nothing to do with the audio. The stereo was being fed by an SB conected by wire to a server a long ways away. The computers under test were just doing things like web broswsing etc. The worst by FAR were the laptops, almost everyone was audible in some way, either directly emitting sound (through the stereo) or changing audio that was playing. Desk tops fared quite a bit better, either not audible at all or not as much affect. Small general purpose computers such as Mac minis did a little better, but still could be heard under some situations, embedded devices such as a FitPC were inaudible no matter what we had them doing. The method of "contamination" from the laptops seemed to EMI, they did just as bad when run off batteries. We tried wrapping one in aluminum foil (kind of hard to use in that state!) and it's affect went way down. Whether the screen was up or down also had a significant affect. So yes computers CAN affect sound quality, and something which is changing the underlying behavior of said computer could very well change it's impact on sound quality. I have no knowledge about the program in question here so I'm not making any comment about it, I was primarily refering to the use of the "pull the ethernet cable" test as being definitive, if the server computer is still running there is the possibility that it can still be affecting the sound quality through means other than direct connection to the SB. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95644 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Transporter as server (digital outs)- a few basic questions
The Transporter will not do 176 or 192 on it's outputs. If you play such a file the server will down sample 176.4 to 88.2 and 192 to 96. The Touch will output 176.4 and 192 on it's digital outs if you use the EDO applet from Triode. EDO also allows you to hook up a USB DAC, which can also handle 192 if you have a USB DAC that can support UAC2. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95476 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Impedance Matching Squeezeboxen to DACs
Jeff Flowerday wrote: > I'd be interested in John's opinion on these transformer based > converters. These transformers can be quite decent. This is how I would do an AES/EBU output. On anything I do with an S/PDIF output I provide a 75ohm BNC, if the user wants to go to an AES/EBU input they can use one of these transformers. Whether it's going to be the best way to get into a DAC is going to be DAC specific. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95614 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
Turnandcough wrote: > Sorry if this has been covered but I couldn't find anything. > > 24/96 is more than enough for my ears so I'm not really interested in > the 24/192 aspect here. > > 1) Can I use the Touch USB output with an Audiophilleo? Do I need to > install EDO? > 2) Theoretically(or empirically) would this setup be an improvement on a > straight coax out to coax in on my DAC > > Thank you I think you can use an Audiopilleo with the Touch. I don't have one so I can't be sure. It sounds like it is uac2 compatible so it should work with the Touch. You DO need EDO in order to use a USB audio device with the Touch. Will it be better than the coax out from the Touch? Who knows, that is VERY dependent on the DAC. For some DACs it won't make a difference, for some it will be better and there will probably be some for which it sounds worse. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Fidelizer & SBT.. Why should it work?
chill wrote: > You make a good point. I stand corrected. > > So do you think the OP is interested in the THE simple test that will > demonstrate that SuperQ's suggested mechanism is the correct one? You > know, the 'just pull the plug' test. But that test is not necessarily defintive, it only tells you if the processing on the Touch is the issue. There are other possible paths from server to ears such as EMI radiated from server, noise injected on power mains etc. Something which is changing the whole operating environment of the server COULD be changing something which gets transfered through one of these indirect paths, even when audio data is not actively being transmitted to the Touch. Now if you unplugged the the Touch from the ethernet AND unplugged the power form the server at the same time, THEN you would have a more convincing test. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95644 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Impedance Matching Squeezeboxen to DACs
konut wrote: > The prevailing wisdom asserts that no RCA termination will give a true > 75 ohm result. Then again if the source and DAC were designed to be > impedance optimized with the RCA jacks then this wouldn't really matter. Unfortunately impedance was not a design characteristic of the RCA plug, in actuality they vary all over the place, from something like 25 to 45 ohms or so. They were designed as cheap audio plugs with no thought that they needed tight impedance characteristics for RF frequencies. BNCs OTOH WERE designed as RF connectors so do include impedance as a design spec. Speaking of impedance and BNCs, they come in two flavors, 50 ohm and 75 ohm. They are interchangable from a mechanical perspective. The 50 was the original version and existed for a long time in the marketplace. Then someone realized that you could take off some of the solid insullation and make it air insullated and presto, its 75 ohms, perfect for 75 ohm systems such as S/PDIF. Unfortunately MANY companies that sell 75 ohm cables with BNC connectors use 75 ohm cable but use 50 ohm connectors! I did a very unscientific sampling of cables available on the internet and found that only 30% used real 75 ohm connectors. So even if you get equipment with BNCs make SURE it uses a 75 ohm jack and that you get cables with 75 ohm plugs. And it's not just "audiofile" companies that get this wrong, I bought some cables from several cable places that specialize in selling cables to pro broadcast customers and their 75 ohm cables had 50 ohm connectors on them! I'm completely at a loss to understand how this happens, that a cable company doesn't know the difference and the "professional" customers don't either. It's interesting that Blue Jeans gets it right, they are inexpensive yet extremely high quality cables done right. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95614 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Impedance Matching Squeezeboxen to DACs
I knew someone was going to ask this! The above post was actually an oversimplification, in actuality the impedance can vary with frequency. Its similar to saying a speaker is 8 ohms, very rarely is it 8 ohms at all frequencies in the audio range. Some reviews will publish the full graph of speaker impedance VS frequency, but how does a consumer use this? How can a consumer figure out whether this speaker is going to sound better than another using a graph for each? The same sort of thing applies to S/PDIF systems. IF you want full information you need a impedance VS frequency chart (covering at least 1MHz to 200MHz). So you have one of those for the source and one for the connecting cable and one for the receiver. What do you DO with this information? A human can probably do some gross interpretaions with these, particularly if the spread for each one is not too large, but if each one has a large spread, how do you make meaningfull interpretations of how the system as whole is going to behave? There ARE ways to take this information into a computer and simulate what the total SYSTEM behavior is going to be, but then you have to know how the particular DAC product is sensitive to this, different ones are sensitive in different ways. Once a group of people have discovered the sensitivities of a particular DAC then this information might be useful. (note that very few manufacturers even think about any of this stuff, so information on this is NOT going to come from a manufacturer) There certainly are test equipment that can measure this, the telecom industry uses this stuff all the time, but they are not cheap, $20,000 to $100,000. It should be possible to design a piece of fairly inexpensive test equipment that has a USB interface to a computer so the computer can do all the heavy lifting and have it just do this one function over just the frequency range we are interested in. Unfortunately the market will be small, so even if technically it is not too difficult, it will still be fairly expensive. Somebody could probably market such a device for $4k-$5k. If a bunch of DIY types get together to do the circuit design and programming it could be a DIY project for a lot less. Note that DIY, open source, cooperative projects for test equipment are quite rare! It would be interesting to see what would happen in the industry if reviwers started publishing this information. My guess is that LONG term it might be usefull, manufacturers might actually start measuring their products and put some effort into getting things closer to spec. But short term it might be a major problem because most consumers would have no clue what it all meant, the market (audiophiles and reviewers) would wind up fixating on some aspect to make comparisons on, which may or may not have any relationship to something important. The important thing would be to get the market to focus on what is primarily important, low spread, and averaging around 75 ohms. This is something that is very obvious by looking at the impedance VS frequency charts, as long as all the charts are done the same way, have the same units etc. For a particular peice of equipment built using surface mount parts on PC boards the impedance is going to be quite uniform from box to box as long as the board design is not changed and the components stay the same. Changing a chip from one manufacturer toanother CAN have a significant change impedance, since that is usually not a parameter that is "tied" to the part number. Yes it IS possible in many cases to build a device that you can insert between boxes to improve the match, but they would have to be engineered for a specific combination, there is no such thing as one box that you can use that does everything. I suppose it's possible to build a device that measures the system and configures a network to optimize it, but such a device would be exceedingly expensive. It would be way cheaper to just build the sources and recivers in such a way that they don't care. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95614 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Transporter as server (digital outs)- a few basic questions
jh901 wrote: > > > 2) I'm using the coax output of the Transporter for now, but I'm > switching to AES since the rest of my system is balanced. Are any of > the Transporter outs considered superior (for sound quality)? Most > importantly, is the digital output from the Transporter being clocked by > the Cary? (I get confused about "Master/Slave") > > Josh There is no way to tell which connection method is going to be the best before hand, you just have to try them. There are far to many variables in implementation and cables to make generic judgements. No, the transporter is not being clocked by the Cary, it's a one way connection between the Transporter and the Cary. IF the Cary had a word clock output the Transporter could be slaved to it, but from reading the website I see no mention made of a wordclock output. So most likely there is a traditional receiver in the Cary which uses a PLL to extract the clock from the digital signal coming from the Transporter. This is almost guaranteed to be significantly higher jiter than the internal clock in the Cary. The sales literature spends time talking about how low jitter the internal clock is, but nothing about how good the input receiver is, so it's probably fairly generic. (If a company has spent effort making a very low jitter receiver they usually make a big deal about it, the Cary does not) The upshot is that there is not a lot you can do about it, try the three different connection methods and see what you like the best, stick with that. I am the one who is not particularly fond of AES/EBU, sometimes it can sound the best, but my experience has been that in a lot of cases it is NOT the best sounding interface. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95476 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] NAS versus PC - any sound quality difference?
cfraser wrote: > Hey Ron, is there any more handy info about this style/brand of server > that you'd recommend? It sounds like exactly what I want, especially the > low power consumption for 24/7 op. And a "small" useful project just > when I'm out of them. Thanks. I also have a FitPC2 as one of my servers, but I'm running VortexBox software on it (which runs linux) which is free. Its a whole environment designed to be a server. Besides LMS it can also be used as a general purpose NAS if you wish. The FitPC is not the cheapest route to go, they are very small, use very little power, are rugged as all get out, but fairly expensive. You can build your own low power atom or similar computer for a lot less, but they usually don't come close to a FitPC in power usage. Vortexbox also has it's own hardware called the VortexBox appliance which comes pre-installed with the software, it's ready to go out of the box. Its larger and not as low power as the FitPC, but it can hold much larger disks and has a CD drive for ripping. There are some other inexpensive very low power options such as a sheevaplug, but they are not powerful enough to run transcoding and some plugins. But quite a few people have successfully used them. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=95276 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
lake_eleven wrote: > I deleted the existing jive_alsa. Copied 0.6 version, changed the file > access, then rebooted. Am I missing the "sync", Not sure what is this > "sync" step. When changing files the changes go into a RAM buffer rather than immediately onto the disk (flash in this case). When the buffer gets full it gets written to the disk. Thus if you make changes to files and just unplug the Touch those changes do not get written to the disk. The sync command writes the buffer to the disk, it "syncs up" the disk to the buffer. So if you make changes and reboot the touch by pulling the plug you MUST use a sync command first, otherwise your changes may not show up (or even worse, partially show up). The sync command is typed at the command line like any other linux command (over your SSH connection with PuTTY or whatever you are using). The reboot command automatically does a sync, but those of us that have been doing unix/linux for a long time have gotten into the habit of typing a sync before rebooting no matter what. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
Triode wrote: > It has been reported elsewhere that tuning the kernel module parameter > nrpacks may have an impact on sound quality with EDO when using a usb > dac. > > This can be achieved by: > > echo NUMBER > /sys/module/snd_usb_audio/parameters/nrpacks > > Please note that if you try this tuning parameter there are two things > to look out for: > 1) The kernel we use will limit values of nrpacks to between 1 and 20 - > any values larger than 20 will be silently treated as 20 > 2) This parameter is only read at the time the usb audio device is > created. With EDO this occurs while the kernel is booting as we need to > have the device available for the squeezeplay application. I'm not > convinced at present that we can effectively change this parameter with > EDO in operation. I'll produce a test kernel to check this. > > Edit: Confirmed - nrpacks is only set when the usb device is detected. > With the current kernel we can set it to a value between 1 and 20 but > only if it is done before the usb device is connected/powered on, or if > the usb dac is turned off and on after setting the parameter. So to > tune this parameter at present will require manual tuning and then > turning off/on the dac which must also have been on at startup for > squeezeplay to recognise it. I am yet to be convinced of an impact on > sound quality, but I wanted to make sure people know how to tune it to > have any effect... Hi Triode, back in the days when I was working with HRT streamer with my own hacked up jive_alsa and manually editing files I could definitely hear a significant difference in sound by changing nrpacks. If I remember correctly the default is 8, going lower improved the sound and going higher made it worse. The system still worked with nrpacks set to 1 and this gave the best sound of all. I haven't had time to try this with EDO yet, and it's probably going to be a few more weeks before I can get some time to try some of the recent optimizations. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode plugin with M-DAC
JJZolx wrote: > How does jitter get through an asynchronous interface? Isn't that a bit > like talking about jitter on an Ethernet connection? Good question, unfortunately things are not as simple as it seems on the surface. The supposition goes that as long as you have a local clock, the only jitter you can have is what is inherant to the clock itself, but there is more to it. It's primarily noise on the groundplane. Let's first look at how "digital logic" works. You have transistor circuits running in a high gain mode with a "threshold" which is some ratio between the VDD and VSS power nets (power and ground). The signals going between chips are not perfect, they take a finite amount of time to transition between a high and low (and low to high) state, the voltage "ramps" between the states. Exactly when those transistors "switch" is dependant on when that ramp reaches the threshold of the receiver, and that threshold is dependant on the instantaneous voltages on VDD and VSS. Thus any noise on either power or ground will cause the time at which the threshold is reached to vary, otherwise known as jitter. This noise on the supply nets comes from current flowing through the "wires" in three places, the chip itself, the package the chip is in, and the board the chip is soldered to. The first two are just influenced by what is happening in the receiving chip itself. Note that this includes all the input signals. Every time an input changes state current flows through circuitry in the chip causing noise which will add to jitter of other input signals and outputs. How much noise happens internally is extremely chip dependant. A very robust power network in the chip will generate very little noise, but a very robust power network increases chip size and cost, there is always a tradeoff here by the chip makers. Certain chip functions (such as recloking flops) can actually change the sound depending on which manufacturer and logic family is used, simply because of variations in internal power networks. Noise developed across package wires is similar to chip noise, though much simpler since it is JUST dealing with the I/O signals. Here smaller packages are usually better. The groundplane is where all the fun comes in, because here we can have chips whose signals are not connected affecting another chip, and processing going on in a chip affecting another chip. Groundplanes are NOT equipotential everywhere, currents flowing through the plane DO generate voltages across the plane. They are not huge, but they ARE there. Unless you are very careful about parts placement and groundplane design it's very easy to have circuitry on the board causing noise which can significantly increase the jitter of that "ultra low jitter" clock you are relying on to provide a very low jitter clock to your DAC chip. I hope it's obvious by now that this groundplane noise is not static, it ebbs and flows with the switching going on in the chips, which can change due to jitter on the inputs of THOSE chips. Not just jitter but things like packet timing (both USB AND ethernet) can have a big impact on the dynamic nature of this groundplane noise. This is why my prefered method is to have the USB receiver chip powered by VBUS with an isolator on the OUTPUT signals going to the DAC chips. This way the ground planes are completely separate and the noise caused by all the processing going on in the USB receiver chip cannot get get into the sensitive DAC groundplane. Jitter on the signals can still cause groundplane noise, but that is much less than the noise from the processing in the receiver chip. Ethernet has exactly the same issue (if not worse), ideally you would have a separate ethernet processor with an isolated groundplane so all the stuff going on in there is not producing noise getting coupled into the groundplane around the clock and DAC chips. For something like the Touch with an integrated processor there should be separate groundplane for all the digital stuff and the "audio" stuff (clocks, reclocking flops, clock muxes, DAC chip, S/PDIF output). So yes even an asynchronous interface with a local clock can be affected by jitter and other timing issues on the input interface. The causes and fixes are much more subtle and difficult to analyze than the "first order" effects such as PLL jitter. Getting rid of them in a design are much more implementation detgails such as exactly how the groundplanes are implemented and exactly where the chip are places and signals routed rather tghan broad catagories such "asynchronous" or PLL or ASRC. These are real and do affect sound but it's really hard to talk about in marketing literature! John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94822 ___ aud
Re: [SlimDevices: Audiophiles] Triode plugin with M-DAC
The USB isolators block a potential path for a ground loop through the ground of the USB connector of the "computer" (Touch in this case). Whether this is necessary will of course depend on the computer and DAC and how the power supplies are hooked up. There has been so little use of USB DACs with the Touch that all the reports I know of are using the isolators with regular computers, so may or may not have relevance for the Touch. The isolators themselves that I am aware of put the isolator in front of the USB receiver in the DAC, they all use the same chip. These chips DO add a significant amount of jitter to the USB signals themselves, whether this jitter winds up as jitter on the clock feeding the DAC chips (the only place where jitter really matters) is going to be VERY implementation dependant. In some DACs it won't get through to the clock, in others it will. So in some situations it comes down to a tradeoff between higher jitter and lower ground loop noise, or it may not do anything at all. Or it may make things worse! My personal favorite approach is to have a USB reciver running off the VBUS with the logic level outputs going through isolators (I prefer GMRs) to the rest of the DAC. This way everything having to do with the USB stays on the computer domain. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94822 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
lake_eleven wrote: > John, > I have placed my order for HRT Music Streamer II+, hence very interested > to learn the changes to your system with HRT. I have: > - Installed EDO > - Installed TT3.0 over it > - Changed priorities in TT file to Logitech default > - Made the changes suggested by SBGK on the various .lua files > - My buffer setting in the .lua files are 4200 > - Disconnected screen > - Removed Toslink on SBT > > You are also suggesting kernel settings change. Can you share what the > kernel changes are?. Also what buffer size is best. The kernel settings I'm talking about are part TT3.0, they are automatically there when you install TT3.0. I just did some experimentation turning different optimizations on and off and found that the kernel ones made the most difference, and that several settings that made a big difference to analog out made no difference that I could tell with the HRT. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New SBT person old Audiophile
totoro wrote: > Actually, it's not all that clear to me that what you're saying makes > any sense, either. Even if you have a bunch of threads running, > depending on the types of locks they're using underneath (spinlocks or > some kind of sleep locks), if threads are waiting on locks, their mere > existence may or may not produce any constant load. So screwing around > with thread priorities may or may not have much effect: and you have to > remember that setting thread priorities is only a suggestion in any > event: the os thread scheduler is going to eventually get to all of > them. The whole thing is a fairly complex dynamic system, and these > explanations really sound like somewhat optimistic guesses at what is > happening. > > Is it the case that someone here has actually measured the electricity > consumption of the unit over time with these tweaks in place (perhaps > John Swenson?)? If not, in the absence of either hard evidence or some > blind testing or _something_ that can't be written off as pure placebo, > it escapes me why anyone would take this any more seriously than > homeopathy. If this has been done, then I stand corrected, but I haven't > seen any mention of such tests. > > Just in terms of internet forum badinage: how do you know he's > unemployed or underemployed? As a piece of invective, that wasn't really > any better than just saying "you're stupid" back: in fact, it was worse, > since it was based on nothing that can be inferred from the conversation > that had been had previously. Since anyone can claim to be employed as a > sultan, have a phd in astrophysics from caltech, etc at will on this > kind of forum, these sorts of taunts are completely pointless: from the > outsider's point of view, there isn't any reason to believe you are > employed in any more remunerative or intellectually taxing occupation > than pski, so this just makes you look like a bit of a newbie. > > That being said, I can't see how you deserved that attack. Pski did seem > a bit rabid there. I have actually measured ground plane noise differences with some of the TT3.0 changes. I have not done an exhaustive check of all the different tweaks and what they do, let alone come to any conclusions about what the mechanism is for that change in noise, nor any correlations between those noise changes and audibility. I was more about "is there ANY measurable difference". I'm in the process of building a much more sensitive ground plane noise measurement system which should help in further tests. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94912 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
The way EDO works when selecting the 192 digital out will use less memory and will result in less processing going on. (please note that this output will do all sample rates up to and including 192, not JUST 192). What Triode has done here COULD have been done with the original S/PDIF driver (and IS done in TT3.0). Triode has just chosen to bundle the changes in the way the driver is used with the new driver, as far as I can tell there is nothing different in the new driver other than support for 192. The differences are all in the way the driver is used. Are these differences going to affect sound? Maybe. First off it depends very much on how sensitive a DAC is to whats going on with the input. Some are quite sensitive to jitter in the input, noise, reflections etc. Others are fairly insensitive to this. (I have yet to find a DAC that is COMPLETELY insensitive to what's happening on the input) (all this of course is assuming the the bits are all getting across without errors) I don't think anybody has a complete definitive explanation of what's going on, but there are some speculations. My theory is that most of the changes to the S/PDIF signal are coming about because of ground plane noise. There is a perception amoungst a lot of people that a ground plane is a large equipotential (same voltage everywhere) system, this is far from the truth. The impedance of a ground plane is NOT zero, its not a lot, but it is still there. A digital system like the Touch has all kinds of high frequency currents flowing all over the place through the ground plane, the impedance of the plane is enough to cause voltage differences to develope on the plane due to those currents flowing through it. Everything on the board is going to be affected by this noise to some degree. Digital logic IS affected by it, but in most cases it's not great enough to cause any issues. Where it becomes an issue are things like the local oscillators, the recloking flop, DAC chip etc. For S/PDIF output, the two most critical parts are the local oscilators and the reclocking flop. This noise can cause increased jitter on the clock and is also directly injected as noise onto the output. I have actually measured this ground plane noise. I built a simple device that detects and amplifies the noise and sends it to a spectrum analyzer. I can actually see changes in this ground plane noise with changes in what is going on in the Touch. I'm in the process of building a much more sensitive ground plane analyzer, sometime this summer I hope to have it up and running and can do some better tests. Of course then you have to do correlations between this ground plane noise and what changes this makes in SOUND at the output of a DAC. That is a whole different kettle of fish. This is going to be hampered by the way that different DACs will deal differently with any changes. So the upshot is that yes I have actually measured real electrical changes in the Touch due to these types of changes, these changes have the possibility that they may cause audible changes, whether they do has not been determined. If they due cause an affect, that affect will almost certainly be different or non-existent for different DACs. Can I personally hear the difference? On some DACs yes and on some no. On the ones that I can hear the difference it's not a huge night and day difference. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Triode's USB 24/192 plug in - sound quality impressions
I have a HRT Music StreamerII (not the +), it works very well with Triodes plugin and a hub. Adding TT3.0 significantly improves the sound, with this combination it's getting astonishingly good. (it's not the best, my homebuilt DAC still blows it away). The II with TT3.0 is sounding better than several other way more expensive S/PDIF input DACs I have. (except for my own design -- I'm not biased am I?) I had to do a little work to get TT3.0 to work with it, I had to set the buffer to over 5000, and I had to change some of the priority settings. The esiest way to change the priorities was to just turn them off all together. I didn't have time to figure out which one caused a problem. I tested the screen off part of TT3.0 and it didn't make any difference so I left the screen on. The part of TT3.0 that seemed to make the difference was the kernal settings, when I implemented just them I got the same sound improvement. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94855 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Blind listening - TT3.0, HWmods and Teddy Pardo PSU
Hi Rolf, have you tried listening to the analog outs of the two Touchs? My experience with putting lower jitter clocks in SB boxes is that it does affect the analog outs. I haven't measured the clock mod that was done, but I HOPE that anyone who is making a 3rd party clock board is going to be producing one that is much lower jitter than the factory oscillators. John S. JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94418 ___ 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] The answer my friends is...
SuperQ;699224 Wrote: > Nice, does it play FLAC? Hmmm, remember the 4 channel disks from the seventies? They encoded two extra channels in an ultrasonic carrier, you needed a special cartridge to follow those high frequency wiggles in the groove. It just might be possible to encode flac in that ultrasonic carrier. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94428 ___ 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
gital 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. -- 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] Blind listening - TT3.0, HWmods and Teddy Pardo PSU
NoRoDa;699015 Wrote: > Interesting findings yesterday! > > HWmods made no difference, but at least did not sound worse. > > Regards Which hardware mods had been implemented? Were you using coax or optical to the DAC? Thanks, John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94418 ___ 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
ld be better? Unfortuntely this process DOES change the bits! That is it's whole purpose in life, for a given point in time it looks at a whole bunch of samples before and after that point and takes a little of this, a little of that, munges them around and comes up with a value that it thinks should be the vallue for the output clock. I'm not going into the details of how they work (its fairly complex), but theoretically it should do a good job. But unfortunately we live in the real world and EVERY ASRC design on the market today makes compromises in the design. They all cut corners to save cost. Now of course every designer will tell you "Of course it doesn't make any difference, you can't possibly hear the results of THAT". >From my listening to these devices I've come to the conclusion that yes you can hear the difference. Its not an obvious distortion, its an obscuring of detail, the infamous "veils". So why are they so popular? Because it DOES work, its the great equalizer, you don't have to work nearly so hard on the input circuit, just shove ti through an ASRC and you cut down significantly on the outside world interactions. BUT then you get that subtle degradation of sound. The upshot is that a DAC which does NOT use an ASRC and spends a lot of work on the input side, AND you feed it with a good source, good impedance match etc will sound better than the same design which used an ASRC. So this is where we are in audiophiledom right now, people that have DACs with ASRCs can pretty much ignore a lot of this tweaking sources, cabling etc. But those whose DACs do not have ASRCs will get different results with the tweaking etc, and if they get it right will have somewhat better sound than those with ASRC DACs. If they DON'T get the tweaking right it will sound much worse. Every user needs to decide where they want to be. Choose an ASRC DAC and ignore all this tweaking, or get a non ASRC DAC and and sweat the details in the hopes of someday getting it just right and getting blown away. BTW what do I like? I prefere #2 in the list above, its hard to do right but the results are astonishing if you do get it right. Of course then you have to deal with the ground plane issue, but thats a whole different ball game. 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] Which Squeezebox?
ralphpnj;697252 Wrote: > All true but since this is the Audiophile section of the forum we must > mention all the things that are not "high end" about the Touch. > > 1) It is made of plastic rather than machined from one solid piece of > aircraft grade aluminum. > > 2) It has a simple wall wart power supply instead of a super duper, > machined from one solid piece of aircraft grade aluminum, linear power > supply. > > 3) While it has a USB input it does not have a USB output (neither > asynchronous or synchronous), which is rapidly making the Touch a > non-starter for most of the high end audio press. > > 4) It features bit perfect playback of digital audio files in almost > all common formats (mp3, wav, flac, etc.) with bit depths and sampling > rates up to and including 24 bit and 88.2/96 kHz. It does not support > playback of 32bit / 176.4, 352.8, 192 or 384 kHz files without > down-sampling to 24/88.2 or 96 kHz - thus making the Touch a > non-starter for most of the high end audio press. > > 5) The Touch can stream almost all internet radio and other audio > streams which means that one can listen to the Touch in the background > while writing and editing one's review a music streaming device from an > established high end audio manufacturer which is machined from one solid > piece of aircraft grade aluminum and features a super duper, machined > from one solid piece of aircraft grade aluminum, linear power supply, > an asynchronous USB output and full support for playback of 32bit / > 176.4, 352.8, 192 or 384 kHz files without down-sampling. > > 6) The Touch also features very easy setup which does not require that > the CEO of Logitech come to your home to assist you with the setup. > Again just one more thing that makes the Touch a non-starter for most > of the high end audio press. > > Otherwise the Touch is a really good little audio device :) The comments on the Touch and USB are not true any more. Triode's recent work has made it possible to connect most USB DACs to the Touch, including audio class 2.0 DACs which can play 192. Yes you can play 192 with these DACS! He also has a kernel which will play 192 over the S/PDIF output of the Touch. BTW the Touch has always been able to play 16 bit adaptive USB, its been the 24 bit async that's been the problem. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=94253 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side
pippin;694602 Wrote: > Cache miss rates would be the same or lower with flac. In the end you > are processing the same amount of data and it's a lot and it's size is > determined by the DECODED file size, the ENCODED file size is lower > with flac and you actually have to process all of it, even if wav is > just a raw pcm stream. > > So if your rationale holds, wav must be worse than flac. Actually I was thinking more about code than data. The assumption being that a mainloop for processing PCM is probably a lot simpler than the loop for processing flac, thus giving a higher probability of a cache miss on the code. I haven't actually analyzed the code or run them through a processor simulator etc so I don't know for sure. In a couple weeks I will have my new groundplane noise analyzer up and running and I will be able to tell if there really is a difference there. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93970 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side
I'm going to stick my neck out here and say that I am one of those people that can tell the difference between flac decode on server or Touch. I have tested it with DBT and it is there. Its not huge, many other things make a bigger difference. I have not been able to tell any difference between the source file on the server being flac or wav. I did do a trial with streaming flac with different compression levels, that one was pretty inconclusive. There was maybe a hint that the higher compression levels are slightly worse, but I wouldn't stake anything on it. As to the exact mechanism, I don't know, I have a guess, but no easy way to test it. The standard explanation that its "processor load" I'm pretty sure is false, my guess is that its differences in processor cache hit rates. The lower the cache hit rate the more the main memory has to be accessed. Every memory access puts a large electrical load on the system which shows up as noise on the PS and groundplane. I have done some tests with a simple ground plane analyzer and DO see differences in noise level and spectrum with different things going on in the Touch. I'm in the process of building a much more sensitive device and will be sure to do some tests with streaming type once I get it up and running. I run my system wired to the Touch with everything sent PCM, I have not had any downside to streaming PCM, and it does have a slight improvement in sound for me. If doing so caused any issues (not playing things, dropouts, cliks, pops, etc) I would have no qualms about streaming flac. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93970 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side
I've looked carefully into the 192 issue with the Touch and I see no hardware reason why it cannot be done. The crystals are the correct frequencies, the reclocking flops will easily work with those frequencies etc. The only hardware issue might be TOSLINK. Very few TOLINK receivers have a high enough bandwidth to reliably handle 192, even if it could the jitter at the TOSLINK receiver is going to something fierce, it would take a very special DAC to deal with that. The coax output is a different story, it should be able to handle 192 without any problems. The big issue is the driver. It simply was not written for 192. Someone would have to sit down with the processor databook and work out all the register values to allow 192 to work and then rewrite the driver. There is more to it than even that. There are a number of buffers in squeezeplay and ALSA that would probably have to change as well. The parameters have all been originally tuned for 44.1, at higher sample rates they may start not working quite as well. At 96 some of these are getting sort of marginal, at 192 the system may become pretty unstable. Its not necessarily that the hardware "can't keep up" but that the software needs to be retuned for higher sample rates. I've seen a few places where the code has already been tweaked slighty to handle 96, there will most likely be more of these for 192. So while I think it could be done, and it is "just a matter of software", the probability of it happening anytime soon is pretty low. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93970 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Touch output caps
If you are going to perform surgery on the Touch you are much better off getting rid of or bypassing the output caps all together. The DAC chip already has a circuit which provides a signal that has no DC bias, there is no need for a cap on the output. I really have no idea why there is one in the first place. The only thing I can think of is that the engineer that added the caps thought the DAC had a DC bias on its output since it just uses a single 5V supply. (BTW the DAC chip has a switching DC-DC converter inside to make the negative voltage) Because the cap has its + terminal connected to the DAC chip you want to apply a negative voltage to the other side to properly bias it. 6V should be about right. An easy way to do this is a 6V alkaline lantern battery. Connect the + lead of the battery to GND of the interconnect, then run a resistor from the - terminal to each signal wire. (that's two resistors, one for each channel, the other ends connected to the negative 6V) IF the amp has a film cap (or bipolar electrolytic) on its input, that's all you need. IF the AMP is DC coupled you will need to add a film cap to each signal channel. If you don't know what is on the input of the amp the safest bet is to use the caps as well as the resistors, but if you know the amp already has one, there is no reason to add another. You can play around with the resistor value. I'm going to say somewhere between 10K and 50K. The higher the value the less current will flow through the battery, which means a longer battery life. Higher values have a little more noise so there is a tradeoff here. You can use inexpensive resistors or expensive resistors, that's up to you. If you need the cap, that starts getting a bit more complicated. The bottom end of the range is probably .5uf but you could also go up to several uf depending on the amp input impedance etc. There is also a huge range in price for different caps, from 20 cents to several hundred dollars. Again I am not going to make ANY recommendation on that. Just remember that there is already a cheap electrolytic coupling cap in the chain. I was specifically looking at an article that was analyzing what happens with an unbiased polarized electrolytic while varying the input voltage from .1V RMS to 2V RMS. I don't remember exactly where it is, I can't find it right now. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93875 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Touch output caps
That article is for a properly biased electrolytic, an unbiased one is MUCH worse, which is the situation we find in the Touch. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93875 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
[SlimDevices: Audiophiles] Touch output caps
As many of you know by now the SB Touch includes inexpensive electrolytic caps on its outputs. These caps are polarized, which means they are supposed to be run with a bias voltage across the cap, this bias actually makes the cap work properly. But in the case of the touch there is NO bias voltage, I've been trying to find out what this lack of bias voltage does to the sound. I finally tracked down some articles on the web where people have been looking into this. It turns out that without bias a polarized capacitor still behaves as a capacitor for small AC voltages, BUT the distortion through the capacitor goes up exponentially as the AC voltage increases. The upshot is that at the full output of the Touch (2V RMS) the unbiased cap has 22 times as much distortion as a film capacitor does. At lower output voltages this distortion goes down significantly, but even at its best it's still about 5-6 times as much as a film cap. This has some interesting ramifications for connections to a system. The conventional wisdom says that you shoulkd run the Touch at 100% volume all the time and use an analog domain volume control, thus maximizing the bits used and the signal to noise ratio. But that doesn't take into account this exponential distortion. A number of people have reported that they get much better sound running the Touch directly into a power amp and using the digital volume to bring the level down. This is usually attributed to preamps not being very "transparent", but in this case it might be the significantly lower distortion when you run the Touch at a lower output level. Of course all the above is JUST for the analog outs, it has nothing to do with running the digital signal into an external DAC. So what to do about it? You can use an external DAC, that has been talked about a LOT. You can open up the Touch and bypass the output caps. Opening the Touch and performing surgery and getiing it back together is not an easy task, so don't attempt this unless you know what you are doing and are willing to risk destroying it. There are threads on this over in the DIY forum. You can also build an external bias circuit that biases the capacitors without having to open up the Touch. I'll try and get a thread on this started over in the DIY forum in a day or two. I'm kind of surprised that none of the companies that supply "tweaks" haven't marketed one of these yet. It very simple, a 6V source (AC supply or battery) and 2 resistors. You can put it in a box with connectors, but then you need an extra set of interconnects. You can have pictails sticking out of a box, you can take an interconnect and cut it in half and insert this etc. There are several ways you can do it. Its still not as good as getting rid of the capacitors all together, but its much lower distortion than what you get out of the box, and you don't have to do anything to the Touch itself. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93875 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Tin-like sound
One possibility is that you damaged the internects somehow. I have had it happen that you break the conductor inside, but it is just barely not connected, its so close it forms a capacitor which cuts off the bass. Such a situation will be very sensitive to position of the cable, moving things around will either make the connection normal, or no sound at all. It could also be a broken solder joint in the connector (cable, or Touch), again usually wigling things around will change it. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93825 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Theoretical benefits to hi-res if only max 20kHz signal?
Phil Leigh;692199 Wrote: > So John, are you saying that on 44.1 material you prefer a NOS with a > good analogue filter set for 22Khz vs an ASRC DAC running internally at > 384 or 768 with an appropriate digital filter? > > Just curious. > > That "dirt" you refer to is exactly what I don't like about all of the > NOS DAC's I've heard. Of course this doesn't mean ALL NOS DAC's are > empirically bad... simply that I personally haven't heard what I > consider to be a good one. > > Interesting about the heat phenomena. The heat must be causing the > Johnson noise to rise inside the DAC, no:? - maybe the noise is having > a "dithering" effect? > regards > Phil My preference is a DAC which uses properly implemented digital filters, unfortunately these are rare. The NOS DAC and software approach is a way to get around trying to find hardware with good digital filters. By doing the oversampling in software you have FAR more flexibility in playing with different filter parameters. For my own DACs I've spent years going back and forth between NOS and digital filters. A NOS DAC with several poles of analog filtering can sound quite good. But the best is still digital filtering done right. Of course that takes significant digital horse power, which means you have to be very carfeful that the noise from the filter doesn't make it into the clock circuits, DAC chips and analog circuits. That takes very careful grounding and PS design, none of which comes cheap. I personally do not like ASRCs. I have not heard a single one that I really like. Of course it may not be the ASRC per se. I expect part of it might be the interaction of the imperfect digital filters in both the ASRC and the DAC chip interacting with each other. On the "HOT DAC" I suspect its due to internal PS noise. As the chips get hotter the FETs in the circuits get higher resistance, which slows them down (RC goes up) so the peak current from each switching goes down, thus less noise on power and ground traces in the chip and package. As the temperature goes up at some point things get so slow it stops working, you don't want to go that far! John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93483 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Theoretical benefits to hi-res if only max 20kHz signal?
Archimago;692235 Wrote: > Thanks for sharing your experience with DAC's John! > > Over the years I have played with a cheap DIY 4-chip TDA1543 design > similar to the stuff on eBay as well as a more expensive Mhdt > Constantine (TDA1545 I think). Can't say I was enamoured with the > sound... I found the highs a bit too rolled off for my taste so > happily went back to the Transporter sound. > > Wondering if there was a commercially available NOS DAC you think > represents a good design. I don't have time anymore to fool around with > DIY's. > > As for the slow roll-off filter, any opinions on the Transporter's > AK4396 slow-roll vs. standard filter? If I were going to buy a NOS DAC today I would get the Audio-gd DAC-19, it uses a pair of 1704K chips (my personal favorites). It also does what I do, it implements its own digital filters in an FPGA (strangely enough exactly the same FPGA chip I use in my latest DAC). You can also set it to NOS mode if you desire. Its not the cheapest DAC nor the most expensive. I think it does a good job all around (except for maybe the USB input). I have not had a change to listen to the Transporter's slow rolloff mode so I can't comment on that. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93483 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Theoretical benefits to hi-res if only max 20kHz signal?
ed as cascaded filters they all don't sound so great. With filters implemented as single proper filters (enough internal bits for the number of taps and enough bits for the coefficients) differences in filter functions CAN be heard but they are not very large. Just getting the filters implemented properly is the biggie. This brings up the issue of software upsampling and NOS DACs. First off NOS does not mean no filter, just not a digital filter, you can still put an analog filter on a NOS DAC. If the builtin digital filters are the problem, it seems that a good NOS DAC playing upsampled files that were generated with a properly implemented software filter should provide good sound. And my experience is that indeed it does, especially if you put a 2 or 3 pole analog filter after the NOS DAC to get rid of residual high frequency noise. Note this has to be a GOOD NOS DAC, not one of those cheap 16 bit ones from people that have no clue what they are doing. There are a lot of people that are doing software upsampling and feeding the results into soundcards and external DACs that I think are trying to do the same sort of thing, but the data is still going through the compromised digital filter in the DAC chip. It would be much better if they fed it through a good NOS DAC. An interesting side bar on this is an early experiment I did. I had been reading about people that stacked 8 1543 DAC chips, I tried this in two different ways. One group of people spread the chips out on a board (see the picture somewhere up in this thread), but others actually stacked the DAC chips on top of each other. I tried both and found the stacked on top of each other approach sounded much better. Note this was EXACTLY the same circuit, just a different physical layout of the chips. The difference was that the stacked chips got HOT. Doing so cut off the airflow so they got to much higher temperatures. I hypothesized that it was this higher temperature rather better linearity or lower noise that made the improvement. I decided to test this by gluing a power resistor to a single chip and pumping DC through it to raise the temperature. I added a thermocouple so I could check the temperature of the chip (well the temperature of the case, not the actual chip). I then very slowly raised the temperature of the chip and low and behold it sounded way better as the temperature went up high. (still not all that great, the single chip by itself is a pretty bad sounding DAC chip) So all that theory that it sounded better because of the increased bit depth because of stacked chips was hooey, it sounded better because it got HOT. BTW the one with the 8 chips spread out on the board sounded worse than a single chip, that was just a bad idea. Things HAVE been getting better. The latest crop of chips seem to not have as bad digital implementations as previous ones (with the decrease in cost of compute hardware, its probably cheaper to just do it right than spend the money trying to develop creative corner cutting). That doesn't mean they are perfect. Every one I have tried I have been able to make sound better by disabling the internal digital filter and using a properly implemented external filter. The only ones I have found that seem to do a pretty good job of their internal filters are the Sabre chips. Well there you have it, some of my exploration of NOS DACs. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93483 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
pski;691193 Wrote: > Again: I assume this is software. > > What does it do? > > What is it alleged to do? > > Skeptics, Fans, and Developer are welcome to reply. > > Please reply including one of the above in the title. > > Obnoxious science deniers need not reply > > P Why not just look at it yourself. Its a tar file which contains some shell scripts, just download it, untar it and look at the scripts to see exactly what it does. You can do that on any computer that has a way to untar a file. You don't have to be on the Touch to look at it. It has 4 primary categories, kernel parameter changes, thread priority changes, ALSA configuration changes and turning off things. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93649 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
richieleo;691496 Wrote: > ok then... > > I came across Klaus' blog and by all accounts, the toolbox works > wonders. > > However, I must admit that I am somewhat of a amateur when it comes to > doing mods like the ones he suggested and he has made it easy at least > for windows users. I, however use a mac at home. I cannot find a step > by step (and I do mean step by step) guideline to follow to modify my > squeezbox touch to be used with my mac. > > So I guess I am asking if someone can help me out or can point me in > the right direction? I mean, I am not sure how to even install the > toolbox (yeah...I know...really bad)! > > As of now, the sound streaming from my apple tv is way better than the > squeezebox (even with an upgraded powersupply). > > I hope somone can help. :( Hmm, I seem to remember that there ARE MAC instructions for TT3.0. 80% of what you do is the same no matter what platform you are on, its just the tools you use to get access to the Touch that are different. You want to look at the section for MAC and linux, they are the same. The only difference is that you have to start a command window on the Mac, this I don't know how to do since I don't have a Mac, others can tell you how to do that. Once you have THAT done what you type will be the same for Mac and linux: scp to copy the file to the Touch and ssh to login to the Touch. Once you get that far all platforms are exactly the same. As to the Apple TV sounding way better, that's a bit unusual, I don't think I've ever heard anybody say that. If you don't mind could you give us some details on your setup, what you have LMS running on, are you using ethernet or wifi to connect the Touch and or computer running LMS. Are you listening to your own files or internet radio or streaming services? If you are listening to your own files are they on another computer or are you using a USB drive plugged into the Touch? What file format are the files? What are your file format settings for the filetypes you are using? The Touch itself handles a certain number of file formats by itself, the server can handle a much larger set of file types and transcodes those into one of the formats the Touch does understand. The file types is how you tell LMS what "streaming format" to send the audio data to the Touch. Its possible that you have things setup to use a lossy format to send the data. You get to the file formats by using a browser to login to the LMS web page :9000 then click on settings, then advanced. Then click on the pull down arrow on the left side and select file types. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Soundman;691031 Wrote: > John, please have a closer look. These are the commands in TT3: > > Options: > -i : initialize toolbox > -s : toolbox status > -x : display on/off > -k : kills daemons > -w : wlan on/off (persistent) > -v : locks volume at 100% (persistent) > -d : display on/off (persistent) > -ir : infrared receiver on/off (persistent) > -b N : buffer size N= 3200-5 (persistent) > -o N : output routing (persistent) > N= 0=digital 1=analog > -r : restore original configuration > -rbt : reboot > -h : help > > The commands "-k" (mod), "-h" and "-s" (both not mods) are the only of > these which work without a reboot. And after "-k" a reboot is required > to open access to the box again. The last remaining command would be > "-x" to switch the screen on and off on the fly, but this is only > possible if you use "-d" before to switch the screen on (which is off > by default). But as I said before: this is pointless because the mods > only really make sense, if you use all (or most) of them together. The base install adds a bunch of changes, many of which do not need a reboot to take affect, but they do not have a separate tt - switch to turn them on and off. Even the wlan and volume lock do not NEED a reboot to take affect, its just how Klaus has implemented them in his system. He has some scripts that run at boot time with the modifications, the tt - commands modify the scripts, they don't actually make the change to the system, then a reboot is done which applies all the system modifications at once. Its by far the easiest way to make sure that both changes that require a reboot and those that don't get properly applied. If you want to apply these changes only through the tt interface then yes you are very limited as to what can be done. But you should be able to take some of the modifications that do not require a reboot and put those directly into Triode's framework without having to use the tt interface. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Soundman;690874 Wrote: > As I already said: The only TT-mod that works without rebooting is "tt > -k", which by itself doesen't make a big difference. But for me it's > not so important anymore to make more tests, because my friends and > myself (five people) did a blind-test and all five were able to > identify the Touch with the TT-mods (accuracy was above 95%). And all > the five of us concluded, that the Touch sounds better with > Soundcheck's mods. Why sould I do more tests, since I'm very happy with > my modded Touch? Klaus did a very good job indeed! Klaus has grouped a bunch of changes together, some of which require a reboot and some which do not. It should be possible to run the ones which do not require a reeboot under Triode's framework. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] psycho acoustics: the nightmare of listening tests
I have no problem at all with the concept of doing a DBT as long as the protocol allows for some time for relaxed listening. Here is a proposal for a DBT of the infamous TT3.0. This one is ideal because it's software only, no switching between hardware needed which should simplify the testing procedure. A procedure gets written which can switch between TT3.0 and "standard" by loging into the Touch from a remote computer. (this should not be too hard to do) Three scripts are written which can be run from a remote computer which switches to "A", another which switches to "B" and another that does an "X" which may be either A or B. The sequence of X configurations is determined by an excrypted file. The person under test does not know the sequence. Another program generates the encrypted files which are emailed to the participants. At this point no human knows the sequences in any of these files. After the test is complete the same program will generate the human readable sequence in each file. During the test itself the participants will have the Touch hidden from view (it's possible to tell which is which if you can see the screen durring the reboot). All interactions with the Touch will be done through one of the remote options (iPeng, Android app, server web page etc), none of these have anyway to tell if TT3.0 is active or not. The person under test can use the three scripts to select A or B or X as many times as they want, listen to as many songs as they want, take as much time as they want, do it in one day, do it in a week, whatever. Some things for discussion, how many X changes should be done? Should the identity of A and B be known by the testers? The encrypted file could also be used to keep the identity of A and B secret. Possible ways to cheat: look at the Touch when rebooting. For this to work we have to trust people not to do this. Logging on using SSH to see if TT3.0 is urned on. Again we have to trust people on that one. The sounds made durring boot could also be used to tell which is which, but these can be globally turned off. We can just have the scripts do that. Any takers? Any discussion? Does this satisfy the DBT "rules"? John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93380 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Phil Leigh;689021 Wrote: > > > The ONLY possible way in which the server/network could affect the > sound quality on a wired network is by introducing electrical noise > into the touch via the NIC. If this was happening - and it IS possible > - the cable pull test will detect it. > Actually there is another way: packet timing. When a packet comes in the processor does a lot of work moving that data around to different buffers up and down the protocol "stack" causing lots of memory accesses, which creates noise on the PS traces and groundplane. Different packet timing causes different spectrum of this noise. I have actually measured this on the groundplane of a different small computer (FitPC) (I haven't done it on a Touch because its a pain to open and close the thing!). Now how this relates to sound quality is a whole different story, but its not inconceivable that spectral changes in groundplane noise might affect the sound. Changes in OS thread priorites, scheduling policies etc CAN change packet timing, as can using a switch, going from GB to 100MB etc. And as you mention pulling the cable should should fairly quickly stop any noise generated by processing packets. So I think it is possible that what is going on in the server and or the network could have an impact on sound. Whether it actually does will take a LOT of experimentation. Because things like switches in the path can completelly change the timing behavior, network topology is also very critical. There are a lot of factors to take into account. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] psycho acoustics: the nightmare of listening tests
darrenyeats;688938 Wrote: > John, imagine we picked two pieces of sky, one with a nebula and one > without and we repeatedly and randomly showed you each and asked if it > was the nebula or non-nebula sky you were seeing. If you got it right > 50% of the time, what should we conclude? > > DBTs do not necessarily entail "snippets" - that is a misconception. > You can take as long as you like. There is no excuse for rejecting > blind listening as a concept. > Darren Isn't that exactly what I was saying? I thought I specifically said that DBTs are valuable as long as you go into them understanding that the very act of trying to concentrate on differences can alter your perceptions. I guess I didn't make myself clear enough since everybody is assuming I said DBTs are not valuable. Oh well, I'll try better next time. On the "snippets" bit, I know that very well, its just that I have seen and been part of quite a few tests that do just that, I was trying to point out that this type of test fosters the type of testing I find less valuable and there is a different way to do it. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93380 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] psycho acoustics: the nightmare of listening tests
different equipment shows the contrasts. I know there will be some who will say, "but how do you know this "contrast" is real and not something the brain made up?" My take is this: we know the brain filters what we perceive, what is more likely, the lack of contrast is reality and the more contrast is made up, OR the lack of contrast is the brain filtering out information and the higher contrast is closer to reality? My experience has been that the brain does far more filtering than it does creating out of nothing. So I'm going to go for assuming the greater contrast I hear when listening to the totality of the music is closer to reality and the lack of contrast when critically listening is the brain filtering information, thus I'm going to make judgements about what makes a difference based on what I hear when in the "totality" mode rather than in the critical listening mode. I hope all this makes some sort of sense. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93380 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Audiophile rednecks
maggior;687656 Wrote: > - If your new DAC sits on top of your old broken DAC, you may be a > redneck. > > (attributed to Jeff Foxworthy's similar comment regarding a new TV > sitting on top of a broken console TV). When I was young (a long time ago) everybody had large black and white console TVs, when color sets started coming out people would frequently get a "portable" color model that had to sit on top of something, what better place than the old B&W console. No need to buy a "TV stand". John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93236 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Phil Leigh;686368 Wrote: > Quite. It was the ground loops I was thinking of. It wouldn't be much > good transformer coupling/isolating a DAC if the coupling didn't pass > high frequencies :-) Hi Phil, I wasn't refering to you, I know YOU know how it works, but there are quite a few people that seem to belive that that sticking a pulse transformer on a S/PDIF stream automatically cleans up all noise. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Just a bit of clarification on galvanic isolation via transformers, there seems to be some mis-understanding about them by some people. These devices are wideband transformers designed to pass a very broad range of high frequncies. They block DC and low frequencies, but pass most high frequencies. So they work well at blocking "ground loops" (60 or 120 Hz), but any high frequency noise on the S/PDIF stream is going to go right on through the transformer. They can even make it worse. ALL transformers have various resonances, if the circuit the transformer is in doesn't damp those resonances they can wind up increasing certain incoming noise frequencies. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
vett93;685930 Wrote: > Thanks. I have tried and implemented many tweaks in my audio systems. > But reducing buffer size just does not make sense to me at all My current thought is that buffer size affects the processor cache hit rate. The assumption here is that one of the main causes of noise in a computer (power rails, groundplane and radiated) is the main memory bus. These are very wide fast busses, when data goes over these busses the drivers produce huge current pulses which cause lots of noise. Each bit is not too bad but there can be a hundred bits switching at exactly the same time. So to me what makes sense is to cut down on main memory access. Small buffers mean that its possible that the data can stay in the processor cache and never have to go out to main memory. There is a tradeoff here because a small buffer also means more interrupts and process switches which means more code gets swapped in and out of main memory. This is something that is really hard to measure since OSes don't usually give you a processor cach hit rate graph you can look at. (of course the process of monitoring it would decrease the hit 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=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Point of increasing returns
magiccarpetride;685370 Wrote: > I now feel that there comes a time when you somehow manage to fine tune > your audio system to the point where apparent minor upgrades carry the > potential of making major inroads in terms of improved bass > reproduction, lowered noise floor, lowered upper midrange glassiness > and glare, more crisp and unwavering soundstage, etc. > > Raise your hand if you're with me! I have experienced this several times in other fields as well as audio. One of the most dramtic is in color printing (making color prints in a darkroom, film chemicals all that sort of thing, for you young people this is the way we used to do it back in the stone age). The problem was that the color characteristics of different film and paper varied significantly, you had to use different color filters in the light path to get a good color balance in the print. When you started you made a guess, set the filters appropriately and made a test print. Invariably it was not right, it was too green or too magenta etc. You then made another guess as to how to change the filter "pack" to push it in the right direction, you started out with large changes which didn't seem to make much difference. Then as you got close to the correct filter pack, there was a massive perceptual shift, the result started looking "real" and now very small changes in the filter pack made big differences in the perceived image. If you actually measured the differences in the colors in the print they were much smaller than the changes you were making at the beginning, but they LOOKED much larger when you were close to "getting it right". I've experienced this in audio systems many times. If you have something in the system that is causing an issue it can push the perception of the sound far enough away from "real" (whatever that is), that other fairly large changes don't make much perceptual difference, but when that issue gets corrected now all of the sudden small changes in other areas can start becomming perceptually significant. And as others have stated, of COURSE its all in your head, its about perception. It doesn't have to be an issue that is excrutiatingly bad either. Sometimes it things like room acoustics or speaker placement. At one point I moved speakers around in the room, it didn't make a huge night and difference, but now other things being done to the system started making a bigger difference in the perceived sound. At one point someone sent me a DAC to look at, the designer had done a very good job in certain parts of the analog circuit, but no so great in others. The owner had been tweaking the analog stage and was expecting some differences in sound based on doing similar things to other DACs. I took this DAC and analyzed the digital section and found the clock had fairly high jitter, I modified this circuit and reduced the jitter by about 20 times (it still wasn't really low jitter, but it was much better than before). NOW those changes in the analog stage did make significant differences. Now further changes to the DAC circuit to get get the jitter down into the really low jitter ranges made big differences in sound. If I'd have made those changes originally you never would have heard it, they would have been swamped by the bad clock to begin with. Note that at no point did this DAC sound "BAD" or obviously distorted, it just made music sound significantly more real as the changes were made. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=93154 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
For those taling about modding the S/PDIF out of a Touch a few thoughts: the S/PDIF stream is generated in the processor chip itself, it then goes to a reclocking flop, its in a very tiny single gate package, from there it goes to a network of a couple resistors and a cap, which I presume are doing an impedance match from the flop to the output. Do NOT bypass that flop! The signal coming directly out of the processor is quite bad, the flop reclocks it with the raw clock. As to putting in a BNC jack, it certainly can be done, but make sure you use a real 75 ohm jack, BNCs come in both 50 and 75 ohm types, the 50 ohm types are far more prevalent than the 75 ohm types, if you just get something that says its a BNC jack it will probably be a 50 ohm type, it has to explicitely say its 75 ohms. As to the clock in the Touch being almost perfect, its not. Its a normal cmos inverter and xtal oscillator, they are not "BAD" but they are not great either. Its definately possible to get oscillators that are MUCH better. And going with a better clock does significantly improve the sound. A while back I was doing a I2S out, clock fed back from DAC scheme and decided to listen to the analog outs just for kicks, I was blown away, the sound was way better than normal. I actually expected it to be worse, what with the issues of sending the clock over the cable I expected the clock jitter to be worse. Even with the cable the clock was better than the built-in one and it was quite obvious. So I would say that improving the clock is actually quite a worthwhile mode if you can do it. Unfortunately there is VERY little room in there to do it. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
I have a theory as to what is going on, and SOME actual measurements to back it up. The theory is that what is most important is main memory access, NOT processor load. Think about it for a minute, the processor is connected to the main memory chips with a bus that is hundreds of bits wide, when a memory access occurs a large percentage of these switch all at the same time (actually within 50 pico-seconds of each other). The capacitance on a wire going from one chip to another is orders of magnitude greater than the capacitance inside a chip, thus every time that external bus switches huge currents flow in order to charge and discharge those capactitances. This produces huge currents in the groundplane and PS traces as well as EMI emitted from the signal wires themselves. My analysis of groundplanes indicates that noise caused by the memory interface is much larger than that from the processor itself doing its thing. Thus the most critical thing you can do is to cut down on main memory accesses. The easiest way to do this is to make sure that as much as possible stays in the processor cache. This I think is where most of these tweaks actual work. Decreasing the buffer size means that frequently the buffer can stay in cache and the data doesn't have to go in and out of main memory as much. Same for processes, every time the OS switches to a new process the cache is cleared and code and data have to come from main memory. This can also help explain things like flac or PCM decoding. The flac decoding may not take all that much more processing power, but the code is much more complex, the PCM code will have a higher probability of staying in cache. This may not be everything that is going on, but I think it is a major contributor to what is happening. Unfortunately there is no tool you can run that gives you a graph of processor cache misses, it would take a logic analyzer looking at the memory bus. I think that focusing on processor utilization is a red herring and not worth worrying about as long as it doesn't get too high. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Update: I left everything on overnight and now the output from the DAC sounds amazing, I guess something needed to warm up. But the DAC and Touch are always on so its most likely the big tube amp which I don't leave on all the time. I've never heard such a big difference between only been on half an hour and been on for 10 hours. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Hi all, you may have noticed that I have been absent from this discussion so far, I have been very busy developing a new DAC with a radical new S/PDIF input stage which hopefully will make some of this tweaking a little less necessary. Anyway the first prototype of this new DAC is going out to get assembled tomorrow, so I found my self with a free evening, so I decided to finally try TT3.0. I first tried it with the analog outs. It was pretty good, maybe a bit better than 2.0, but still not nearly as good as a GOOD external DAC. Then I tried the daemon killler, WOW, what a difference that made. That opened up the sound significantly, a wide sound stage, tons of detail but smooth and "organic". Bass was significantly extended. I was listening to an organ and choir piece I have listened to many times before and heard a counter melody in the lower pedal notes that I had never noticed before. Then I tried into some S/PDIF DACs, very strange, this actually didn't sound as good as the analog out! I was getting that infamous "fuzziness" people have been talking about. Very strange. I haven't had time to try out all the different priorities people have been talking about. One of the DACs is my own design from about 2 years ago, which sounded really good with tt2.0. >From trying different DACs it sounds like there is something going on here which is exciting differences in S/PDIF input receiver circuits. Different ones seem to respond differently to the same settings. It does seem like there might be a fair amount of tuning going on rather than an absolute "best for everything" setting. On the analog outs, especially with the daemon killer, tt3.0 seems to be a big step forward, making the results sound better than anything else I had heard from most external DACs. (remember this is with a good external supply and output caps removed) But with S/PDIF things seems to be a little more challenging to get really really good results. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] help me find linear power supply for Touch?
Putting a choke on the output of an existing supply does not do anywhere near the effect it does when it put in the right location, between the rectifiers and the regulator. And you can't just put a choke into any existing linear supply, the whole system has to be tuned to work properly. The value of the choke, caps and parameters of the transformer all have to be right to get the best results. If the values are wrong you could very easily make things worse. The choke does two primary things: It loads the transformer properly so no high frequency harmonics are injected back into the mains and there are no high frequency harmonics sent on to the regulator. With a common cap only filter current only flows out of the transformer/rectifiers when the voltage is greater than the voltage on the cap. This means current flows through the transformer in short high current spikes. For example I've measured a simple cap only design, for the Touch and it has over 20 amp current spikes, even though the Touch is only drawing 1 amp! The high current spikes inject a lot of noise back into the mains, which get into your other components. The output waveform from a cap only supply is a sawtooth, the transformer charges the cap up quickly (those 20 amp spikes), then the load discharges it slowly, this results in a sawtooth. This waveform contains lots of high frequency components which are fed to the regulator. Most regulators work very well at dealing with low frequencies, but do less well as the frequencies increase. This is one of the reasons why a lot of the "audiophile" supplies use high quality discrete voltage regulators, they do a MUCH better job of dealing with high frequencies. But with a properly designed choke supply there are no high frequencies being sent to the regulator, just a pure 120Hz sine wave which just about anything can deal with. There are other advantages but these are the two main ones. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91383 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
cymbop;671090 Wrote: > I've got a hunch that the VU and spectrum meters are running in the > background even when the screensaver (set to "Screen off") kicks in on > top of it. They're part of the Now Playing view unless you kill them > manually. > > I don't know much, though. Eager to see if anybody else can hear the > difference. The meters and spectrum analyzer are not separate processes. They are implemented as functions in the low level C code that handles the audio processing. There is a check at the beginning of the function that checks a variable to see if the appropriate screenview is active and just returns if that capability is not being displayed. Whats probably happening in that with the display turned off you can't tell if the meter or spectrum analyzer is showing or not. Randomly touching the screen can switch between the display modes so it's certainly possible to accidentally turn on one of these modes without realizing it. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] help me find linear power supply for Touch?
Mnyb;669537 Wrote: > A likely explanation , beside the one that the ps disturb other > equipment . > > So called linear supplies also disturbs the feeding net by a great deal > their input current is very distorted . > > If had the brains to design a supply I would design a supply that draw > sinusoidal current from the net with a power-factor close to 1 this can > be done with switch-mode and LCL filters (I work with such devices but > they are in the MW range ) > > Lore ? I think everything never than the direct heated triode is > considered bad ;) . > > It is quite possible to design both bad and good supplies with any > method, as folklore is part of audiophiledom it is hard to sell in a > good switch-mode supply, instead you put together a run of the mill > linear supply in nice looking aluminum case and you earn money :) This is one of my big issues with a lot of linear supplies. The design I have posted on these forums uses a CLC filter which loads the transformer with an almost pure sine current load. It also feeds pure 120Hz to the regulator which makes it's job MUCH easier, it doesn't have to deal with all those high frequency harmonics. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91383 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0
Hi Soundcheck, on the USB front I have been working on a udev setup that consistantly numbers all devices. Its going to need something in rcS or rcS.local and since you are using that I would like to get it integrated in with yours so they are not fighting each other. A while back rgo posted a udev script which always numbered the USB dac card3, but unfortunately the other devices were frequently also renumbered. I know have a udev setup that always numbers the USB dac as card5 and all the other devices (analog, digital and effects) always have their normnal numbers no matter what. Its working great now for booting with the USB dac plugged in, but hot plugging is not working yet, I need a little more time for that. It would also be cool to put in a system that could automatically choose different alsa setups depending on what is plugged in, but that is probably a next generation enhancement. On the USB dac fron in general I still have not got it completely figured out, I'm still getting ocassional xruns no matter what I do. I'm starting to belive its down inside the low level USB driver, but I'm not sure about that. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91322 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Stylish tiny amp for Touch
castalla;664451 Wrote: > I read this note on the Muse m50: > > The machine is BTL output, that is to negative side of speaker out is > not public grounded (the two speakers cable must independent connect, > no grounding together), we must strictly in accordance with the marking > of plus or minus cable > > I've no idea what this means - just sounds scary to me as a complete > audio novice! > > Can anyone explain? On many "traditional" power amps the negative (-) outputs were connected together in the amp. It was actually possible to have a common ground wire for both speakers. This is NOT true for this amp, you cannot tie the negative outputs from the two channels together. As long as you have two wires for each channel and don't connect then to each other or to anything else there wont be a problem. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=91027 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Headless Touch
You can write a script and give it the right name and put it on an SD card or USB stick and it will get run at boot time. (I don't remember the name of the file off the top of my head). You could put the commands to start up the SSH server in this script, no need to turn it on with the display. Once you can get in you can modify the startup scripts so you don't need the USB/SD anymore. You can do a lot of things with this script if you want. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=90739 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Optical (toslink) connection
decide if its worth it. But if you try enough you will probably find a less expensive one that sounds as good. The best advice I can give is "don't take anybody elses opinion as gospel", there is such a huge variation in equipment so that what sounds the best to one person is almost guaranteed NOT to be the best for you. You HAVE to do the testing for your self with your system. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=90211 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Optical (toslink) connection
The receivers add a fair amount of jitter on their own. Either the photodetectors or the electronics used to amplify them are slow and noisy, the signal takes a long time to rise and there is a fair amount of noise on that ramp, which means the uncertainty in the threshold can cover a fair amount of time causing jitter. There are ways to make this much better, the best of which is to have a multistage amplify and clamp system, this can detect the optical edge with very little added jitter, but costs way more than the "audio" receivers they currently sell. Toshiba certainly knows how to do this, some of their optical systems do it very well, but they don't offer that technology with the connector used for audio. 7 years ago I built a DAC with a TOSLINK receiver that has significantly lower jitter than what you can buy today, but Toshiba decided not to make those any more. (Probably because very few people were willing to pay the extra cost, it was something like 4X the price) Yes you CAN get extremely good performance on optical interconnects, the ones I deal with at work have about 10ps of jitter, but they cost several hundred dollars per channel, its hard to get that level of performance for 25 cents, which is about what the electronics in an audio TOSLINK receiver costs. If you boost it up to $2 worth of electronics you can get much better performance, but now the receiver costs $12 instead of $1.50. It seems there just is not a big enough market for $12 TOSLINK receivers. What I don't get is how come some enterprising people haven't started selling the improved optical receivers with the $2 of electronics for $100 a piece to the audiophile market. For people that drop $500 on a cable the extra $100 for a really good optical receiver should be a slam dunk. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=90211 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Optical (toslink) connection
Yes the fiber can handle huge bandwidths, but there are two major issues with common "TOSLINK". One is the fiber itself, its a "multimode" fiber, its physically much larger than a wavelength of the light used. The result is that light entering the fiber at different angles can take different path lengths (hence time delay through the fiber), the result is that edges get stretched out in time. There are some interesting attempts at fixing this, the most common one today uses a bundle of single mode fibers, so whichever fiber the light ray hits, it takes the same amount of time through the bundle. The other is the optical <-> electrical converters. 99.9% of the industry just uses off the shelf TOSLINK transmitters and receivers. For whatever reasons these are not very good parts, particularly the receivers. They are slow and noisy. They have actually gotten worse over time. Toshiba used to make some pretty good ones but they stopped making them several years ago. (I think HDMI has pretty much killed off the optical market and Toshiba couldn't make any money off of higher end modules). It is certainly possible to do MUCH better than the TOSLINK parts, BUT none of those have the same connector. Quite a while back there was an attempt to do this with the ST optical interface, but it never really caught on. Until someone starts making a GOOD optical receiver with the same TOSLINK connector not much is going to change. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=90211 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 2.0
The issue of ground plane noise can cause problems even with an optical connection. It has to do with the transmitter, either electrical (coax) or optical, the input to the transmitter has a "threshold", a voltage at which it sees the input as changing from a one to a zero, noise on the ground pin of the transmitter causes that threshold to move up and down causing the point at which it senses the signal as changing from one to zero to change. Since signals do not change instantaneously from a low to a high value, changing the threshold also changes the time at which the transmitter sees the change taking place. This happens with both electrical and optical. The ground plane noise can also get coupled to the output as noise on the signal itself, this noise can cause the receiver to misinterpret when the change happens as well. Different receiver circuit vary significantly in how they handle noise on the signal. There is a famous example of a circuit that tried to be very immune to input noise, but they way it was implemented majorly screwed up the RF characteristics causing reflections on the line which were much worse than the noise it was trying to fix. This can also happen with optical, although because the common implementation is so poor to begin with noise on the signal is rarely noticed. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=84742 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 2.0
rgro;653136 Wrote: > You probably need to reinstall the toolbox. I had to after the Touch > 7.6.1 firmware update. Soundcheck does warn that the toolbox may not > survive firmware updates. If you're using dynaudiorules' mods, you > should also check to see if those survived. > > And, somebody can correct me if I'm wrong, but I also believe if you > ever need to do a factory reset, it will wipe out both of these mods. The Touch has a "union filesystem". The firmware from Logitech is the base filesystem. Any changes made to any files or new files are put in a "modified" filesystem. What you see is the union of those two filesystems. A factory reset deletes everfything in the "modified" filesystem, thus you see the "as delivered" firmware. A firmware update deletes some of whats in the modified filesystem, thus some changes will go away and some will stay. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=84742 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
Jeff Flowerday;651240 Wrote: > Just a wild guess but maybe the jive_alsa process hasn't spawned yet > when the rcS script fires? This is true, the last thing in the rcS starts squeezeplay (the big program that does all the SB stuff). Squeezeplay starts the jive_alsa processes and then goes into its own loop, it never retruns back to the rcS script. So in order to get the jive_alsa process number you wave to check after squeezeplay has started. The easiest way to do this is to put your commands in a separate file which has a sleep at the beginning, here is what it might look like with a name such as SB_priority.sh: sleep 100 chrt... ... ... In the end of the rcS file, just before the squeezeplay command type: SB_priority.sh & The '&' at the end is important, this causes the script to run as a separate process in parallel to the squeezeplay process. That should do it. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
Normally the Touch has two jive_alsa processes, one for the sound effects (beeps, blips etc), and one for the music outputs, it handles both analog outs and digital outs. The "music" one will have a -d default in its command line, the effects one has a -d plughw:2,0. When you use TT2.0 the effects process is disabled and both the effects and music are routed to the same jive_alsa process (which is why its imperitive to turn off sound effects when using TT2.0). You do have to be careful with the process numbers, they can change. Up this thread a ways someone posted a varient of the script which determines on the fly what the process numbers are, you just need to add detection of the correct jive_alsa thread to this script. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
I've been playing with jive_alsa as well as spdif, I find I like jive_alsa at 55 and spdif at 51. It, much more "lively", more open, far more subtle than the default, without the harshness and keeps the perceived frequency balance intact. Of course everybody is going to like something different. This sort of "gelled" for me, like when you are adjusting VTA and get it right. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 2.0
Whats going on here is that in the Touch are two DACs, one connected to the RCA jacks (the "hi-fi" DAC) and one connected to the internal speaker which produces the bleeps and blurps ect, (the "effects" DAC). Normally there is a separate jive-alsa process just for the effects channel, this is a rather big waste of resources and interrupts etc, so TT2.0 disables that separate process, but that means there is no place for the effects noises to go, so they wind up getting routed to the main system along with the music. The problem is that the effects noises are at a lower sample rate so when they get played on the main channel they cause all kinds of crackling etc because they are at a different sample rate than the music. As has been mentioned the solution is just to turn off all effect sounds. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=84742 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
I got this up and running on my main system this weekend, I set spdif to 59 and was shocked, it was so harsh, and the bass seemed to almost disappear. I thought I had broken the DAC. Then I set it back down to 44 and the bass was back and the harshness went away. I was not prepared for the amount of change this caused. 44 is an interesting number because the jive-alsa thread has a default priority of 45. So by setting spdif to 45 or above you are now putting it at a higher priority than the jive_alsa thread, which is what actually might be causing the sound changes. It might be an interesting test to try running jive_alsa at a higher priority (say 55) and then try running s/pdif in the 50 to 60 range. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Very Long S/PDIF Cable to DAC?
Mnyb;649419 Wrote: > You can use indoor 75 ohm catv cable, it is a little stiff but low > damping, or other 75 ohm video coax. RG 6 ? Or other options . > > 10 meter should not be a problem unless something is broken, spdiff is > very undemanding compared to video, wich 75 ohm cable is also used for > ( besides industrial feildbusses etc ). I'll second this. I've done 100ft with off the shelf RG-6 with standard F connectors on the ends. I also picked up two F-RCA adapters so it would plug into normal RCA jacks. (why oh why didn't Philips use F connectors for S/PDIF in the first place, they work well, are cheap, and readily available) John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89561 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 2.0
Tom186;649217 Wrote: > Most certainly. Is it going to run with Async USB at all? No luck with > my current setup (an Arcam rDAC with USB), with any of the current USB > Audio solutions out of the SBT, including the one in soundcheck's tools > and the one John Swenson proposed. > > Best, Thomas I've been working on the USB DAC issued again this week, its getting better but still not perfect. There are a couple different issues, most of which relate to 24 bitness not async per se. One issue is that most USB DACs that use 24 bit use a different 24 bit format than the The Touch uses, this requires you use the PLUG interface, which adds an extra layer and messes things up. I have a custom jive_alsa which supports the other 24 bit format, which allows you to not use PLUG, this helps a lot but isn't perfect, I still get occasional clicks and pops. I have been playing a lot with thread priorities and can significantly change the pattern of clicks, but so far I have not been able to get rid of them entirely. Another thing that helps is the udev script that was posted here a while back, this makes sure that the USB DAC winds up with the same card number. Unfortunately it doesn't go far enough and sometimes it can still wind up in conflict with existing devices (S/PDIF or one of the two DACs). This can cause some issues as well. I need to get some time to revamp the udev script so it handles all sound devices and makes sure all devices are always unique and the same name. I don't think its hopeless, sometime we will figure out how to get modern USB DACs to work well with the Touch. I don't think its a driver issue, the existing one does seem to handle async properly, it seems more like its interrupt priority/latency issues more than anything else. Its actually very dependant on what else is going on inside the Touch, so all the stuff Soundcheck has been working on helps a lot. I just think I have not found the right combination of priorities and buffer sizes etc. The interesting thing is that when I go back to 16 bit mode everything works fine, no clicks or pops, but the moment I turn on 24 bit mode, problems start happening. IF your DAC supports 16 bit, and you are listening to 16 bit music, and you have the volume set to 100 then there should not be a problem going to 16 bit mode, BUT a lot of modern DACs will not run in 16 bit mode, which means I need to get this figured out. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=84742 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 2.0
Hi Klaus, on the network front, the multilayer protocol "stack" is partly to blame here. Each layer has at least one buffer (usually more) and the data gets pushed from one layer to the next, getting copied from one buffer to the next, different code blocks get swapped in and out of cache, all this causes lots of processor use and memory accesses, WAY more than is necessary to actually get the data from the wire to the audio threads. Using UDP rather than TCP/IP helps quite a bit. I've done tests on other embedded hardware where I have switched between TCP and UDP and the UDP winds up sounding significantly better. But I don't think that change is going to happen with the squeezebox system. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=84742 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] Touch: Upsampling with sox?
What format is the file? Are you running an external SBS or are you plugging in a USB drive to the Touch? Dou you get that message from the Touch or from the server? The Touch can definately play 88.2 files, and nothing in either Soundcheck's nor Dynaudiorules mods changes anything related to sample rate. I'm having a hard time figuring out how you would get that message. The file might be corrupt and give the wrong sample rate, but that should get caught by the server and converted to a supported sample rate (IF running an external SBS). The only other possibility I can think of is if you have SOX setup to change the sample rate to something the Touch doesn't support. I would check the server log and the Touch log to see if there are any errors reported when trying to play this file. The server log is listed in the server web interface somewhere under the advanced setup menu. On the Touch you need to SSH into the Touch (use PuTTY on windows) and cd /var/log less messages When done type q then exit to get out. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=80903 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
kiat;647695 Wrote: > Thanks John. That's sad, looking at the change log. It would be nice > to use the latest version. > > Another question. What if we don't use the latest version. Is there > any later version of ALSA usable in Touch environment? No, the version in the Touch is the last one before the change. The drivers in the Touch were derrived from the generic drivers from the chip manufacturers, they probably have ones now that will work with the latest ALSA but they would have to have similar modifications made. About a year and a half ago I worked on an upgraded USB driver since there had been a LOT of improvements. Logitech did not want to incorporate it into the system at that time. Its being resurected now, I'm working on it with Triode, but the code from a year and a half ago will not compile on the current system, its going to take some time to debug all that. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
john4456;648083 Wrote: > Dynobot, thanks for sharing your mod. > > I'm using the USB audio out on the Touchis the mod likely to have > an effect in this scenario (or does it only impact spdif)? > > Thanks Yes it does, I've been playing with this all weekend. In the instructions run the command that lists the interrupts, search for the USB one. Use this interrupt number instead of the one for spdif in the rest of the instructions. Play around with the priority number you give this IRQ, it can significantly change the sound. I've been playing around with these techniques trying to get hi res 24 bit playback to work and its almost there with the HRT. It still once in a while, but it's much better than it was. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
Re: [SlimDevices: Audiophiles] New Squeezebox Touch Mod
This is something I have been working towards for a long time. Unfortunately its not simple. The current ALSA interface is different than it was when the Touch drivers were written. So in order to use the latest ALSA system the Touch drivers would have to be modified and Logitech is (understandably) very reluctant to do that. I don't see this happening any time soon. To actually answer the question, yes I have tried it, and you get gazillions of errors with the drivers. John S. -- JohnSwenson JohnSwenson's Profile: http://forums.slimdevices.com/member.php?userid=5974 View this thread: http://forums.slimdevices.com/showthread.php?t=89359 ___ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
