Bottlehead Forum
Other Gear => Digital => Topic started by: glynnw on October 27, 2011, 07:01:11 AM
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My preference is that you come out with a great DAC, not a cheap DAC. Do not need USB, though it is a nice option. Currently using a Wavelength Brick with Mortar power supply (retail $2K) and recently bought and then sold (preferred the Brick) Ayre QB-9 (retail $3K). I have no problem paying over a grand for something top drawer. Just my 2 cents.
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being able to handle 24/192 would be nice.
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You guys and I are on the same track. I will try to keep the basic version around $1000 if possible (really can't say for sure yet), and there will be upgrade options like USB input, multiple power supplies and a couple choices of sample rate display. And it will do 24/192. The project got slowed down a bit again with RMAF, but we are working on ramping it back up ASAP.
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Great. I really liked the Ayre having a sample rate display and wish my Brick did.
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I bet it doesn't use Nixie tubes...
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If no USB, how do I get music from my computer to the dac?
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You would provide an SPDIF output in the form of a USB to SPDIF converter of some kind like an M2tech Hiface, or via a sound card with an SPDIF output.
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I don't care out a USB, but it would be nice to have more than one SPDIF input. for multiple digital sources.
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The plan is to offer the kit with an SPDIF coax input. There will be a spot on the DAC board for an optional USB input daughterboard. Perhaps the builder could repurpose that as another SPDIF input instead, I'm not sure.
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I bet it doesn't use Nixie tubes...
Is that to say that the Bottlehead DAC uses Nixies? COOOOOL!
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We are discussing having a card edge slot that will allow one to have an assortment of display options. It might be possible to do a Nixie display.
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USB is important to Me...John
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Nixie tubes seem like a cool idea. I'd also like to retain use of a USB connection. USB is improving and I think may be "the " platform for a host of improvements. Great to being hearing some details about the BH-DAC.
Just a thought, lets have a contest to name the new DAC.................??????????????
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John, as soon as we go for USB there will come a new and better way of connecting to a DAC. Digital changes way too quickly for me. I, too, think that USB will be the best thing for the DAC. I am supposing I will end up with some sort of music server and a computer for my stereo.
That just sounds wrong, a computer for my stereo. Well, I can change.
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My question about a USB input is will it be a USB 1.0 or USB 2.0 option. I really doubt that Doc wants to get into the business of writing device drivers but without one I believe a USB 1.0 connection will be limited to 16 bits / 48 kHz. That's fine for CD quality music, but it won't support high resolution data.
For me a second SPDIF input would be preferable to a USB 1.0 input. If your only option is a USB output there are several good USB to SPDIF converters on the market now. Just be sure that the one you choose supports the data resolution(s) that you require.
Ken
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That just sounds wrong, a computer for my stereo. Well, I can change.
I thought the same way. Not anymore though. Now my all analog, all BH system has a non-BH computer and non-BH DAC.
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Might as well go for USB 3.0 now. That seems to be the latest.
My preference, though, would be fiber-optic. I use a TOSLINK from my Squeezebox Touch to my DACMagic. I like the DAC in the DACmagic better but the sueezebox touch is just plain awesome as a way to serve music off of your PC. You can even control everything from an iPhone app.
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Yes! Toslink.
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You guys know TOSLINK can't do 192K, yes? I'm not agin' it, it's what I'm using at the moment. But it will hold you back if you want to buy some 192K files.
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Dan,
Didn't you post some of the proposed specs here somewhere? Could you link that in this thread?
Thanks!
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I am waiting for Doc also, and i agree with everybody about price , something TOP drawer etc. and it sounds like Doc understands the whole very difficult, ever changing dac tech world... I dont envy him with this mammoth project.. though I have great faith in him and Paul, the only thing I also want like everyone (including Doc and Paul) is that the final product has reasonable upgrade ability... future proof is impossible of course, but it would be nice if it doesnt go into obsolescence in a year or so....thats why i think THIS project is worth the wait and so very important, it is worth getting it "right",,, due to the inherent flexible nature of DIY and the possibility of designing and building a dac that could be competing with dacs costing MUCH more on the commercial market and this one having more flexibility.... This one should be a "must have"...... so i agree... putting out "cheapie Dac" ONLY (important for some customers however) would be a waste of time.....
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IIRC, the software work is being done by John Swenson.
The first DAC I bought guaranteed upgradability. It was an Audio Alchemy DDE (Digital Decoding Engine) V1.0. It had an I2S port that was that upgrade path... then the DDE V2.0 came out and the I2S port was never used. Moral of the story, there is no future proof DAC.
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I'm lost now as to how the clocking will be done. IIRC the original idea was to reclock the sound card with the DACs clock but that changed, right? What's the current plan ?
Thanks,
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Grainger, perfectly said... it is absurd for a company to talk about "future proof" unless they have ABSOLUTE control of the entire industry, technology and market... and even then, corruption would rear its ugly head and they would ensure that we STILL have to chip and change our gear often enough to garner them substantial profits..... I went through 2 in a row of the very best AV amplifier/processors that Denon had to offer and quite expensive (for me) also i might add.. and they were said to have "future proof" ability, and though they still, to this day, have some of the latest stuff within and can play extremely well cinematically.... However, they dont and CANNOT have the "latest" technology in them anymore...They should just say "future proof for 2 years" ..... The funny thing about Audio Alchemy (unlike Denon) is they might think the product was future proof, however they didnt stop to think that maybe the COMPANY wasnt future proof...........
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Dialogue between marketing and engineers:
Marketing: "We're going to sell this by calling it 'future proof'".
Engineers: "You can't say that! You have no idea what will happen in the future."
Marketing: "True, but we can always say things developed beyond our control and we did our best"
Engineers: "Then you are simply lying!"
Marketing: "No, we'll sell more units and can pay for you to develop the next 'future proof' product".
Engineers: "I don't like this."
Marketing: "But it pays well."
Douglas Adams said that the Marketing Managers at the Sirius Cybernetics Corporation were the first against the wall when the revolution came.
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I think this is all very funny. We make a decent living selling vacuum tube amplifiers and reel to reel tapes. I used to buy both by the gross for pennies on the dollar because they were not future proof. Then they both were "rediscovered". The thing that is not future proof is a person's attention span.
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Well, there it is... TRUER words have never been spoken....and that ends our "future proof" comedy with an exclamation point! It is just nice to know, we can be smarter than the mass market industry.... And there are people like Doc and Paul etc. that are willing to provide a "way out" of the "corporate audio" mess that I, at least , had no choice but to have lived in before.... Thank you guys!
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I'd like a DAC that also functions as a simple preamp (say three analog inputs -Tuner, Phono, normal CD player ) and a volume control.
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I want a DAC that can double as an indoor grill to cook t-bones medium rare w/o burning the edges. It must also be good to the environment by drawing less than 100 mA to help reduce its carbon footprint!
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Sounds like you are just wishing for a bigger DAC!.... :)
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Well, we could make it two inches longer. Pulse stretcher?
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OK, I'm not trying to cause trouble by saying what I'd like in the new DAC - but one more thing did just occur to me that is very important for anyone wishing to go directly into amps, as I now do through a Bent TVC. An output voltage of at least 2 V - preferably more. The Brick and Ayre are at 2 - the Wavelength Cosecant (which I am hoping yours will be at least the equivalent of) puts out 4. Pretty cheeky of me to ask for something that makes the Foreplay less necessary, I know, but I am willing to place a deposit whenever you are ready.
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Dan,
Regarding 24/192 toslink -- I jusst got a 24/192 capable toslink recieverr chip as an upgrade to my AudioGD NFB-2 dac -- that and a new daughterboard with a Wolfson wm-8805 spdif reciver chip so the entire dac now is 24/192 capable except the usb input. This one will be fed from an Aurality PK-100 music server and will serve tunes either from an attached usb HD, or perhaps over the ethernet connection to files stored on a NAS -- whichever ends up sounding better.
Also, will there be a BNC option in place of the RCA? If I end up using one of your dacs in my main rig, it will be using a modified mac mini into an Audiophilleo AP2 usb -> spdif converter, which is best used with a direct BNC coupling -- no digital cable required.
If you're interested, I can look into the 24/192 toslink reciver module -- up until a couple weeks ago I thought that the current recievers were only 24/96 capable (though there waere some toshiba recievers that were discontinued some time back that could do 24/192 -- though I believe they were only select parts that tested at the higher rate, not a different design.
Unfortunately, most of the sources -- like the various apple products, are limited to 24/96 on the output side with optical, so that's still a limitation.
My Auraliti should be here any day now, so will soon be giving it a try using a Black Cat Veloce spdif cable.
I still prefer direct usb, but to do that right is somewhat of a large engineering undertaking as the off-the-shelf solutions -- whether async or isosync are still a good deal behind the SOTA usb implementations, which I know is the scale of project that BH probably doesn't want to get involved with, so in that case, I think the spdif is probably the best choice. Not all usb and not all spdif are equal, and good examples of each can easily outperform average examples of the other.
The direct connect Audiophilleo (especially after cryoing and an external 5v supply for the usb port) is pretty amazing.
Very much looking forward to what will come of all this, especially after my initial taste testing at RMAF.
-- Jim
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Doc,
I would like a "standby" mode where the DAC portion can be left on while the analog output is turned off. In my experience tubes seem to warm up a lot quicker than digital. A "standby" mode would help balance tube life and power usage while shortening the time required to warm up.
Andrew
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I would like a "standby" mode where the DAC portion can be left on while the analog output is turned off. In my experience tubes seem to warm up a lot quicker than digital. A "standby" mode would help balance tube life and power usage while shortening the time required to warm up.
Personally, I would prefer an espresso option so I can get tweak while I wait for the warm-up!
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Regarding 24/192 toslink -- I jusst got a 24/192 capable toslink recieverr chip as an upgrade to my AudioGD NFB-2 dac -- that and a new daughterboard with a Wolfson wm-8805 spdif reciver chip so the entire dac now is 24/192 capable except the usb input. This one will be fed from an Aurality PK-100 music server and will serve tunes either from an attached usb HD, or perhaps over the ethernet connection to files stored on a NAS -- whichever ends up sounding better.
Unfortunately, most of the sources -- like the various apple products, are limited to 24/96 on the output side with optical, so that's still a limitation.
I still prefer direct usb, but to do that right is somewhat of a large engineering undertaking as the off-the-shelf solutions -- whether async or isosync are still a good deal behind the SOTA usb implementations, which I know is the scale of project that BH probably doesn't want to get involved with, so in that case, I think the spdif is probably the best choice. Not all usb and not all spdif are equal, and good examples of each can easily outperform average examples of the other.
Hey Jim, I am a little confused here. As I understand it TOSLINK is limited physically by its bandwidth to 24/96. That being said I don't see how a 24/192 receiver module could do what it is purported to do based on this limitation. Also, I know that Amarra will only playback in native mode (mostly 16/44.1 in my house), and to be honest I have not seen many 24/96 audio recordings out there, let alone 24/192. Consequently, I am not really sure what the advantage of having a receiver chip that accepts at 192. The only advantage may be in preparing for the future, but given the evolution of ICs then the price will only go down with time.
The Amarra I bought at RMAF makes a HUGE difference in the audio quality and I am using TOSLINK, USB, and FW. One set-up has the audio leaving the mini at 16/44.1, and is sent to a DIP Combo (wordclock that can upsample to 96). The audio is then fed into a 24/96 tube DAC. I also have Amarra feeding into a FW Duet 24/96. To be honest, I cannot tell any difference in flipping from the native 44.1 to 96 on the DIP since getting Amarra. Before Amarra there was a perceived difference when I used the Audio Midi Setup in Lion or Snow, only then I can hear a difference in feeding my Duet but that is only when Amarra is off. From this end, it all seems to be in the software.
A recent EENews letter had a good article on USB vs. Firewire for audio and just blasted USB. They slammed the bandwidth issue. Yea, there is USB 3 now, but it does not seem to be on many devices. My Duet uses FW and gives me 24/96 audio. I think it sounds far superior to the E-Mu 0404 USB that I have. They say USB 3 will have a theoretical speed of 3Gb/s and that Apple is working on a new FW standard of 5Gb/s. Personally, I think that both will be gone in a few years and replaced with Thunderbolt. For now though, 24/96 seems to be where it is at.
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Yoder,
Sorry, I keep using toslink and optical interchangeably but what I generally mean is optical. You can indeed run 24/192 on optical but you have to use ST (a.k.a. - AT&T) adaptors and cabling) The HiFace Evo, for example delivers 24/192 via optical. However, even if we limit the discussion to official Toslink, people have run it at 24/176.4, as long as all devices in the chain meet or exceed the manufacturing toleances for the interface chips.
The IEEE can say whatever they want, but I know people running 32-bit 384khz via usb 2.0 with no problems. There's so much bad press onusb, but it can be done right and my ears tell me it sounds quite fine -- still my favorite dac I've ever owned is usb only, and mostly because the company has so far not been able to get the spdif version sounding as good.
I don't know where you're looking, or not looking, but I have a growing collection of 24/96, 24/176.4 and 24/192 recordings and even on a 16/44.1 dac it is very easy to tell the difference.
Amarra won't upsample? Not that I use it as upsampling pretty much destroys the integrity of the music, and I'd expect just about any device that upsamples to be a compromise somewhere, somehow.
Does Amarra support integer mode yet? BTW, even if it does, it won't do so under Lion as Apple did not include a usb driver with integer support in Lion -- you have to go back to SL for that feature, and it is a great one, especially if your dac, like mine is limited to 44/16 and the higher bit rate files have to be downsampled -- different from upsampling in that it is interpolation instead of extrapolation.
Pure Music and Audirvana both support integer mode, and Decibel may now too.
Yes, the software is a big part of it -- especially anything that does away with the iTunes playback engine (Quicktime), which is most of the 3rd party apps. However, digital audio is so vastly complex and multi-faceted that it's nearly impossible to single out any one aspect as the most important. Like amplifiers, it's a combination of circuit, components, tubes, execution, and some dumb luck thrown in for good measure.
As for predicting where this business will go, I'll leave that to the psycchics -- they're probably as reliable as anybody else.
Case in point: back in the late 80s when I was a top tier IT guy at a very large company, all the industry rags were predicting the impending death of ethernet -- with the exception that it *may* still have some use in industrial automation systems. The wave of the future was Novell and IBM token ring networks. Seen any of those lately? And where did that ethernet go? Around the globe, and around and around, and around, and everywhere in between.
I thnk all of these are viable interfaces and all of them will be or some time.
And yes, there are 32-bit 384 khz digital files available, though not many, and it's absurd anyway (with the possible exception of dithered digital volume controls.) Even 24 bits is basically insane and not really fully used -- that's 144 dB dynamic range -- show me a recording that can even come close to that.
Me, I'd be very happy with a 20 bit, 250 khz dac, but no such standard exists, nor will it ever, but to me that seems to make the most sense in terms of reproducing music.
When my Metrum Octave arrives (probably another month yet) I plan o try it with both the audiophilleo AP2 -- at a maximum of 24/176.4, and directly from he mini toslink port on the mini, which will limit everything to 24/96, which means the usb converter only gives me the benefit of 24/176.4 in native mode instead of downsampling to 24/88, and until I try it, I wo't know which is better, but my suspicion is that a glass optical connection with everything above 24/96 downsampled (in integer mode) by a factor of 2 -- meaning that 24/176 will play at 24/88 and 24/192 will play at 24/96. Not a bad compromise, but to my ears the sampling frequency is much more important than the bit depth once you're beyond 16 bits (or 20 if you have the few files in that forma, whatever it was called -- now owned by microsoft and supported on only a handful of CD players.
-- Jim
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How about Ethernet input? --- Probably really complicated I would assume (very few DAC's use it). With the software end and all that. I use Ethernet with my Squeezebox Transporter. I think its fantastic and it makes sense to me as one of the optimal "bit perfect" low jitter digital platforms. As others have expressed, USB continues to be less than optimum from an audiophile point of view, and it probably will be replaced at some point. I mean, being tethered by a 3ft, $500 USB cable is no pick-nick, when I can use a 25ft $10 Ethernet cable with no effect on sound.
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Hey Jim, regarding the various types of music and the bit rate, I really cannot find a lot of stuff that I like that goes beyond 24/96..if it even hits that. I know the Stones are putting out their entire library with some pretty high specs, but I have not like the Stones since their fiasco at Altamont Speedway. I have searched several stores for downloads and just a few seem to offer at or/above 24/96, and again the selections are very slim.
Lion uses integer up to 24/96, and 32 bit floating. I spent quite awhile talking to an engineer from Amarra, and he said that it will only use the native rate since "software up-sampling does nothing more than tack on a bunch of useless 0's." Also, as I understand it, Amarra has always supported integer mode since that has all that has been available from CoreAudio and Apple for awhile. I use FW with Lion and integer mode is supported (my FW Duet can go up to 24/96 in Lion), and then when I plug in TOSLINK it uses digital streaming. Since most of my current library is comprised of burned CDs and I use Amarra, then TOSLINK is more than adequate for my needs.
I agree that if one has a digital recording that is say 24/96, or 24/192 then there will be a profound difference when compared to 16/44. But, if one if merely playing Redbook CD's then what is the purpose of sending a 24/96 or 24/192 signal to the DAC? It is in those instances of going from 16/44 to say 24/96 on the software side that I cannot perceive any difference.
Personally, I prefer FW over USB, and TOSLINK has been good to me when run into a decent wordclock. Yea, I referred to IEEE but have heard those in the recording say the same thing in regards to FW vs USB. Since Thunderbolt will soon be rolling out with all Intel boards and Sony is now shipping it under a different name, then it is just a matter of time before it begins to dominate. It is already making its way into recording interfaces and storage devices. Please don't get me wrong in thinking that Thunderbolt will someday dominate in the audio market, at least to the extent of comparing me to the IT guy, but if audio, graphics, storage, and general computer power are to continue to evolve and meet the needs of the consumer then a much higher standard than what is available today will be required.
Regarding the 32/384 files, your comment kind of is where I am coming from regarding the 24/192. As I stated above there are not that many selections available with that format, especially ones that I like, and so it seems that it would be a bit premature to buy into it now since the field is still growing.
The digital audio recording industry still has a lot of growing to do, especially in being convinced that consumers are deserving of better audio quality than that currently provided.
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I think I see what's going on here -- are you assuming that all 24/192 dacs have to be fed a 24/192 stream? Sure, there are many 24/192 dacs that internally upsample everything to 24/192, but I'd never choose one like that myself. Any dac I get needs to support all the common formats natively as I don't care much for upsampling. Sample rate conversion is what I typically mean to be downsampling -- meanint my 24/192 and 24/176 files get converted down to 24/96, and so on, again assuming the dac is a 24/96 capable one. I also vastly prefer downsampling in multiples of 2, so the processing overhead is much less (especially in integer mode where it's just a matter of bit shifting for the most part) and no fancy interpolation algorithms have to be used.
When it comes to digital, I'm very much a less is more, Zen kind of guy -- I don't like upsamplers, re-clockers, etc. -- the more you add, the more you take away, IMO.
As far as core audio is concerned, I honestly don't know -- I was always under the impression that it is strictly a floating point internal archicteture, but I'm not an Apple wonk by any means. From what Rob Robinson at PureMusic told me, in order to do integer mode (assuming your dac supports it, and many do not), he had to completely bypass core audio, disable EQ and dithered volume control, and communicate directly with the usb hardware in port exclusive mode (hog mode). And the receiver chip on the other side of the equation has to be able to handle the data stream too, which at the present, I haven't looked into and don't fully understand.
Lion of course is another thing entirely, and while they have made some changes/improvements to iTunes, I believe it still is 64-bit floating point and still requires reconfiguration in audiomidi setup in order to switch playback sampling rates -- PM and Amarra, and most other players don't have that limitation.
Once my speakers get here, I plan to download a trial of Amarra Mini to see how the playlist/library stuff works and if the program is compatible with VoiceOver, and if it is and I can get away from iTunes, I'd love it, but it will have to be usable with the screen reader, the library will have to be easily manageable, and of course the sound quality will have to be equal to or better than PM, and to my mind that is still not at all a given -- I have way too many friends who have both and still prefer PM in integer mode oover Amarra. But I'm willing to give it a fair try.
Now, all this aside, and to change the subject somewhat, it seems most people interested in the BH dac are letting the one really most interesting feature of this dac go unnoticed -- that fact that it is a high-res NOS dac. And I don't mean NOS as in using a New Old Stock dac chip (wh8ile that may in fact be the case), it is referring to the fact that this is a Non-OverSampling dac -- a Kusunoki inspired R/2R conversion ladder, not delta sig or DSD. There are very, very few hi-res NOS dacs out there and to my ears they are far closer to analog sonics than the delta-sig chips (when done right, that is).
Does anybody get this? This is big, and what people should be excited about, not just that it's a BH product, that it will have a transformer coupled tube output stage, does 24/192, etc. I believe you can still count the number of commercially available hi-res NOS dacs on one hand, and for the most part, they are quite expensive.
Anyway, if you (meaning the editorial "you", not anybody in particular) have not grokked this, you should -- this is really the most exciting aspect of this whole project, and certainly what we've come to expect in terms of sonics from BH.
-- Jim
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I just completed my "what I want in a DAC" (aside from the fact that it's available:-)
Audio Note DAC 4.1 Limited Edition with new Triple C Output Transformers. Got about 24 hours on it, and really excited to know that it will get better as it burns in. From the first track after powering up, it sounds better than my NuForce modded Oppo SE 83.
Using a Mach2 music server over USB to feed the DAC and then on the VCap'd Extended Foreplay and Paramounts with MQ full nickel and upgraded caps running EML 2a3 to modded Klipsch RF-7s. Things have never sounded so good.
I'm currently playing with ITAP VNC on my iPad to control the Mac Mini. Not sure I like that, but at least it's better than messing with a monitor, mouse, and keyboard....
Tickwomp
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Hi Ken,
Nice setup there! What are you using for your playback software? The VNC approach is very slow, as I'm sure you've noticed, but there is supposedly a remote control app for Amarra, though I haven't confirmed or tried that yet (no amarra yet), and the iTunes remote app works fine with PM, and once configured, you can just set things up so that the PM startup script launches at boot time, and then you just use the iTunes remote app on your iPod touch or iPad, or iPhone. There is no volume control with this setup (and not sure if teh Amarra app does this either), but I run in integer mode and don't use software volume control myself.
-- Jim
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You can use the iTunes remote while running Amarra, and Amarra will allow you to control the volume if you wish. I listened to a talk by one of the guys who founded Amarra (Sonic Studios) last night and he mention both floating point and integer mode and the various algorithms used, but the only thing I was sure of at the end was that Amarra is "bit perfect." He went into quite a bit of detail about the mathematics used, and the overall development of the application.
To get Amarra or any app to run at login with OS X, then goto Preferences->Accounts->(Select User Name)->Login Items and then add (+) your application. When Amarra starts, then iTunes will also start. If you do it manually you must start Amarra first.
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Hi Ken,
Nice setup there! What are you using for your playback software? The VNC approach is very slow, as I'm sure you've noticed, but there is supposedly a remote control app for Amarra, though I haven't confirmed or tried that yet (no amarra yet), and the iTunes remote app works fine with PM, and once configured, you can just set things up so that the PM startup script launches at boot time, and then you just use the iTunes remote app on your iPod touch or iPad, or iPhone. There is no volume control with this setup (and not sure if teh Amarra app does this either), but I run in integer mode and don't use software volume control myself.
-- Jim
I bought my Mach2 setup with PureMusic. My understanding is that the iTunes remote will work with PM too. But, the remote won't do what the iPad can do regarding searching through almost 2 TB of AIFF files;-) The Mach2 is configurated for battery use, nice and clean!
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Ken,
I just got my mini converted to battery power as well, along with a Pi battery buss, but haven't tried that setup yet. Very soon, I hope. The Oyen mini-pro 800 FW drive will also get it's power from the battery buss, which is also supposed to make a nice difference. All to be determined shortly...
-- Jim
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You could also go the cheap route that I did and have a local shop install a SSD drive for the boot drive, and then use the existing drive (500 Gb) as the secondary (have the music on this one.) It cost me $145 to have a 64 Gb Crucial SSD installed. If you have an Apple "Superdrive" then there also exists a kit that allows you to pull it and replace it with either a SSD or conventional HD. They are starting to produce external SSDs with a Thunderbolt interface that would also work as a boot drive...a little pricey though, but then all HDs have gone up quite a bit lately due to the flooding. Though the flooding happened a couple of months ago, I think the industry is still trying to capitalize on it with the continued high prices.
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Back to the topic at hand: "What I want in a DAC"
Having the capability to do something like this would be a nice for future upgrades:
Full Upgrade (http://www.ebay.com/itm/Audio-Note-DAC-Full-Level-1-2-Upgrade-/370484856300?pt=LH_DefaultDomain_0&hash=item56429e45ec#ht_1223wt_1165)
Digital Upgrade (http://www.ebay.com/itm/Digital-Upgrade-Audio-Note-DAC-24bit-96khz-Level-1-/370535571162?pt=LH_DefaultDomain_0&hash=item5645a41eda#ht_1483wt_1165)
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If I had the choice to buy any DAC available today under $10K I would go with the new and improved Berkeley Audio Alpha DAC II and not look back. If I had the $5K or so to spend there are other bigger bang for the buck that would improve my system, ie. amps and speakers. I think we get hung up on the bits though.
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This is "off topic", but the AN upgrade parts I think is a super idea, and it hit on a question that i have always had, i have a cd/dac combo that was $6000 a few years back , I love it, however, i saw that it has 2 BB 1702 chips in it, and i was always wondering if i could just pull those out and install some 1704ks in it, without having to do anything else? i know nothing about this stuff.... but just for fun....
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Hey Chris - I'm hearing you on your urge to tinker! I have blown a couple of components by getting my fingers (and soldering iron) in places I probably shouldnt have been. Fortunately they were "expendable" in terms of price and I could move on. Sometimes I get the yen to crack open my Squeezebox Transporter ($2K). I so far have resisted, and I think that as far as DAC's are concerned, thats the best policy.
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Wax... I think you are absolutely correct, and Docs Dac will have the 1704 in it anyway, I will just get that one and have fun comparing.... ruining a great component is definitely a "tinkerers" worst nightmare...
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Hi Guys, I had a few minutes to spare and saw this thread, I thought I would give a little clarity as to what will be in the first pass of the DAC. (Doc doesn't even know all this!)
The physical packaging that Doc and I came up with is somewhat limiting, the board size is 4x3 inches which sort of limits what I can put directly on the main board.
Whats on there right now (which is most likely what will be there for the real thing, I really don't want to re-lay out this board) is S/PDIF coax (BNC), S/PDIF TOSLINK and USB 1.0. I chose BNC so that if you really do have a good proper 75 ohm source (not very common) you have a way to get it directly into the DAC. For those that have RCA cables you can use an RCA to BNC adapter. another reason is that you can buy AES/EBU adapters that will plug directly into this connector. The TOSLINK is standard TOSLINK, it's the fastest one Toshiba makes today, I really don't know if it will do 192 since I don't have anything that produces a 192 TOSLINK signal.
The S/PDIF coax input is very special. I do NOT use an off the shelf receiver chip. I designed a special circuit which radically "cleans up" the input signal which makes it far less susceptible to reflections and noise, which hopefully is going to get rid of a fair amount of influence from source and cables. Unfortunately this circuit takes about 25 parts, which takes up a lot of room on the board. The output of this circuit goes directly to the FPGA which does all the S/PDIF decoding.
I was going to do the same sort of input circuit for the TOSLINK, but I ran out of room on the board and figured it was more important to get it on the coax input. If the coax winds up being significantly better than the TOSLINK and there is interest I might wind up making an external TOSLINK module that uses the same sort of input circuit.
The USB is USB audio spec 1.0, which will go up to 24/96. Doing USB audio 2.0 takes considerably more hardware and I simply didn't have room on the board. The signal from the jack goes into a simple transceiver then directly into the FPGA which does all the USB decoding. Contrary to all modern hype and my own philosophy on the subject, this USB input will not be "asynchronous", it will be adaptive. See the next paragraph as to why this is not so bad.
The "secret weapon" of this design is the clock circuit. A major issue with S/PDIF and adaptive USB is that the DAC must somehow "sync up" with the data rate from the source. This is usually done with a PLL which creates a fair amount of jitter on its own. There is another approach which uses a low jitter adjustable oscillator and a buffer, the input puts data in the buffer and the local clock takes it out. When the buffer starts getting close to full or empty the circuit speeds up or slows down the clock. This is what I have in here. Doing this well is not easy which is why its not done very often. It is true that an adjustable oscillator of a given price is always going to be higher jitter than a fixed one, but this is a very good adjustable one. It is the single most expensive part in the whole design. Using this circuit with the USB lets me get away with using adaptive, but its going to have lower jitter than all but the very best async designs.
There is also a card edge connector which will have a slot in the top panel. The original purpose was for a sample rate display. If you want the display, plug in the board, if you don't want the display, don't plug it in. It uses standard LED displays which come in many colors, so if you want a different color than what Doc provides in the kit you can easily buy them. The display is static, meaning that there are NO changing signals going to the board except when the number changes, this cuts down a lot on noise generated by a display.
Then I started thinking about the connector and what I could do with it. I've come up with a universal expansion protocol that can be used with this connector. It uses LVDS which radically cuts down on radiated noise so it won't be too bad. This lets anybody make an expansion module that plugs into this slot. Some possibilities are:
Multiple S/PDIF input board, USB 2.0 input board, fancy displays such as Nixie tubes or VU meters, or even magic eye VU meters etc. And my favorite (but don't tell Doc), the possibility to slave DACs so you can have a multichannel DAC. A USB 2.0 input would have plenty of bandwidth for say 6 channels, it would also have a special connector on it. Each additional DAC would have an expansion board in its slots and cables would daisy chain between them. I'm sure it could be made to work, but it would take a lot of effort.
My idea with this is to publish this expansion protocol so anybody that wants to make a module that plugs into this slot can do so. Its not super simple to make one, you need a processor or FPGA to talk the protocol, but its not super difficult either. Over time I'm sure I will be coming up with some of these and Doc will be selling them. But don't expect any of these right away.
One other thing, it comes with firmware in a socketed chip, so that when we come up with a firmware change Doc can send out new chips in the mail so you can get easily add new features. But with the above mentioned expansion protocol you won't need to change the firmware just to add a new module.
I hope that gives some idea of what is going to be in there.
John S.
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Thanks for the update, John. I'm really looking forward to putting together whatever kit you and Dan come up with.
RE: adaptive vs. asynchronous... I know I'd prefer a well-done adaptive board over a copy & paste of some chip manufacturer's asynrchronous development layout.
Regards,
John
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My thanks to John for weighing in. If I understood all of it that would be great but I get the gist.
The real cookie there is the expansion slot.
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John S,
Thanks much for the update and more details. Glad the toslink is in there too and as far as I can see, the new toshiba toslink receiver and trnsmitter chips are 24/192, but there aren't all that many 24/192 sources on the market yet -- the hiface Evo is one though.
I really am impressed with the clocking subsystem -- as you said, not seen very often, but really does level the playing field between adaptive and async.
Anyway, my listen to the prototype at RMAF was very, very promising, so I'll be having my eyes on this one for sure.
-- Jim
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John S,
Thanks for the tech info on clocking and USB etc. It add to the concepts I am trying to build in my head about how all this stuff works! I was wondering if you could explain the issues/advantages etc. with a ethernet protocol. I know its complex and needs custom software etc. Any info would help. -- Eric
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Hi Eric,
the subject is quite large and complex, I could easily write a couple books about it, but I'll try and put it all together in a nutshell here.
To make all the rest make sense you have to have SOME understanding of jitter, its impossible to understand why different implementations may or may not be better without some understanding of what it is. Jitter is the variation in timing of samples applied to a DAC chip, this is what matters. The whole concept of digital audio is that "samples" (numbers that represent the musical signal value at a precise point in time) are applied to a DAC chip at very precise times. (a CD applies these samples at 44100 times per second). If anything causes the time between samples to change even slightly, the waveform at the output of the DAC is distorted. Human hearing seems to be very sensitive to some forms of these timing deviations.
A few things about jitter, number one, it is ALWAYS there, no matter what some marketing departments tell you, it is impossible to produce a zero jitter signal. You can get it pretty small, but its always there. Different circuits produce different amounts and types of jitter. Even the same circuits can produce radically differening amounts of jitter with different amounts of noise on the power and ground lines supplying the circuit.
There is a major tradeoff here, many of the methods used to decrease jitter actually wind up producing significant amounts of ground plane noise, which winds up producing more jitter than you thought you were getting rid of in the first place. This is not understood by a lot of designers today, thier attempts at reducing jitter actually make things worse.
Now on to some specifics. First lets talk about S/PDIF. This is a class of protocols in which the source is "in control". The rate at which the data comes over the wire is completely determined by the source. The DAC has to somehow "lock on" to the data stream and produce a signal going to its DAC chips based on the rate at which data is coming over the wire. This has traditionally been difficult to do with very low jitter. This is traditionally been done with a device called a Phase Locked Loop (PLL). A PLL locks on to the input data stream and produces it own "clock" which tracks the input signal. Unfortunately the jitter from a PLL can have anywhere from 2 - 200 times as much jitter than a good fixed frequency clock (such as a crystal oscillator). How good that PLL does is VERY dependant on the signal feeding it, ie the transport. In the early days many transports were not very good and the PLLs in the DACs did a very poor job of extracting a clock from these signals. This gave rise to the "reclocking boxes". These contained a a much better PLL (and expensive) PLL which did a much better job of extracting a clock from the source and feeding it to the DAC, which now had a better signal to start from. Transports have gotten better and DAC receivers have gotten better, so outboard PLLs are less effective than they used to be.
So this brings up the first thing people realized, trying to track something else will almost always have more jitter than if the DAC generates its own clock. Thus was born various methods to have the source follow the DAC. Some companies put the clock in the DAC and fed it to the transport. This should completely fix the problem, but it didn't. Other "computer" protocols were tried such as asynchronous USB and ethernet protocols. Both these allow the low jitter clock to be in the DAC, and the source follows the DAC. But non of these have been completely successful, in that ALL of them still are susceptible to what is going on in the source. The problem is that most of the designers are ignoring the fact that that these "feed forward" systems generate more noise in the DAC power supply and ground plane, which increases the jitter of the local clock.
I'm a very big fan of the Squeezebox system, what drew me to it in the first place was the architecture, a local low jitter clock that fed the DACs and a system which grabbed data from a server when needed rather than a server which shoved data out when it wanted to. Unfortunately the ethernet system requires a local computer (ALL ethernet streamers have a computer built in) , and complex software, which generates a ton of noise on the PS and ground planes which compromises the theoretical extremely good jitter performance. Unfortunately none of the companies making ethernet streamers have realized this and designed their systems with this in mind. I personally think ethernet streamers are a very good way to go, but there is a lot of work to fully realize the potential.
Asynch USB should be a good compromise, the protocol is much simpler than ethernet protocols, so properly using them should generate much less noise, and it still offers a way for source to be slaved to the DAC. But it STILL does generate some noise from implementing the protocol which will generate noise on the ground plane. Designers have been almost ignoring this aspect. A few have made some attempts at mitigating this, but NOBODY as really come close to addressing this fully.
So why have I chosen the path I'm taking with this DAC? Its a good compromise. S/PDIF is very common, almost every device that deals with digital audio supports S/PDIF, async USB and ethernet audio are a much smaller space, I wanted to produce a DAC that would be useful to a very broad range of people and circumstances. The only problem was coming up with a way to implement it that would be as good as other methods. I think I have achieved that.
So for people that don't have an S/PDIF output but do have a USB jack, I added a USB input which should be quite good. If you want to go ethernet streaming you can get a squeezebox Touch and plug it into this DAC, no need for a very expensive ethernet streamer made by an audiophile company.
I hope that sheds a little light onto this rather large mess that is digital audio.
John S.
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Thanks John! - It does help piece the mess together a bit. It seems SPDIF was played as the "enemy" in the early USB days, to get audiophiles to consider the emerging technology. I never quite bought it, and now SPDIF does seem to have come back into acceptance when utilized intelligently with modern techniques. Thanks for the info! - Eric
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John, always great to hear about the process of building the DAC which is my favorite part of designing and building things. Although I have little technical knowledge about the inner workings of electronics I like most on this site enjoy learning what goes into the product development stage. In reading about the various platforms and the various trade off between them I cant help but think that S/PDIF will only continue to shrink as a digital platform where others such as toslink, usb, and ethernet will proliferate. If the computer is gaining popularity as the predominate source for the audio data which I see as being true then I would look at what is the most common data transfer protocols. If I were to pick three I would say USB, Ethernet and Wifi are by far the most common methods to move data between devices. When was the last time you saw a computer built by HP, Dell, or Apple with a S/PDIF connection sticking out the back?
From what I gathered from reading your posts is that S/PDIF is able to most easily deliver the packets of data between the computer and DAC with a high degree of precision and fewer trade offs which might degrade the over all sonic integrity. What I am wanting to know is how difficult is it to get the least amount of noise on a growing audio transfer protocol like USB? Think about computers ten years from now and how the audio market will shift. The focus of the DAC should be how easily can I get the best sound of my digital files and into my amazing sounding Bottlehead product. My Ideal situation would be a wireless DAC that can serve my setup located in another room. Most people dont want to stick their computer right next to all their Hifi gear, they just want it to store their audio tracks in a digital format. Now that I have an awesome hi-rez music library I want to be able to hear it on my speaker rig in the living room AND my headphone system in the bedroom. The squeeze box which you talk about is the closest product I have seen which does what I am describing. It too has the ability to connect to my wireless home network and really leverage the benefits of computer audio which is holding an enormous music library on a small magnetic disk and accessible from anywhere on the planet.
I'm sure developing a decent wireless system is much more complex and expensive to develop than a simpler method like USB or S/PDIF but looking at this with an eye on the future of where things are headed and what would be the most valuable features to have I don't see wireless technology declining. Using an open source platform like Arduino or ZigBee might yeild something promising and you will end up leapfrogging the competition. Maybe this can all be the foundation of the Bottlehead DAC2?
Regards,
Tom
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Hi Tom - my two cents worth of experience with the squeezebox Transporter using wifi vs. wired ethernet. This has been discussed ad-nausium on the squeezbox forums with heated discussion on the logical conclusion that there should be no sonic difference between wifi and wired ethernet. Well, there is. The Transporter sounds better over ethernet, not by a small amount. Steve alluded to the noise in these applications, and I think that is the reason. The added noise from the radio etc impacts the sound. On the Transporter with ethernet engaged the radio is off. The plot thickens!
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This topic/threads/discussion is the most comprehensive i have seen on DAC. With that, as it appears a lot of technical effort is going to be applied to the DAC, my inputs follow:
There was a post that indicated ability to switch multiple inputs; ( pre-amp sourse selector). Given the ability to leverage software play lists etc,
i am not sure if this is beyong the call of duty.
Also, looking to the future; as it appears than the days remaining of CD's is somewhat limited, whatever the DAC includes as functionality/interface should accommodate the growth or road where digital music is going; itunes.
just some thoughts
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I hope the upcoming DAC is worth the wait!
My impatience got the best of me and I was couldn't wait to a music server -> DAC into my system. The only time I actually handle CDs now is to rip them into storage. I'm using an iPad and iTap VNC to run the Mac Mini headlessly and things couldn't be better/easier. Like a super remote control accessing approx 1500 CDs worth of music. My wife came in the other night and said that I was driving her crazy bouncing around and ask me to let something play for longer than a minute:-)
Now that I've lived with my first outboard DAC for a while, I'd like something that would simply play materiel in it's native resolution without upsampling or at least have that option. Need lots of input options too including USB, FW, SPDIF, etc...
Be nice to have the 'digital board' modularized and separated from the power and analog sections enough to allow future tweaking/upgrading.
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Guys we are making progress. Pretty soon we will be able to rename this thread "What you're gonna get in a DAC". We're in another one of those modes where it's all about budgeting the money to continue to develop the project. Things should be moving again soon. If anyone wants to help propel the project along, a few Paramount orders might help...I'm just sayin'.
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At this stage, i believe an important stage is go back to "requirements" vs. "functionality", and compare against what is perceived to be the competitive product(s) if any exist. What is more important here is leadership. There should be every attempt to review the "system" and somehow fortify leadership strength in that a competetive product for example , should not be able to change a connector or cable type and then identify company and product leadership. The "system" in its completed form should be what others try to achieve.
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I'm sorry, I don't understand what you are saying.
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What i am saying is that; there is and has been so much in this thread that i cannot actually identify all functionality that readers and potential customers are requesting. It is a task beyond by ability but certainly a task within your grasp to figure out what everyone is saying and where the product goes.
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Peter,
The poster starting the thread is a customer. Doc, Paul J. and John S. have been working on the Bottlehead design. That is a different thread all together.
No DAC on the market will include everything that is listed in this thread as what each poster wants.
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i would imagine the main goal is, to provide as much flexibility and future upgrade ability as can be reasonably achieved... and then to smoke anything, anywhere near its price with its sonic bodacity!!!!!!!!!!!
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An optional wine chiller attachment would be well received as it combines two things many audiophiles enjoy. :P
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I'm very interested in seeing how this one continues to progress. How did the Head-Fi meet go? Any impressions? I didn't see any Bottlehead DAC impressions in the Head-Fi threads.
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I was able to heard the DAC at last weeks Head-Fi meet in NorCal. It's really quite big and occupies two enclosures which are about the size of the SEX amp. The sound was really great listening through the Smack and SEX, but it's hard to tell how much the DAC is adding to the sound when you can't directly compare it to another source. I was also fortunate to meet the creator of the PCB (John) for the DAC and even see the next gen circuit board for the new iteration. Really cool and unconventional things are in the works but it seemed like production version is still going to be another few months away.
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I'm curious to hear it. My Juli@ sound card running with J.River Music Console sounds fantastic. I have it doing the DAC. I really can't complain about the sound. Someone on another forum said the Juli@'s internal DAC matched his Naim DAC. Can that be true? In any event, it would be great to hear what BH's DAC (or any DAC for that matter) does for my FLAC.
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I have been using a modded squeezebox2 for a about 6 years now and have been very happy and felt that it is a great interface for accessing music and to me it sounds better then using a CD player. At least the ones that I can afford. I have been on the fence about upgrading to something else waiting for the BH DAC. I want a DAC that is not limited by the USB interface, i.e. can play 24bit and files up to 192k hz. I have looked at the SB Touch in the past but held off because it only was only able to do 96K. Now there appears to be an app that allows it to stream up to 192K FLAC files via USB and S/PDIF. I'm thinking this is maybe the a very good option that has future potential with the BH DAC. Any thoughts from the forum?
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The new applet for the Touch works very well. You will be able to use it to send up to 192 to the Bottlehead DAC via S/PDIF, either via coax or optical. The optical is a little iffy, the data rate is theoretically faster than what the optical can handle, but in reality if you use a fairly short good quality cable it should work. But if it doesn't, use coax, that WILL work.
The USB input that will be part of the Bottlehead DAC will only go up to 96KHz. This is primarily a space and time limitation, doing 192 takes a lot more hardware and we just don't have room on the board. In addition it will take a fair amount of work to get it working properly and I don't want to delay the release of the DAC just for that.
The external interface that WILL be a part of the DAC will allow us to add a 192 capable USB input at a later time, exactly how much later, who knows!
John S.
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In my personal opinion 192k isn't all that important for playback at all. The people at Benchmark who build some fairly serious professional gear that can easily handle 192k data rates tend to agree and have written about it on their website.
High data rates are great for digital processing. Most of the math that gets used in studios - recording, mixing and mastering - works better at higher sample rates, because the audio frequencies that are being manipulated are nowhere near the sampling frequency. DAC's and ADC's on the other hand don't benefit much. 24/96 is plenty good enough. I often go with 24/48 even when I could just as easily use 24/96.
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I just found the quote by John Siau of Benchmark Media. It is from March 31, 2012:
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I have a Pacific Microsonics Model Two sourced from a DAW loaded with Pyramix Software and a Mykerinos card. The difference between 24 192 and 24 96 is easily detectable when listening to it through a high resolution system.
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Don't want to stray of topic, but we are discussing DACs. Where exactly do the hi-rez files originate? I know little or nothing about recording and mastering, but my guess would be that the only source suitable for conversion to 24/192 resolution would be the masters or copies of them. I have found that alot of the files available at HD Tracks and others are vinyl rips. I have been doing some vinyl ripping of my own, using a borrowed Benchmark ADC. I was told that converting vinyl to anything above redbook was a waste of time, but I found ripping to 24/96 gives me the best results. The editing software I use, to remove pops and clicks and so on seems to do a better job with more data. Back to my original thought, I noticed NAD has released a DAC that is capable of 32 bit / 384 khz resolution. What for? Are they hoping that at some point the recording studios will begin producing hi-rez along with other formats for general consumption? Just wondering if these capabilities available now are worth spending money for.
Cheers,
Shawn
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All sounds very promising (John), do you folks have any idea yet if this will be a one or two chassis dac? I know that one of the recent prototypes shown was in 2 chassis, but was that for prototype purposes, or is the final dac looking like it will need to be configured as such?
I've got some good time on my CEntrance DAC Mini now and I'm not really warming up to it as I had thought I would, which is opening the door for the BH dac to be part of my stereomour/Orcas system, but rack space is at a real premium and I'm not sure I could manage a 2-chassis configuration.
Thanks for any info you can share,
Jim
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How many boxes is really up to Doc and PJ, but the current thinking is that it will be a single box, with an optional external power supply to get it to REALLY sing.
Power supply isolation is one of the critical issues with DACs, but to do this DAC full bore will take so many power transformers and chokes that there is no way to fit it all in one box. So the idea is that the base unit will include the power supplies needed to make it sound very good, and have a second box that will bring it over the top.
You will be able to add the external supply at any time so you can buy the base unit first then add the external supply if and when you wish to.
I hope you don't get the impression the base unit is going to sound bad, far from it, In the lab right now I have it running with a power supply configuration close to what will be in the base unit and it smokes everything else I've ever heard (I'm just slightly biased). Maybe "smoked" was the wrong word to use for a tube based DAC, but I hope you get the idea.
John S.
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Hi John,
Thanks for the reply and that sounds perfect to me.
I don't know how things have progressed since last RMAF, but I was really impressed with what I heard of the dac at that point -- even more so when I later learned that it was being fed directly from the optical out on the Mac Mini.
Now I'm really looking forward to all of this! :-)
-- Jim
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... my guess would be that the only source suitable for conversion to 24/192 resolution would be the masters or copies of them. I have found that alot of the files available at HD Tracks and others are vinyl rips.
As I understand it they use 24/192 for the master in digital recording now days, though some are probably beginning to push that envelop, and are starting to push 24/96 as the consumer standard. Though Apple's iTunes Plus will present a different story about hi-res mp3's.
You sure about the vinyl rips? Most re-masteriing uses the original master tape (There are some good movies on Netflix under "music" called "Classic Albums." You will see Steely Dan, the Dead, etc. using what looks like 24 track tapes and some 8 track from way back.)
I have been doing some vinyl ripping of my own, using a borrowed Benchmark ADC. I was told that converting vinyl to anything above redbook was a waste of time, but I found ripping to 24/96 gives me the best results.
Somebody does not know what they are talking about. Going above Redbook would only be a waste if your intent was to burn all of your vinyl music to CDs. If you have the hardware and software to handle 24/96, or better yet 24/192, and plan on using computer playback then go for it. 24-bit is far superior to 16-bit.
I noticed NAD has released a DAC that is capable of 32 bit / 384 khz resolution. What for? Are they hoping that at some point the recording studios will begin producing hi-rez along with other formats for general consumption? Just wondering if these capabilities available now are worth spending money for.
It is just a numbers game. I have not heard any advantage in up sampling beyond the native rate. That being said, I have not even seen any 32 bit audio for sale. Regarding the purchasing of such, it is NEVER cheaper to buy new technologies. Personally, I don't thing going beyond 24/96 would make much difference to my aged/tired ear drums, and so I am perfectly content with 24/94.
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Interesting discussion.... What I might add to the mix is that, IMO and experience, there are potentially other factors that influence the sound at various sampling rates/resolutions. What I have found is that my laptop upsampled redbook (24/96) using Sox and minimum phase settings (0%) sounds the best, better than the usual default linear phase (50%). This is a curious result because my Transporter upsamples redbook in realtime with Sox as well, but passes high res. without upsampling. I have tried various phase settings from linear to minimum. All of them sound better than native redbook, minimum phase is the most 'vinyl like' IMO. This may have to do with filter effects on ring/pre-ring.
So, in this case, redbook that is upsampled before hand appears to sounds superior to redbook that is passed to the DAC as 16/44.1 and upsampled by the DAC. This is not a small effect. It is very clearly different. Everytime I get to doubting myself and switch back to native redbooks rips I come to the same conclusion. I wish it was not true, because I could have my entire 1000 CD collection on my laptop in native rips. Now I have to settle for 250 (large upsampled) albums or so. I would quantify the differences I hear in the area of 'ease and liquidity'. A 'naturalness'. Musical and believable with an added lack of fatigue and irritation. Very clear and obvious.
This effect has also narrowed the gap as far as my system is concerned, if there even is one, between native HD tracks 24/96 downloads and my upsampled redbook. I have not found that high res. downloads sound any better than a good redbook recordings massaged with Sox.
Now the question that is raised in my mind, is what role the various 'decoding' (filtering) processes have on sound quality, irrespective of resolution. Sorry if I'm making things more complicated! The plot thickens!
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Interesting new article on the subject...
http://www.enjoythemusic.com/magazine/manufacture/0512/
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Interesting discussion....
Interesting indeed...
I run my audio through Amarra that uses only the native rate, and fixed point. The difference between Amarra and iTunes is profound. (They have a little button one can use to make comparisons.) Use the Amarra parametric equalizer and the difference is even more dramatic. I feed my streaming digital audio from my mini into a 24-bit tube DAC and so really can't provide any feedback as far as 16 vs 24 bit in the DAC goes, but when I play 24/48 or 24/44 files through the system the difference it is like night or day when compared to Redbook specs. The key, for me, seems to be in the bit-depth. I have an external up sampler/clock that allows me to flip from 44.1 to 96 on the fly, and I can sit in front of the speakers and flip 24/44 files between 44.1 and 96 all day long and do not notice any difference. I am going to order a couple of SACDs today (24/96 and 24/192) and will try the same thing. My guess is that there will be a noticeable difference when I downsample a 96 or 192 file to 44. When upsampling the sample rate all that is happening is that a bunch of zeros are being tacked on, and my guess is that that is why I do not hear any difference.
There are some transports/DACs that advertise as having various capabilites using high-end DSPs, and adaptive time filtering "that applies polynomial curve-fitting interpolation that allows for greater data buffering and re-clocking" so as to remove "almost" all digital jitter. For me this creates a conundrum. Namely, all things being equal should I per sue higher bit-rates that that will sound great when played through excellent gear at the native rate, or is it worth per suing high-end DSPs/hardware and the improved algorithms that come with them (firmware that is actually altering the input audio in various ways)?
Personally, I am very content with going for hi-res audio and playing it at its native sample rate. Correct me if I am wrong, but this is pure, unaltered audio versus audio that has had various algorithms applied to it. Granted, algorithms are used in all digital playback but there seems to be a point where when applying these algorithms we go from decoding the digital signal to actually altering it in various ways from its original form in order to create a "better" sound.
Anyway, 24/48 audio blows me away as it is and I can hardly wait to hear the 24/96 I have on order. I will let you know if I hear any difference when I down sample the sample rate. The two SACDs I am getting are: Pink Floyd-Dark Side of the Moon, and The Band-Cahoots.
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Hello, Yoder and Eric, thanks for the discussion. First off, I should edit the statement about HD Tracks. There seems to be more natively recorded hi-res files available. On many of the early downloads from HD Tracks, and other sites you can hear the vinyl rumble. Very faint, but there nonetheless. It seems there are more recording companies willing to supply Hi-res digital. I made some downloads from Harmonia Mundi and a Norweigan company called 2L,(who also have files available on their website), that sound absolutely fantastic. I am using Pure Vinyl to do my rips from vinyl and I am very happy with the results. I still prefer a pure analog vinyl rig but I think we are making progress.
Cheers,
Shawn
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Hey Yoder - Thanks for verifying that I am not the only crazy MF out there obsessing over this stuff!
Its pretty cool that your system gives you that much A/B control to be able to make some comparisons on the fly. I dont have that. I am convinced that shifting the aliasing etc. to higher frequencies with higher sample rates is key. Probably equally as important the filtering algorithms that come into play at the different sample rates. From my systems perspective, I'm not hearing a huge difference with native high res. material. It has more to do with the recording quality. A good example are the 2009 Beatles redbook remasters. Upconverted with Sox they sound as good or better than the native Paul McCartney 'Band on the Run' 24/96 HD Tracks download.
Cheers - Eric
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On many of the early downloads from HD Tracks, and other sites you can hear the vinyl rumble. Very faint, but there nonetheless.
If there is true remastering of older audio going on then they are using reel-to-reel tapes. Only with the original master tape can the sound engineer go in and play with the various tracks on the tape. The movies I referred to above go into a lot of detail about this and they play with eliminating various instruments, enhancing them, etc. This type of work can only be done with the original master tape and not vinyl or with any of the digital codecs (AiFF, WAV,mp3, mv4, etc.) once the tracks have been "compressed" from 24-tracks to stereo. The "rumble" you hear is probably the result of a poor quality master tape. Finding mint perfect master tapes seems to the real challenge today.
Here are a couple of quotes about some HDTracks hi-res audio. "...with our exclusive Rolling Stones release, ABKCO Records undertook a long and painstaking re-mastering process employing state-of-the-art master tape to digital file transfers using the best in Analog to Digital converters to provide HDtracks with the highest quality music files possible."
The engineer for Cat Stevens said:
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Not all masters are multitrack and thus you can't make that generalization. The term for mixing multitrack session tapes to stereo is mixing, not compressing. A bad master tape will not have lathe rumble. That is an artifact of cutting lacquer or playback on a noisy turntable. A bad master tape can have a lot of other problems - dropouts, rolled off highs, dulled transients, stretched spots and splices, sticky shed, maybe even "rocks", but not lathe or turntable rumble.
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I have a download of Nojima plays Listz that you can hear the stylus hit the record, you can only hear it with volume up above normal listening levels. I have listened to enough vinyl in the last 30 some odd years to recognize that sound, and I am not the only person to notice this on various tracks. Please, don't get me wrong there is nothing wrong with the files quality, the Nojima piece sounds great. I simply, in my own lack of experience, wonder how this business works. Where exactly is the media originating? Are we paying for files prepared by professionals with studio equipment and backing, or something done half-assed in somebody's basement? (similar to mine)! ;)
Cheers,
Shawn
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I doubt very much that Nojima plays Liszt would have been transferred from a vinyl pressing to a high res file for commercial sale. Our Tape Project album of that title has been made from the original tape recording - which is in very good condition. AFAIK it was recorded simultaneously in 176/24, so there would be little reason to waste time transferring it from an LP.
Your description of the level being set above normal listening levels to hear the effect makes me think you are hearing the tape hiss starting as the tape was rolled (if it was transferred from the analog master) or possibly air conditioning rumble in the noise floor.
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Good point Shawn!
That is exactly my sentiment when I think about purchasing a 'new' 'high res' download. If there was all this effort placed in remix/remaster etc, etc, I would think that the studio would also be interested in releasing it in a blu-ray, DVD, CD, box set etc. for mass distribution. Anyone can, and do (illegally), upload a vinyl high res. 'remaster'. When buying a new remaster I take into account who did the work (ex. Rhino) and what reviewers are saying about it.
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Your description of the level being set above normal listening levels to hear the effect makes me think you are hearing the tape hiss starting as the tape was rolled (if it was transferred from the analog master) or possibly air conditioning rumble in the noise floor.
So many avenues for noise to get into the process, no wonder sound engineers tend to have obsessive personalities. I would love to hear the Tape Project version of Nojima Plays Liszt, it is a favorite of mine. I think you are right about the noise, the sound quality is darn good at 24/96, not as good as my vinyl version but good enough to listen to regularly. Maybe I should quit thinking about it and just listen!
Cheers,
Shawn
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Dang, three high-and-inside fast balls!
Not all masters are multitrack and thus you can't make that generalization.
True, but when talking about high-resolution audio don't you think it is a fair assumption that they are using multi-track studio tape masters?
The term for mixing multitrack session tapes to stereo is mixing, not compressing.
After years of working with Logic, I was taught that the term used to combine several tracks into mono or stereo is "bounce;" it is also a standard term in audio. Likewise, "mixing" was a term used to described the process of shaping the overall sound of the project by adjusting vol levels, panning, messing with the EQ, and using other effects on the project. But, being half brain dead and suffering from tip-of-the-tongue phenomenon at the time of the writing, all I could think of was compression, which actually deals with the "attack" and "release" in audio. So, I used the computer file definition of compression and related it to making a file smaller as in going from a huge multi-track project to a smaller file like an AIFF.
A bad master tape will not have lathe rumble.
Oops, forgot a word--perceived. 'The perceived "rumble" you hear is probably the result of a poor quality master tape.' In other words, I was questioning if the perceived "rumble" was some other sound on the tape and was being mistaken as "rumble." In any case, I was suggesting that the poor sound was due to the tape.
Personally, I do not see how a company like HDTracks could burn "hi-res" audio off of vinyl and re-sale it as being "remastered." There are some crappy re-masterings out there, but they usually seem to be casualties of the "loudness wars," or as if someone went out for coffee and put the process on auto-pilot and allowed over clipping to take place.
@earwaxer: I got my hybrid two hybrid SACDs the other day. Looks like SACD uses the same licensing scheme as HDMI does. Ergo, SACD audio can only be played through analog outs. As soon as I change the output to my SPDIF or optical the player reverts to CD mode. Then it is in big print in the manual that "SACD can only be used with analog outs." Bad news is that one is reduced to using the DAC in the SACD player, but if one wanted a better DAC then there are always the high-end CD/SACD players available. So, I was unable to see if there is a much difference in down-sampling from 96 to 44. The good news is that I can still burn the Redbook CD to my hard drive.
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I think that companies that are reissuing stereo releases in high res are most often working from stereo mixes rather than multitracks. Bouncing is that act of mixing two or more tracks together onto an unused track so that the original tracks can be repurposed.
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Yes, what Doc says -- most of these companies are using stereo masters even though sometimes they use differently mixed masters from the album that is in general circulation -- there is a thread on the Hi-rez music circle on audio circle that has a list of bad or alternate mixes being used by HD Tracks and others and people being mad about the fact that they don't sound like the same album they were expecting, but in hi-res.
As for SACD,it's not so much the licensing scheme as it is the underlying technology. SACD is DSD encoding (Direct Streaming Digital), not PCM so there is only a streaming rate spec, not a bit-depth (as it is all one-bit depth) This is in contrast to what Ryan Mintz of Core Audio seeems to be implying in his article in ETM that was mentioned earlier, and in fact, one-bit digital is here and has been here for a while and is obviously within reach of many. Furthermore you can almost always bet that if you see a file on HDTracks that only comes in 24/88.2 PCM format, it was made from an SACD/DSD master as that is the PCM limit for commercial SACD recordings.
As a side note, Pure Music can natively play DSD files if the dac it is feeding has a DSD chip, but it can also take dDSD files and convert them on-the-fly to PCM format and play them on your normal PCM dac.
From my subjective perspective, I have a 2-layer SACD of Jorma Kaukonen and friends' "Blue Country Heart" and I'd far rather listen to the redbook layer through a good PCM dac than the sacd layer -- just sounds more like real music to me.
Several people have conducted a/b/c tests with vinyl, redbook and SACD, and most people preferred them in that order with SACD a fairly distant third.
BTW, there are ways to rip the DSD layer of an SACD to PCM, but it does take some special software and a special disc drive, but not having done it myself, and only owning a very small handful of SACDs, I don't plan to to this myself -- especially when PM can do it for me.
-- Jim
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From my subjective perspective, I have a 2-layer SACD of Jorma Kaukonen and friends' "Blue Country Heart" and I'd far rather listen to the redbook layer through a good PCM dac than the sacd layer -- just sounds more like real music to me.
That is a good CD, though I think "Quah" is his all time classic and is on my top ten list.
I burned the CD version of "Cahoots" onto my hd and played it through Amarra and, like you, preferred the CD version over the SACD. Think I will stick to CDs, and hi-res downloads.
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Yeah, Jorma is one of my all-time fave guitarists. I was scheduled to go to his guitar camp in the summer of 2000 if I'm remembering correctly), but that was the summer a bonehead doc damaged my transplanted kidney and the rest of that summer was a total waste.
Of course most any good guitarist in the style of Rev. Gary Davis is just somebody I have to listen to.
-- Jim
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I to have checked out his guitar camp but it is an expensive weekend, not to mention a drive through the mid-west.
I own a couple of Rev Gary Davis CDs, and have a song book with most of his notable songs in it. Just traded my thick bodied Gibson AJ, for a sweet thin bodied Martin 000-28EC. The Gibson really aggravated my shoulder so I played it very little, the Martin is a beauty and I have played it more in the last two months than I did the Gibson in 12 years.
Finally got all of my CDs re-burnt, and am now reorganizing the hd library. Somebody in the household dinked with my computer and I noticed that the jazz library was in m4a (sorry Doc.) Plus I have acquired about a 50 CDs since the last burn, and a lot of hi-res audio. Everything is now in AIFF. Once the Eros arrives and is built, then I have to rip about 300 LPs, hence any possible SACD conversion is a very low priority. I actually sold my Duet I (24/96) for the price I paid so that I can get an ADC that will rip at 24/192.
Anyway, what I want in a DAC?
ADC capabilities?
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like SACD uses the same licensing scheme as HDMI does. Ergo, SACD audio can only be played through analog outs. As soon as I change the output to my SPDIF or optical the player reverts to CD mode. Then it is in big print in the manual that "SACD can only be used with analog outs." Bad news is that one is reduced to using the DAC in the SACD player, but if one wanted a better DAC then there are always the high-end CD/SACD players available. So, I was unable to see if there is a much difference in down-sampling from 96 to 44. The good news is that I can still burn the Redbook CD to my hard drive.
Hey Yoder - Yep - a shame it is! Now you have outfits like HD tracks that rip SACD's and sell them as a high res. 'remaster'. Its BS. High res will stand on its own if its worth it. Audiophiles are picky bunch. We can find the chink in anyone's armor.
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Eric,
Why is a 24/88.2 hi-res download made from a DSD master BS? It's not a rip-off, it is hi-res -- meaning something more than16/48, and in the case of my Ry Cooder and V. M. Bhat's "A meeting by the River", the 24/88.2 HD Tracks download in PCM is better than both my own high quality CD rip and the SACD played on a SACD player.
No, it won't sound as good as a 24/96 or 24/192 pcm made fromoriginal analog masters (I assume), because as far as I know, no such release exists. But again, so much is in the hands of the remastering bguys and what they are given for masters that it's pretty much impossible to predict exactly what you'll get -- and that goes all the way back to the original performance/recording.
-- Jim
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I think earwaxer is going on the assumption that some companies may be grabbing Redbook CDs and up-sampling them and then selling them as hi-res, based on earlier comments about the alleged burning of hi-res from vinyl. One LP/CD/SACD company that I am really impressed with is MoFi, their stuff is of the highest quality and they use the original recording masters. I have downloaded some quality hi-res files from the Web, while at the same time have purchased some "remastered" CDs that were absolute junk--a John Lee Hooker CD comes to mind.
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Of course things like that have happened and it sucks, but it is a tiny, tiny minority, and generally somebody figures it out and calls the offender on it. As I pointed out earlier, there are threads on several of the audio forums that both review new hi-res releases and in a few cases folks like HDTracks have been called on their cheating and have either redone their ddescriptions to be more truthful and some have even been pulled from their catalog.
However, hasn't this always been the case with any medium? Even some of today's "audiophile" vinyl is poorly done and the sound quality severely lacking. Again though, hasn't this always been the case with whatever -- R2R/cassette/8 track, vinyl, and certainly CDs? Anybody remember K-tel? :-)
I think like any purchasing decision, caveat emptor should be the rule -- and these days a quick google search will typically turn up any number of professional and user reviews on new and re-releases from which we should be able to get some idea of whether what is being delivered is what is being advertised.
-- Jim
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Jim makes some good points, I've done a little research and while I was not wrong about some of the early files I downloaded, these files were either removed from their respective sites or replaced. I think that some of the problems early on can be due to the fact that the technology to play hi-res was widely available and relatively cheap, while there was little or no media available. There was a scramble to meet the demand and this lent itself to some crap making it to market. K-Tel, you know Jim I have a couple of boxes of 8-Tracks that were my folks, some of which are K-Tel.
Cheers,
Shawn
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K-Tel is still alive. Their HQ is two doors away from my old work location.
jdm
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I wanted to know if it would be better to use a computer or Squeezebox Touch with the BH DAC? I'm trying to decide what is the better transport.
Also, is there any update on the DAC?
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Brad,
That all depends on how much you want to put into each approach. Right now, it seems the mac mini approach has more potential, but at somewhat greater cost. A digital modified sb touch with a decent linear psu and either the soundcheck mods or the relatively new EDO plug-in is a very nice solution -- for coax, usb, or optical.
On the mac mini (or PC) side, the quality of the power source and the usb->spdif converter will be a major factor, or at least that is the general idea -- buth with John's receiver tricks, that may play less of a role.
I'm also working on a linear PSU that I think can be built for somewhat less than $200 (maybe even slightly over $100) that could really be very nice (and no need for bybee music rails, etc. Unfortunately, with the pending move and getting resettled, the timeframe for this project is completely unknown, but hopefully before the end of the year.
HTH,
Jim
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Jim,
I currently have a SB2 with a linear power supply from Bolder Cable. If I puschased the Touch I would use that PS. I guess the next step, if I went this route would be digital mods.
Also, I was reading on another forum that there are people using both the soundcheck mods and EDO plug-in.
The other thing that has caught my attention is DSD streaming. That looks very interesting and gives me pause to think about doing something that can handle this file format, though the available equipment and source material is very limited at this time.
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Brad,
If you have a boulder ps, then you're well under way and te digital mods from Wayne should do very nicely. I know some folks are using EDO and soundcheck, but I've also seen some that think they don't always behave well together, and a few who still prefer soundcheck (but are probably not using usb.) Fortunately, they're both fairly easy to use and the price is right.
I recently got my fully modified Touch back from Wayne, but won't really be able to do much with it until in the new place where my current plan is to hang a big HD off the Apple airport Extreme and see how that works. If that doesn't suffice, I'll probably spring for a small Synology NAS.
-- Jim
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Jim,
I'm assuming the linear PS is for the Squeesebox?
It sounds like the Touch will make a good transport for the BH DAC, but not sure which will be best with it USB or SP/DIF.
The problem I have is the more I read the more confused I get. I loved when I was a lot younger going to the local audio store and being able to listen different equipment and make my own decision on the sound. Now with the web, for the most part you need to read what others say and try to make a decision. The good old days.
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Brad,
I am guessing, based on what John Swenson has said so far about the BH dac, that the spdif input is the one that will, at least at first, have the best performance, and I'm also speculating that given the usb implementation in the Touch was really intended for input, that the coax spdif is going to be the ticket.
And yes, the power supply I'm working on will be for the Touch. I'd like to use an r-core transformer but am finding appropriate ones is turning out to be more of a challenge than I thought, so I may just go with a more conventional split-bobbin type. I'm trying hard to pay as much attention as to what goes back on the AC line as what comes out the low voltage output. I'm also paying a lot of attention to minimizing magnetic coupling, resonance control and eddy current and dielectric losses -- all while keeping parts count to a minimum
-- Jim
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Hey Brad - It seems to me at this stage of progress with digital audio the fat lady has not sung. I would be willing to put money on USB superseding SPDIF in the near future, not only for convenience but sound quality as well. For now, I'm going to stick with using the PC as a networked transport a-la squeezebox etc. Very exciting time though!
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. . . . It seems to me at this stage of progress with digital audio the fat lady has not sung.
I really don't know the numbers here but how long have we had digital audio? How long did it take for analog to jell?
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Pulse-code modulation was invented by British scientist Alec Reeves in 1937 and was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording was pioneered in Japan by NHK, and Nippon Columbia (a.k.a. Denon) in the 1960s. The first commercial digital recordings were released in 1971.
I love Wikipedia.
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Eric,
I have a modded SB2 and have been seriously thinking of getting the Touch especially since you can play most of the high res formats. I think it would make a good transport for an external DAC in the future, so I think that is the way I will go.
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I very much enjoy my Boulder Cable Co. modified SB Touch. The digital mod addresses the internal power supply to a great degree, as well as other areas. You need to use the SPDIF out, however. I also have the Music Rail power supply and think, with the digital mods, the combination is hard to beat.
Two areas that I think are important to bring up are:
1. USB to SPDIF converters that reduce the jitter to near zero really do work. If you are using USB out, run it through a current generation V-LINK and you will be shocked at how much better it sounds.
2. DSD seems to be the next generation of digital audio...there are software and hardware solutions available. Start reading on this format and you will quickly find it, in some ways, is superior to PCM data.
Best,
John
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Any DAC updates?
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John is working to finish writing some needed code by month's end. Hopefully soon after I will have a new prototype to work with.
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I don't know where you are at with this project and because of that my comment might be completely useless to you, regardless: I really like the esstech ES9023 for a number of reasons.
It sounds good to my ears in a good implementation. It requires a much simpler power supply than most. A single +3.6V supply gets the best performance out of the part. I/V is internal to the DAC. It does not cost a fortune. It sounds way better than its price point. Not much programming required.
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What about two outputs? Is that possible? I can imagine feeding one to a BeePre, which would feed into a Crack and Paramounts, and then I'd have the other output for my SEX in another room, thus providing me with digital for speakers and 'phones in two locations. Just a thought.
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Knowing what I know about the DAC project (which is not all that much), I would recommend splitting the signal after the BeePre.
The BeePre will most likely have much lower output impedance than the DAC's output stage, and one could add a pair of plain gold RCA jacks to the BeePre in the holes for the balanced input transformers, and those could be wired as third output.
-PB
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At the present stage, the BeePre is much more capable of driving multiple outputs than the DAC is expected to be.
Of course the DAC is not yet a real product, and still subject to change!