Matt, I suspect that you've made a good decision regarding the single heavy gauge dedicated line. As a point of info regarding the 2.8 amp measurement, though, I would by no means assume that an electrician's current meter necessarily (or even probably) has sufficient bandwidth (i.e., is fast enough) to capture the narrow high amplitude current spikes that are likely to be drawn by the amplifiers at times. Especially in the case of Class D (and also Class AB) amplifiers, whose current draw fluctuates widely with the dynamics of the music.
On another note, let me add my second to the many expressions of appreciation that have been made for the time and effort you are putting into performing and writing up this effort, which is and will be of considerable benefit to many.
Best regards,
-- Al |
Thanks, Erik.
Re my previous post, that was a typo on "Boxster," of course.
Regards,
-- Al |
Steve, I'm surprised at your question. What I'm talking about is the fact that depending on the quality of the design digital noise can couple between different parts of the circuitry in a component, via grounds, stray capacitances, etc. Potentially/theoretically resulting in increased jitter at the point of D/A conversion, and/or effects on analog circuitry. Where did you get your EE degree? BSEE Columbia University MSEE Rensselaer Polytechnic Institute + 33 years experience designing and managing design of analog, digital, A/D, and D/A converter circuits for defense electronics. In any event, the bottom line on both of our posts is the same: Implementation is what matters. Regards, -- Al |
Knghifi, Jon, Charles, Guido, thanks very much for your comments. And Erik, thanks for your inputs as well, about both the DAC and the tires.
JoeCasey, yes, I chose the "PDK" automatic. I'm a "purist" about some things to some degree, but not in this case :-)
As you may be aware, btw, the PDK is a seven-speed "dual clutch" automatic, which actually out-accelerates the manual version (at least if the package which includes "launch control" is opted for, which I did not get).
Best regards,
-- Al |
09-11-14: Aplhifi
But even Almarg (or anyone else with such extensive experience) would agree that the new Berkeley converter should at least make possible for DSD64 through the S/PDIF inputs via DoP format and let their customers decide whether Pure DSD is better than DSD converted to 176/24 PCM.... Please correct me if I am wrong. Having no specific familiarity with the tradeoffs that might be involved, I take no position on that. My only point, as I said in the first of my posts dated 9-10-14, is that "quality of implementation is likely to trump the theoretical advantages and disadvantages of any particular design approach." Audiolabyrinth, thanks for the nice words in your post yesterday. Regards, -- Al |
A few minutes after submitting my post just above I received my copy of the October 2014 issue of TAS, containing the Berkeley review and interview Erik referred to earlier. In the interview Michael Ritter of Berkeley addresses in a reasonable amount of detail his rationale for not including a USB interface and DoP capability in the design.
His comments about the lack of USB are very much along the lines of what I said earlier. His comments about the lack of DoP IMO represent a credible, plausible, and well stated design philosophy, especially for a component in this price range. I certainly recognize, however, that similarly credible and plausible cases could be made for opposing philosophies.
Accordingly, I would not categorize the arguments on either side of the coin as being BS, or as being grounds for either rejection or acceptance of a particular candidate. Again, "quality of implementation is likely to trump the theoretical advantages and disadvantages of any particular design approach." But after reading the review and interview, like Erik if I were willing to spend $16K for a DAC I would certainly put the Berkeley on my short list.
Regards,
-- Al
|
"NFB" = "Negative Feedback."
Regards,
-- Al
|
Guido, thanks for pointing out the Thunderbolt capability of the Weiss 202 DAC. That capability, however, appears to be provided by means of an external Thunderbolt-to-Firewire adapter, which would connect to a Firewire port on the rear of the DAC. And based on a quick look at their website I'm not at all certain that they are even marketing such an adapter themselves, or are simply claiming Thunderbolt compatibility by virtue of the fact that Apple sells Thunderbolt-to-Firewire adapters.
Also, I would feel safe in assuming that their motivation in proclaiming Thunderbolt compatibility derives from the fact that Firewire has been superseded by Thunderbolt on Apple computers. (As I understand it Thunderbolt is only available on non-Apple computers and separately purchasable motherboards to a very limited extent at this time, primarily on high-end workstations, in part because it is considerably more expensive than USB3).
As to whether or not connection of a DAC via a Thunderbolt-to-Firewire adapter would provide any inherent advantage in comparison with the direct Firewire connection that would be possible on older Macs and on other computers having Firewire interfaces, I'd expect that to have to be determined empirically. And I'm not sure how or if that determination could be performed without a great many extraneous hardware-dependent variables also being in play.
Those are my thoughts on the subject, anyway. We'll see if Steve or Alex have anything to add. Best regards,
-- Al
|
Steve, the review Jon is referring to, which was written by Steven Stone, begins on page 94 of TAS issue 247 (the "Spring 2015 Buyer's Guide" issue). The Off-Ramp 5 is mentioned on page 98, just before the "Conclusion" section.
Regards,
-- Al
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11-12-14: Ctsooner
I'd recommend for your electrical is to have totally separate and grounded (to earth) circuits for EACH outlet. I did that and it makes a huge difference.
11-12-14: Agear
I too have dedicated lines with with isolated earth (ionic) grounds. Matt, I would advise against this, if I am correctly interpreting that what is being recommended is running outlet grounds to an earth connection other than the one that grounds the service panel. If that is not a correct interpretation of what is being suggested, I'm sure the others will clarify further. See section 1.2 on pages 7 and 8 of this paper by Bill Whitlock of Jensen Transformers, who is a renowned expert on such matters. Note the concluding sentence: "If multiple ground rods are used, Code requires that they all MUST be bonded to the main utility power grounding electrode." Also note the section starting at the bottom of page 3, "myths about earth grounding and wires." As explained in section 1.2 of the paper, having earth connections for the system that are separate from that of the main service panel is a code violation, a shock hazard and a fire hazard in the event of an equipment fault, and a hazard to the equipment (or worse) in the event of a nearby lightning strike. And I'll add that it would also be a potential excuse for your insurance company to not pay in the event of one of these disasters. You may want to start a separate thread on the subject of AC wiring and grounding, which will probably catch the eye of Jea48 (Jim), who is our leading expert here on such matters. Regards, -- Al |
02-19-15: Rhanson739
One drawback to the RAID plan is that while it is redundant unto itself, it doesn't protect you from complete data loss in the event of a house catastrophe. The better solution there is an off-site backup, or a fire safe. I don't do cloud. +1. While RAID 1 protects against failure of one of the two drives in the array, it does not protect against any number of admittedly very unlikely but nonetheless possible scenarios which could destroy or corrupt both drives simultaneously. In addition to a house catastrophe, those would include the power supply for the array going into an overvoltage condition, its controller circuitry going berserk, a severe AC power surge, a computer virus, or a latent design defect in the RAID controller circuitry or its firmware. More than a few user comments I've seen at Newegg.com concerning certain RAID hardware have recounted data loss caused by the latter possibility. Concerning HDD selection, my perception over the years has been that the best choice has generally been a moving target, often varying from one HD generation to the next. It may be helpful before finalizing a selection to review the user comments at Newegg on whatever tentative choice(s) you settle on, while keeping in mind that negative comments tend to be disproportionately represented there. FWIW, in recent years I've had good results with most of the leading brands, including Western Digital, Seagate, Maxtor, and Samsung (in all cases those being non-enterprise consumer-grade drives). Also FWIW, one of my computers, which includes two 640 gB Western Digital internal HDD's, has been running pretty much continuously for approximately the past 6 years. The drives have accumulated about 50,000 hours, with no hint of trouble and with perfect SMART readings. Good luck. Regards, -- Al |
Matt, for a checklist of manufacturers of time coherent speakers (which also implies phase coherence) see this post, and the response in the second post following it. Many of those mentioned will not meet your other criteria, however. Note: "GMA" = Green Mountain Audio. Regards, -- Al |
From a technical standpoint, the paragraphs George quoted from the MSB literature all make sense to me, as opposed to being the kind of marketing techno-babble that seems all too common in audio-related white papers and other literature. An additional theoretical advantage of using sign-magnitude architecture in a ladder DAC implementation, besides the one that is stated, is that the "offset binary" architecture they seem to be implying is used in other ladder DAC implementations will cause all of the bits to toggle (i.e., to change from 1 to 0 or 0 to 1) **simultaneously** at or near the critical 0 volt crossing. Which in turn can contribute to noise issues at that crossing, and hence degrade resolution at very low signal levels. With sign-magnitude, only a few bits would toggle at or near the 0 crossing. I'd have to add, also, that providing 20.8 to 28.5 effective bits using a ladder approach, depending on model, as shown here, and not just for the D/A circuit but for the unit as a whole, is quite an amazing achievement. Primarily because of the incredibly tight +/- tolerances that would have to be met by the resistors in the ladder. Which leads me to the one concern that is raised in my mind by a quick look at their writeups. Their focus seems to be on achieving the best possible performance at very low signal levels, i.e., near the 0 crossing. Which arguably makes a lot of sense. But consistent with that, I see in the figure I linked to that their measurements comparing their results with those of delta-sigma DACs were taken with the test signal at -90 db, a very low level. So what I wonder is how well the much superior performance of their approach that is depicted in the graph would hold up at moderate to high signal levels, if in fact it would hold up at all. Just some thoughts to keep in mind. Best regards, -- Al |
Guido, I think he's saying that the "infinite slope" crossovers prevent the metal cones from being exposed to frequencies they can't handle without breakup and/or ringing. While a more conventional lower order crossover would provide a much lesser degree of filtering of frequencies that are outside of the range the drivers can handle in an optimal manner.
I have no specific knowledge of Joseph speakers, but that is how I would interpret the statement.
Best regards,
-- Al
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08-14-15: Paul79
The Totaldac Twelve is not cheap to make, no matter where he is located... 600 .01% foil resistors, at $20 a piece retail, it is easy to see where the bulk of the costs are. Add to that SOTA Parts, inputs, outputs, power supplies, cases, boards.... No question about it. And I would add that I wouldn't be surprised if in addition he has to go through a process of screening the resistors further, selecting some and rejecting others. While 0.01% is an exceptionally tight tolerance for a resistor, it does not even come close to supporting 24 bit performance in a ladder configuration. Presumably TotalDAC's use of 6 ladders per channel in the Twelve helps considerably, but consider that 24 bits corresponds to 2^24 = 16,777,216 possible states, which in turn means that the least significant of the 24 bits controls 1/(16,777,216 - 1) = 0.000006%(!) of the full scale (maximum) output of the DAC. As I said in an earlier post in this thread, providing anything close to 24 bit performance via a ladder approach, as TotalDAC and MSB Technology apparently do, is an amazing (and invariably expensive) achievement. Regards, -- Al |
Matt, congratulations on reaching this major milestone in your journey. Enjoy! A point to keep in mind regarding your forthcoming audition of the Aesthetix Romulus Signature: It will be marginal, at best, in terms of its compatibility with the relatively low input impedance of your Burmester preamp (11K balanced/16.5K unbalanced, and I assume you will want to use the balanced inputs). The manual for the non-Signature Romulus states: Recommended load impedance: >20Kohm balanced; >l0Kohm single-ended And I would emphasize the ">"; presumably loading that is not significantly greater than those values would represent at least a small degree of compromise. The corresponding numbers for the Signature version, which don't appear to be published, would probably be somewhat lower (better) than that, due to the output coupling capacitor upgrades it incorporates. But I suspect it would still be just marginally compatible, with the most likely consequence being some amount of rolloff in the bottom octave. So be sure that your assessment includes particular focus on material having significant content in the very deep bass region. A number of other Aesthetix products are similarly marginal with respect to low impedance loads, which is why I suspected there might be an issue in this case and checked into these numbers. Best regards, -- Al |
09-27-15: Mattnshilp
The point about impedance mismatches is a very interesting one and makes me think that the Allnic D5000 may have a similar issue. Matt, no, I doubt that the relatively low input impedance of your Burmester preamp had anything to do with your impressions of the D-5000's sonics. In contrast to the capacitively coupled outputs of the Aesthetix DAC, which can be expected to cause a large rise in output impedance in the bottom octave or two, the Allnic has transformer coupled outputs. And consistent with that it is described in its manual as having an output impedance that is a "constant" (as a function of frequency) 150 ohms. That kind of output characteristic shouldn't result in any issues driving the input impedance of your preamp, or, for that matter, driving longish cables. Of course transformers can have issues of their own, but given the apparent quality of this unit, and the statements made in the manual about Allnic's expertise in designing and manufacturing their own transformers, I suspect that the reason(s) for your less than glowing impressions of it are related to something else. Best regards, -- Al |
10-13-15: Tbg
I have no idea what the last post is on this thread as I am tired of always having to spend time going through all the pages. As an alternative to Mike's suggestion, near the bottom of whatever page opens, just above the "post your response" box, just click on the ">>" if it is present, or on the right-most page number if it is not present. That will take you directly to the latest page, without having to go through the previous pages. Regards, -- Al |
10-17-15: Grateful
As such, what I have found is...
The more you can afford (all else being equal) upstream gets you the best purity. In other words, your creme de la creme should be (in decreasing order) source, pre, mains, speakers - of course you can add cables in-between each one and voltage-cleanup.
I might add, this idea does make physical sense, as as the signal strength is the smallest (upstream) improvements and/or pollution is only amplified downstream. Thanks for your input, Grateful. I certainly don't question your experience and your observations, but I would respectfully disagree with the last sentence in the quote. Notwithstanding the fact that a lot of audiophiles believe similarly. While it is true that "pollution" introduced upstream is in many cases amplified more than pollution introduced downstream (but not always; see the next paragraph), the same is true with respect to the signal. And what matters is the relation between the two, not their individual magnitudes. Furthermore, while the signal level at the source may in many cases be smaller than further downstream, that will often not be true in the case of digital sources in particular, with the preamp's output signal in such cases often and probably usually being at a lower level than its input signal. And even in the case of vinyl playback, where of course extremely low level signals are present at the output of the source, I would comment that from a technical standpoint amplification of very low level signals is not by any means necessarily more problematical than the task of a power amplifier, or the task of a speaker in converting the large amounts of power it receives into sound. The technical challenges in each case are certainly different, but I don't think any conclusions about which are likely to be more critical are supported by rationale that is based on the magnitudes of the signals that are involved. Contrary, as I say, to what a lot of audiophiles seem to believe. Just my $.02. Regards, -- Al |
10-25-15: Dgarretson
Georgelofi, do you recall if the Theta Gen Va is an R2R? I can't find a spec, but there are some on the forums who believe so. I have one in storage that I haven't listened to in years. If it's an R2R then I'll dust it off for comparison to K-01X. Dave, I found multiple indications on the web that the Gen Va used the Burr-Brown PCM63, which was a 20-bit predecessor of the 24-bit PCM1704 discussed above. As can be seen in the PCM63 datasheet it was indeed an R-2R, and the theory of operation section on page 5 reads very similarly to the corresponding section on page 6 of the PCM1704 datasheet. The focus on performance around the zero crossing in both descriptions is reminiscent of the philosophy of the very expensive MSB DACs. Matt, thanks for the light-hearted interlude, which I read and enjoyed. Best regards, -- Al |
Bill (Grannyring), regarding your comments about humidity being a significant variable, here is some remarkable confirmation, from this current thread: 06-17-14: Georgelofi
... this [humidity] is greatly overlooked. Take any hybrid ESL speaker for instance, they are tuned for a certain sound by the factory. The esl panel on a day when the humidity is 70%-100% can loose as much as -3db efficiency, making the balance sound bass heavy as the dynamic bass driver has not lost this -3db.
I have measured it on my new panels, dry day 5kv bias charge and on a humid day 3kv-4kv bias charge, I tend not to do any evaluations or serious listening on days when the humidity is above 70%. Or if I have to, I can with some trouble turn down the gain of the bass electronics by the appropriate amount so all is balanced again. Best regards, -- Al |
09-10-14: Erikminer
In the TAS review there is an interview with the designer (s) of the Dac. They feel that building a USB input with the associated hardware degrades the SQ of the unit. Could that be BS since they sell the pricey outboard unit? Who knows but I'll give them the benefit of knowing what they're doing. It's not BS, as introducing signals having significant high frequency content (e.g., fast risetimes and falltimes), and that likely have significant amounts of noise riding on them, can potentially cause issues at supposedly unrelated circuit points elsewhere in the design. On the other hand, given enough attention to those kinds of possibilities in the design and development process, I see no reason why they couldn't be overcome. And S/PDIF has tradeoffs of its own. As usual, IMO, quality of implementation is likely to trump the theoretical advantages and disadvantages of any particular design approach. A brief off-topic comment: Matt, congrats on the new Porsche. I (and my wife) have been thrilled with our 2014 Cayman S, which as you no doubt realize is the hard-topped counterpart of your Boxter. Be aware, though, that its stock high performance tires cannot handle snow or ice (assuming they are the same tires as on the Cayman S), and MUST be changed to winter tires if you want to drive on those surfaces. Regards, -- Al |
07-21-15: Georgelofi
Too bad the manufacturers are trying to phase out R2R Multibit dac's, they are way harder and more expensive for them to make, and they then charge up to 10 x the price for them. And much more so for hi rez, as accurate reproduction of 24 bit data requires 256 times greater accuracy in the resistor ladder than the already extremely tight accuracy that is required for 16 bit data. As I mentioned earlier, MSB's accomplishment in applying ladder architecture to hi rez (as well as Totaldac's, as pointed out by Agear) simply amazes me. As I also mentioned, though, MSB's use of a sign-magnitude data format will tend to make measurements of low level resolution look as good as possible, relative to competing approaches, when the test signal is extremely low in level (as it was in their published test results). As always, the proof is in the listening. Regards, -- Al |
Ctsooner, based on your latest post it sounds probable that the electrician simply installed what are commonly referred to as dedicated lines for each of the outlets, which is fine. I had been interpreting your earlier statement, and Agear's as well, to amount to what is depicted in the figure at the top of page 8 of the reference I provided. That is done by some audiophiles, but would entail all of the risks I described. Matt, that all sounds fine also. And +1 to Steve's suggestions. Regarding your questions, my instinct would be to connect the monitor to the same outlet as the computer, and to NOT connect it to the outlet powering the turntable motor. I'm envisioning the possibility that if they were on the same outlet some amount of RFI generated by the monitor might find its way to the motor, and radiate from there to the phono signal wiring. My instinct/guess is the same as yours, though, regarding there probably being no need to have the trickle charger and the phono stage on the same circuit. Hummmmmm...... Clever double entendre :-) Regards, -- Al |
02-02-15: Ctsooner
I can't wait to Steve's DAC up and running in my system, I could use the built in super drive but have been told that a remote usb drive is best as you can buy a much better quality one. Plexor was the name Steve game me, but when I went to their site, they seem to only be for window's systems. CTsooner, in my experience with external optical drives (which as a disclaimer I'll mention is limited to Windows and Linux operating systems, and to non-audio applications), you would get much better reliability and consistency of performance across diverse media, as well as greater speed if you choose to use it, if instead of purchasing an external USB optical drive as such you were to purchase a standard size internal drive intended for desktop computers, and mount it in a **good quality** powered external enclosure which provides a SATA interface internally and a USB interface externally. I note that the one optical drive Plextor currently offers is that kind of internal drive. Having essentially no Mac experience I can't say this for sure, but my suspicion is that even though it is described as being for Windows operating systems it would work fine with a 2009 Mac Mini when interfaced via USB through an external enclosure. You can find suitable external enclosures at Newegg.com, which is my "go to" place for computer parts and accessories. Here is one such enclosure that looks promising, although I have no experience with this or other currently listed models. If you click on the "see questions and answers" link to the right of and slightly below the photos, you'll find a couple of people mentioning that they are using it successfully with Mac Minis. Newegg also sells the Plextor drive, btw, for $30 in OEM form or $37 in full retail form. I presume the OEM version comes without box, manual, and mounting screws, although mounting screws are supplied with external enclosures in some cases, apparently including the one I linked to. Regards, -- Al |
06-20-15: Mattnshilp
...he suggested I add a sub to the room connected in series from the amps to speakers (high output terminals). He specifically said not to pull the signal low level from the pre-amp. He said to put the sub behind the listening position in the back of the room to balance the pressure in the room, adjusting the output so its not even audible but enough to offset the main speakers output pressurization. Matt, As far as placing a sub in the rear of the room is concerned, it seems worth trying but the concern would be that arrival time differences at the listening position, between the mains and the sub, may compromise coherence unacceptably. And if you set the sub's volume low enough to prevent that, you may find that it is ineffective with respect to its intended purpose. Also, regarding the reference to "series" connection (which strictly speaking is a misnomer, as from a technical standpoint the sub and mains would be connected in parallel): Assuming the amps are in the front of the room, as you most likely realize it would be best to connect the amps directly to the main speakers, and to run separate connections from either the main speakers or the amp to the sub. Also, as you probably realize, if the sub is a powered one the cables to it would not have to be heavy gauge (even 22 gauge or smaller would work fine) since powered subs have high input impedances (e.g. 10K or more), and hence draw very little current. Regarding the BagEnd product, if you haven't already seen it you will want to check out Kal Rubinson's review. Also, you may want to look into the Spatial Black Hole. I have no experience with or particular knowledge of either product. Good luck. Regards, -- Al |
Avantgarde - Duo Grosso (I know, a horn in a small room. But I have spoken to many people who think they would fit perfectly in my room and I have always enjoyed the way they sound; albeit a bit romantic. I'm also not in love with the fact that they are powered subs and that I would probably benefit from a tubed amp instead of my uber solid state amp that I love so much). Matt, an additional point to keep in mind, certainly with respect to the Avantgarde speakers and perhaps with some of the others that may have particularly high efficiency, is that the high power capability and high sensitivity provided by your Burmester amps would probably result in your using the volume control on the preamp very close to the bottom of its range. Which depending on the design of the preamp may have a number of adverse consequences. Best regards, -- Al |
Gentlemen, be aware that this Saturday a very large asteroid is projected to pass us by at a distance not much greater than the distance of the moon, at a speed of about 78,000 miles per hour. If the calculations are wrong for some reason there may be a good deal less time left for Matt to complete his evaluations than we think.
-- Al :-)
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I basically agree with Guido, but I would modify his statement as follows (changes are in brackets): ... a warm sound is a tone that has a [lower] midrange preference, and perhaps a touch of pillowiness in the mid and mid lower bass. A rich sound for me has significant exposure of harmonics throughout the range, [particularly lower order even harmonics], from low bass to higher treble, without distortions. Of course, usage of these terms among different audiophiles varies widely, and sometimes they are used interchangeably. But strictly speaking, IMO, "warmth" relates to frequency response, and "rich" relates to harmonic balance. Regards, -- Al |
This paper is presented at the Brinkmann site and seems to amount almost to a point-by-point rebuttal of the concerns JA expressed. And in doing so I note that the paper goes considerably beyond what is said in the manufacturer's comment section of the Stereophile issue which presents the review. IMO most of what is said in the paper is persuasive. The only issue I would take with it, at least in the case of a DAC, is in relation to the statement that "over-damping (i.e., filtering) the incoming power might result in better measured values, but often kills the life in the music." While lightening up on the filtering of incoming power might in some DAC designs result in sound that is subjectively preferable, it certainly won't make the sound more true to the source IMO. But in any event the numbers cited in the paper and in JA's measurements are suggestive of that not being a significant issue in this case. Regarding JA's measurements more generally, they are IMO of inestimable value in identifying possible or likely mismatches or incompatibilities between components, and often in diagnosing sonic issues as well. I couldn't begin to count the number of times I and many others have referred to his measurements in responding to questions and issues that are posted here. In that sense JA performs a uniquely valuable service to the audiophile community, especially given the thoroughness of his measurements relative to those that are provided in some other publications, and given the much higher probability that he has measured a component that is being considered, compared to other publications that provide measurements. And every now and then some of his measurements raise legitimate concern about the quality of the engineering that went into the design of a product. Beyond those benefits, though, I would certainly agree that they generally provide little if any insight into how a component will sound. Best regards, -- Al |
+1 Charles. And Steve, thanks for the very informative and excellent post!
Regards,
-- Al
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Hi Hal, I would begin my answer to your question just above by noting the second sentence in the Wikipedia writeup on delta-sigma which you linked to:
It [delta-sigma] is also used to convert high bit-count, low-frequency digital signals
into lower bit-count, higher-frequency digital signals as part of the
process to convert digital signals into analog as part of a digital to analog converter (DAC).
So in that sense, yes, the original samples are "thrown away." However that says nothing at all about the degree to which musical information is lost in the process. And regarding my opinion, I would say that as is usually the case in audio how well a design approach is implemented can be expected to be more important than which approach is chosen. Best regards, -- Al |
Hi Hal, I would put it that delta-sigma is a **less direct** way of accomplishing D/A conversion than R2R. However, the results of BOTH approaches are approximations, in the sense that what comes out does not correspond to what goes in to an infinitely precise degree. And of course the same goes for everything else in the recording and reproduction chain, both theoretically and in terms of the various practical implementations, from the recording microphones to the speakers in our listening rooms and everything in between. Even the Nyquist-Shannon sampling theorem, which underlies A/D conversion, D/A conversion, and essentially all digital signal processing, is an approximation in any real world implementation. (For example, note in that Wikipedia page the various equations involving integrals or summations that are taken from minus infinity to plus infinity). So in that sense **everything** in the recording and reproduction chain is lossy, to some degree. Which of course tells us nothing whatsoever about which design approaches and which practical implementations of those approaches are to be preferred. But my perception and belief, as evidenced in part by various posts from highly experienced audiophiles in this very thread, has been that depending on the specific implementation both approaches can provide results which are satisfactory to many, at pretty much all price points that can be considered to be audiophile-oriented. It’s as simple as that, IMO. Best regards, -- Al |
@gdhal
Hi Hal,
No, my statement was not intended to imply that musical information is lost when data having more bits per sample and a lower sample rate is converted to fewer bits per sample at a higher sample rate. From a theoretical standpoint nothing will be lost if the increase in sample rate is sufficient to compensate for the decrease in bits per sample.
And while I do not have detailed familiarity with the specifics of modern delta-sigma DAC designs, user reports of good results with high end delta-sigma designs, published measurements, comments by some who have such familiarity (such as the comments provided by Steve and Shadorne yesterday), as well as my general belief (shared by many others here) that in audio how well a chosen design approach is implemented is usually more important than which approach is chosen, all lead me to believe that "throwing away the original samples" (as referred to by Schiit) is a non-issue in modern high end designs.
Best regards,
-- Al
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The writeup by Peter Mitchell which Shadorne provided (thanks!) is excellent. But to be precise, the architecture it describes for a "conventional" DAC, in which "the resistors are supposed to be scaled in exact 2:1 ratios,"
is not R2R. In an R2R ladder only two unique resistor values are required, rather than a different resistor value for each of the 16 (or however many) bits in each sample. R2R architecture is explained pretty well in the Wikipedia writeup on "Resistor Ladders."However Mr. Mitchell's basic points about the issues inherent in the "conventional" architecture he describes are nevertheless applicable to R2R as well, as can be seen in the Wikipedia writeup. Regards, -- Al
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Matt, I enjoyed watching all four of the videos, including the one that just went up with the live piano playing. And on my computers, at least, the somewhat low volume and low lighting weren’t any problem at all.
In fact to me the only slight downside of the low lighting, as a collector of antique radios, was that I couldn’t tell for sure if the cathedral radio on the left side was an original or a modern repro. I’m guessing it was a modern repro modelled a bit loosely on the 1930s Philco 90 :-)
Best regards,
-- Al
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I can only speculate that maybe the circuits in the Ethernet receiver
react differently to slower edge-rates and poor signal integrity. Maybe
the setup timing margins are smaller. This might cause the propagated
signals to have more jitter.
To add to Steve's comment, it seems very conceivable to me that in some and perhaps many less than perfect designs a small but potentially significant fraction of the signal energy received by the Ethernet interface may find its way **around** that circuitry (via grounds, power supplies, parasitic capacitances, etc.), and end up introducing noise onto the signals which control timing of D/A conversion. All of the energy of a signal doesn't necessarily follow only its intended/ideal pathway. Especially when the signal contains spectral components at very high frequencies, as in the case of Ethernet. Regards, -- Al |
