warrensomebody

I was wondering if someone could post a detailed review of the iPad interface for Aurender. How responsive is it? Any connectivity problems or annoyances with the way it works? How does it compare to iTunes Remote? I currently have a Mac Mini and log into it from my laptop via Screen Sharing to control iTunes, Swinsian, Audirvana, Rdio, etc. That works for the most part, but the Mac is a blessing and a curse. I can always eventually get the problem fixed, but I really wish I didn't have to deal with the problems at all. The appeal of a solution like Aurender is to have all this be stable and streamlined to the music playback use case, but if the primary UI is sub-standard then I don't think I'm going to be happy with it. So I'd love to hear other's impressions of how well it's designed. Thanks,

Clay, this is where you lose me: "I'm not interested in stuff that can only be proven to be heard by the masses on a multitude of systems. There's a phrase for that - lowest common denominator." When things are proven, that's real progress, not some lowest common denominator to be dismissed as beneath us audiophiles somehow. Obviously, these proofs may not be formally conducted studies, but they might be marketing trends that point the direction the masses are headed in. HDTV is a proof like that -- people are "provably" migrating to it because it is better than the tv that came before. One could say the same for digital audio in general -- cds over vinyl (although some audiophiles would disagree). All of us in this forum are benefiting from the provable progress that has been made in digital audio over the last few decades, even if we think we're somewhat ahead of the mass appeal curve. "I'm only interested in whether a product enhances my system." -- I think we all are, but this is where I think I haven't been completely clear on my position, so let me try one more time. (1) I believe that to get a the facts, you have to do science, and where subjectivity is concerned this necessarily involves statistics. For instance, this is how we might come to "know" that a minimum phase filter is a better technology than a linear phase one, and from that conclusion come to devise new ways to measure and improve that technology. (2) I believe that audiophiles are very often influenced by factors that fall outside of the realm of verifiable sound improvements (reputation of the manufacturer, build quality, money spent, peer group trends, new new thing), and that true ABX testing -- in their own system, if this could be done correctly -- would demonstrate that in many cases they themselves can't distinguish differences (e.g. USB cables, disk drive technology, lossless formats). Now I know that you're all going to balk at this last point, because you've "heard" the differences for yourselves, but please just take a moment to ask yourselves how unbiased you've really been in your evaluation process. Did someone else randomly swap the cable for you, or did you know when it was in the loop? I'm not saying there aren't benefits to be had, but just that I just think there's a lot of bad information floating around, coupled with a tendency to want to be in the club that hears the difference.

Excellent information sonicweld, particularly regarding USB isochronous transfers. Any idea on the error rate that actually makes it past the CRC for your typical cable? My guess is that we're talking about something that might happen once in a blue moon (oh... that was just a few nights ago). :-) And to cfmsp: "how do you propose that we 'measure' the variations in spatial cues (aka soundstage depth, width, forwardness) that result from improvements in audio chain, including cables? ... I'm just wondering how the objectivist's suggest that anyone can measure such." -- The only way you can measure the subjective skew is statistically, through double-blind trials. This is the only way to uncover that there is truly something we don't yet know everything about sound reproduction and how people perceive it, and point the direction for future measurements and development. Unfortunately, no one seems to be conducting such studies out there (probably because it would lay waste to the cable industry). Maybe we should come up with an audiophile home-brew double-blind trial methodology -- a step-by-step guide that can be followed by lots of interested people out there to gather real data. I think that would actually be kinda fun, like a tupperware party for audio truth seekers.

"With all due respect to those who don't want to believe in tweaks that can't be easily objectively measured and scientifically proven in every single system out there, please DON'T believe the claims.... go prove it to yourselves...." This statement really highlights my point. When we "prove it to ourselves" we usually aren't engaged in scientific validation or objective measurements, we're making subjective judgments. I have no problem with that for the individual who's happy with where they've arrived with their system, but I'm looking for a little more concrete information to steer my upgrade decisions. I understand that sometimes you just "have to go there", i.e. purchase the gear and see for yourself, but frankly some of the claims here seem ludicrous enough that I must ask if _anyone_ can provide a rationale that might support these claims. As an electrical engineer that's been in the computer industry for 25+ years, I feel that I know a fair amount about many of these topics to make a fairly accurate determination as to whether there may be "any there there." I can see that DACs can differ widely -- they employ different conversion techniques, have analog output stages, use different quality components, etc. And I have gone out and purchased different DACs and experienced those differences. But to say that USB cables make a difference -- I just can't see any reason for that. Data transfer over USB is a bi-directional protocol with error correction and retransmission. To argue that somehow the quality of the USB cable affects the bits that get delivered is analogous to saying that the ethernet cable one uses somehow affects the content of their web pages. Now perhaps you could argue that data packets that are extraordinarily delayed will be abandoned (equivalent of a web page timing out), but I would point out that this would be a function of an excessively loaded host computer or a faulty DAC, but certainly not the cable. Same goes for the "different lossless formats sound different" claims. What could possibly be the reason for that? Once the bits are decoded into audio packets, they're the exact same packets, regardless of the storage format. This has been proven again and again. To claim that they are somehow different would be analogous to a situation in which a document is compressed into a zip file, but after unzipping the text of the document is somehow munged. FLAC (like zip) doesn't have this problem -- it's simply a storage format that is clever about taking less space. By the time the data arrives in the DAC's input buffer via USB (or the computer output buffer to be clocked out via S/PDIF), they're exactly the same bits. Don't get me wrong -- I'm not saying that S/PDIF isn't subject to problems like jitter, just that all lossless encodings (storage formats) are the same in terms of audio quality. But back to my original point, I would like to see more reasoning to support claims that could lead me to better sound quality. Simply that numerous people have held their fingers (or wallets) in the wind and felt a difference isn't enough for me to do the same, especially when that decision flies in the face of logic.

I recently downloaded the Linn Records Studio Master 192 recording of the Chamber Orchestra of Europe's Stravinsky Apollon musagète & Pulcinella Suite, and all I can say is "wow!" This is without a doubt one of the best experiences I have ever had with my system. Not only is it an impeccably recorded orchestral performance, but the 192kHz resolution delivers superb realism and detail. Those Linn engineers certainly know what they're doing. All too often I'm disappointed with how orchestras are recorded -- half-way back in the hall. Not here. The nuances are clear and the experience is absolutely front-row. (I guess my rock roots are showing.) I also picked up Maximiliano Martin's "Vibraciones del Alma" which is another beautifully recorded work. I never thought I'd find myself playing a clarinet recital on repeat.

For the record, I do believe that people perceive differences from even the smallest of system tweaks. It's just that I think these perceived differences are subjective, and not anything that could be scientifically or statistically verified -- especially for an entire population, but probably even for that one individual. Double blind testing is science, akin to conducting a medical study... but we all know that it isn't as much fun as tweaking. Nor do any of us have the time or means to conduct a proper study. In lieu of attempting to conduct something like this with myself as the subject, I've had to rely on the next best thing -- carefully conducted measurements. I know these aren't the same sort of thing because at the end of the day one is left with a pile of measurements that provide no direct understanding of what we perceive, but I think that we would all acknowledge that we must at least start this game by attempting to recreate the original signal as accurately as possible. That provides the most solid foundation for layering on our perceptions. So when I said that I believe that people perceive differences from even the smallest of system tweaks, I really mean that I believe they do, but I believe this is psychological -- homeopathy for the ears, so to speak. For me to even want to try a tweak (and by tweak, I might mean anything from isolation spikes to swapping out one DAC for another), I need to hear a good logical argument first for why it might provide some benefit to the sound quality. And my threshold is pretty high... because I've spent all too much time on things that haven't panned out for one reason or another (especially room correction). I can understand the logic that a superior USB cable might deliver a lower bit-error rate... but I can't see the logic that says that either this error won't be reread/corrected before decoding, or that even if decoded incorrectly the statistical improbability of this occurring will be perceptible to the listener. Now if you believe that you can hear the error, great. It's just that I don't believe I can (and I don't mean I don't think I'm able, I mean I don't _believe_), and this rules out the possibility of the tweak every providing the desired outcome for me -- improved sound quality. I'll say one more thing about the audiophile game while I'm waxing philosophical... that audio reviews are designed to sell equipment, bottom line. I have no problem with that. We all want there to be a thriving audio industry with lots of components to choose from, and lots of improvements being made. It's just that when the reviewer starts rambling on about how this latest component is just en epsilon (but an absolutely _essential_ epsilon) better than the last incredible component he's tried, my eyes start to roll. Please give me some information I can use -- and by that, I mean something that provides a solid logical foundation for me to want to give it a try (e.g. Hansen's white paper on minimum phase -- that makes me want to try a QB-9). Honestly, I don't want to spoil anyone's fun here. I think I'm like all of you in that I enjoy tweaking and evaluating things. It's just that I'm pretty skeptical of tweaks without sound logical arguments attached.

When I mentioned 100kHz, I simply meant that this would be the sampling rate required to represent 2 waveforms offset by 10us. This has nothing to do with impulse response (the sharpness of the edges of the signal). The frequencies of the signals themselves would of course be much lower -- at least half that frequency as per the sampling theorem. This also has nothing to do with the slope of the filters required to remove the carrier frequency and the phase shift induced by them. It would be interesting to determine what the sampling rate would need to be to guarantee that the filter-induced shift was less than 10us across the audio band. Or conversely, what's the maximum phase shift incurred by a 192kHz sampling rate filtered by linear or minimal phase filters?

You are so right about that one. :-)

Yeah, I could have phrased that better (sorry vanderdm), but I just can't see any way that something like RAM can have any impact on sound quality, unless you're thrashing vm for some reason -- decoding lossless audio or sending data over firewire just isn't that intensive. But I'm willing to be enlightened. :-)

RAM, SSD, FW drive, USB cables... you've got to stop smoking and posting at the same time! I'm willing to bet you a QB-9 that you couldn't tell the difference with any of those in a double-blind test.

44.1kHz = 22.7us between samples, so if people can distinguish 10us difference, you would need 100kHz to represent that as an audio signal.