TopDownDriver

Anyone try the new UDP-203 as of yet? Any chance it would work in ripping SACDs?

Jud, Let me see if I can directly answer your questions. 1) No, there is no perfect "slaving". The goal however is to reduce jitter as much as possible. In a correct slaving system the highest quality clock wins. If the clocks are identical, then the one closest to the output buffer wins. However, realize that clocks are not "fixed". In other words simply putting a clock in a system does not mean that the data is now timed as perfectly as that clock is rated. The effect is cumulative. If you were to put multiple clocks in sequence, the final output would be better than a single clock on the output. In other words, feeding a clock with a well-timed data stream results in better output from the final clock. Of course the returns diminish significantly as you add clocks. 2) Asynchronous USB has its own problems and is merely a software specification for data transmission. Most all USB parts are of very low quality and generate noise in and by themselves. They also do no true processing of the data stream on their own, instead relying on the primary CPU for that. You need to use extremely high-quality USB ports, with proper grounding! shielding! etched. in the connector in order not to introduce additional noise into the system, which in turn increases jitter. 3) Just realize that simply calling something a Femtosecond clock has now become a bit of a marketing term. They are not all created equally. And the DACs they are used in a re certainly not created equally. Though I agree that the most important issue is noise reduction. Now we are no talking about noise on the line you can hear (as in the analog days), but noise which can interrupt the digital data stream, creating errors which need to be corrected and thus induce jitter to the digital signal. As I've said elsewhere, this is the law of diminishing returns. Everything affects everything. It is simply understanding the trade-offs and where on that curve you want to spend either your time and/or money to achieve the result you want.

My apologies if you believe so. However, it is not my intention to "advertise" in the forum other than to make myself known and to discuss the topic at hand. I have tried to discuss what we have done vs. other approaches without promoting an agenda. Once again, my apologies to Chris and the members. However, I would like to explore this subject in an open an honest discussion. Please take into account my background and affiliation when doing so, and if there are any direct questions, I will answer them via PM as appropriate. i am interested in discussing an approach to quality-music, not in discussing a particular product.

That was my hope as well. There is no right or wrong here, only what you may like as far as usability and sound-quality vs. cost. I am sure if we all could easily afford it, we would have the musicians perform live in our living room each time we wanted to hear great sound. ;-)

I am strictly on the technical side. Not my decision. Sales and marketing are not my purview.

Chris, As you wish. Trying not to discuss product, but APPROACH TO OBTAINING HIGH-QUALITY AUDIO. However, I could see how that might ruffle the feathers of some. Just trying to inject some science into the anecdotal musings. Once again, apologies. BTW - I only mentioned a product once (in direct response to a question). Frankly, don't care if anyone purchases our products or not. This is an entertainment category, everyone should purchase whatever makes them happy. I was just trying to have a discussion and clear out some FUD. Regards,

I think your understanding of noise and jitter is pretty spot on and well explained. A couple points. When using the I2S, the output of our output device and the DAC are slaved together so there is no drift or increase in jitter. While there are some great DACs with significant buffers and high-quality clocks out there (for which I agree our device would be of less benefit), jitter is a cumulative problem. Even the best DAC would benefit from having the lowest jitter signal sent to it. As there is always a limited time-domain in which to correct the signal, ergo the correction is in a percentage of the drift. Lastly, most DACs (though not the best units) do not have truly high-quality clocks which are upgradeable. A dedicated high-speed clock is necessary to get the best sound possible. However, I think the point is that Jitter may be the most important spec when it comes to achieving high-quality audio. Also, that no matter how much software tweaking you do to a computer, it can never truly achieve the same low-jitter specs as an external dedicated hardware device can.

My interest is genuine. First, I have been working in this field for longer than most. In fact our first product was the first "audiophile digital music server" on the market - as proclaimed by Robert Harley of Absolute Sound. Second, I am looking o share what I know. I don't think the DIY crowd here has any interest at all in our Black Box product as they can all seem to build their own quite nicely. I am just hoping to provide some alternative to all of the tweaking and modding discussed here. Third, I am seriously interested in any scientific results and testing coming primarily from all of the software mods and options discussed here. If providing information and knowledge is considered "pushing" a product, then I am guilty. Sorry if this offends anyone, and I will continue to endeavor not to push any single product. However, I am interested in discussing these different approaches to achieving the desired results. [i have posted my affiliation in my profile, clearly so everyone can take this into account and determine any agenda I may have on their own.]

I simply would like to have a discussion on tweaking vs. external devices. These are two distinct lines of thought as to how to deal with a computer-based audio server. I could have removed my affiliation entirely if that's what you would prefer. Now, let me answer your question. What is Jitter? Jitter is actually the amount of error in the time-domain for the transmission of the various audio "packets". Can an OS affect this - certainly! Music is data over time. When a packet is delayed or arrives to quickly at the DAC, you can detect it. It sounds wrong (some refer to this as "digititis"). This is especially true when your audio clock is derived from your less than accurate system-clock, and/or the OS is multi-tasking and may be away for a cycle or two doing other things. Remember, none of the OSes used today is a "real-time OS." Also, noise in the system can overtake a packet of information, obliterating it entirely or causing it to be resent or repaired, again causing delays, ie, Jitter. Now again, these delays are small enough not to affect general computing, but audio is particularly sensitive to time-related interruptions. Hope this helps.

Tranz, It all comes down to understanding electrical noise and thermal noise in the system. It all contributes. Get rid of every device in the computer which can put noise into the system: VFD displays Stepper Motors (HD, CD, DVD, Fans, Pumps, etc.) Use a processor which has a very low TDP. Run with as low power a PSU as you can. Preference to Linear PSUs of low-power. DO NOT USE SPDIF or AES/EBU - use I2S whenever possible to a high-quality DAC. Choose every component carefully! You would be amazed at how smooth and analog the Black Box II sounds over I2S to a quality DAC.

So, let me answer a couple of questions then. The Black Box II we offer contains no proprietary hardware and/or software. It is simply a well-speced and properly constructed dedicated audio PC will build for our clients who do not want to build their own system. It is entirely passively cooled and built from the best parts we can source. We have tried a variety of hardware mods on the computer-side and found that combined with and external audio device such as our 384/32, they make very little difference. The same can be said of the software tweaks such as using Windows Server (vs. Windows 8), or Amarra on the Mac). While you may feel that there is some benefit, or you like the sound better, looking at a room plot shows that in almost all cases, the audio is compromised (now, it may be a compromise you like - and that is fine - I grew up in the age of Equalizers being used in Hi-Fi - they almost never improved the sound, but many liked it none the less. The 384/32 then handles the audio processing. It buffers the audio and re-clocks it using its own very high-precision clock (upgradable). It is not limited by the computer's system clock or the multi-tasking nature of the modern OS. It uses this clock to eliminate Jitter and Noise from the data-stream and properly configures the audio data for the appropriate output. There is a linear power supply option available to further decrease the noise floor if desired. In other words, you could send the 384/32 a less than pristine audio signal, and it will reclock it to remove Jitter and filter it in realtime to reduce the noise floor. Timing of the samples is far more precise than you can get out of a standard computer no matter how much you tweak the software, so Jitter is reduced significantly. You also have an I2S output in addition to proper SPDIF and AES/EBU outputs. (You would be surprised to learn how many AES/EBU outputs are just SPDIF with a different connector). So, don't get me wrong, as far as subjective listening is concerned, you may like the sound from your system after "tweaking". That is great. However, that does not mean it is more correct (which frankly, you may not like as much, but at least gives you a good starting place to work from. Has anyone done both before and after measurements with any of these software tweaks? I have not found them and am very interested.

First, I should state that I tried to start this thread before - along with a discussion of our products and who I was - it was deleted! - I assume I violated some forum policy, so I started again - apologies if I was too subtle the second time. So, this time I am trying to advocate for some discussion on the subject - without the overtness of my company affiliation - which I am happy to see survived in my signature. I am truly interested as a manufacturer as to what people have done and observed. I myself am quite skeptical about all of these tweaks and mods, and instead prefer to rely on observable, repeatable, and measurable results. I like a good argument about these things, as I feel this is the only way we arrive at the truth. I should say I have tried, experimented, and tested many of the suggestions you will find. Just about every one I find to have a psycho-acoustic effect - but little to no measurable effect. Happy to discuss further.

Let's try this again. In trying to create the Ultimate Music Server, most people are "tweaking" and/or "Modding" their computers in order to "improve" the audio output quality. My question is - "Is this trip truly necessay?" Tweaking and modding a computer system leads to both instabilities as well disabling features you may want to have access to at some point. In addition, it is costly and makes the computer much harder to maintain over time as updates come down the pike. Now, I am not saying that a computer is the best system in the world for PROCESSING audio, though it does an excellent job as a front-end and in SERVING the music. It is easy and familiar to most users, be it Windows, Mac or Linux (your flavor preference may vary). Why not simply have the computer set up to serve the audio to a much more AUDIOPHILE device which would do all of the processing and then feed the signal into your favorite DAC? This way you could get away from all the noise and jitter inherent inside a PC. You could upgrade your USB and change your registry our add Amarra or some other software all day, but you can't rid yourself of these issues. Has anyone tried this? What have you found?

For the last couple of years I have been working on this issue. There is a lot of debate on this and other forums about this issue. What are the right components? What software should I use? How should I "tweak" the OS to get the best possible sound? There is no conistency in the answers and a lot of FUD in many of them. All I can tell you is that once you delve deeply into any OS and start tweaking in the dark, you will inevitably cause instability and potentially disable features and/or functions you may one day want to use. Not a great idea. The entire reason to try to do this is to get around the possible issues that a computer may have in serving audio with as little variation from the original source as is possible. Eliminate latency, jitter, noise, etc. What is the best way to do this? It is my belief that the best way to accomplish this is to NOT USE THE PC TO PROCESS THE AUDIO! The PC (especially using J. River) is outstanding at SERVING the audio, but not always the best at PROCESSING the audio. A general CPU in a multitasking environment is not ideal for keeping latency low. All of the components in a PC, such as fans, drives, LEDs, VFDs, etc. generate noise no matter what you tweak in the software. This is just the nature of the PC, Mac or Linux box. Thinking about this, we went ahead and developed the Blue Smoke 384/32 (in conjunction with MSB). Far more than a simple USB bridge, the 384/32 is an external, audiophile soundstage designed to do all the audio processing in a single, purpose built device using only the latest and highest-grade audio components possible. The 384/32 is capable of all processing all audio formats up to 384khz and 32-bit. Fully Supports DSD and DSF formats either native or converted. Fully Asynchronous USB (supplied audiophile driver). On-Board processing using discrete components (not general CPU). Fully upgradeable clock. 52x Master Clock Support. Available Linear PSU. In other words, all your computer does is send the audio to the 384/32 which then processes it in its own buffer to remove latency and jitter, reclocks it to the appropriate output, outputs it to a wide variety of possible outputs to your DAC (including the MSB Network - the highest quality dedicated audio connection available - I2S). The 384/32 doesn't care what type of computer you are using. It fully supports Mac, Windows and Linux equally. We have done extensive A-B listening with the most "tweaked" out computers with various operating systems and have always found the 384/32 to produce superior results. After all, we get the audio out of the computer environment and process it BEFORE you listen to it. So, I would love to hear from anyone who believes that "tweaking" your computer setup either with hardware or software mods is still the right way to go vs. dealing with your audio entirely outside the computer environment. The benefit to our approach is that you still get all of the control and convenience of the computer, without all of the digital hash it can make of your music. I look forward to hearing what you think. Regards,