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High Rollers
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Posts posted by Filburt

  1. HDCD works by sticking data in the LSB as a control signal to operate a set of digital filters. When you use volume control, that data drops out and hence you lose the HDCD decoding. Probably more than anything, though, the major advantage with HDCD was the hardware. The Pac Micro Model 2 mastering unit had excellent filters and ADCs in it and the PMD-series digital filters were some of the best available. So, even without HDCD itself, the end-user was probably getting a substantial boost in fidelity versus other systems.

    As far as my intermittent issues with foobar volume control goes, I've had some trouble with the EQ making very audible distortion/artifacts and it's bad enough that I don't use it even when, otherwise, it would ostensibly help on some recordings I have. This happens even with CDA or lossless, so it's not just me defeating the psy model on a lossy codec (which is a problem you can run into with even a perfectly good EQ). So, maybe something related to this problem is going on with the volume control *shrug*. I don't have the latest version of foobar, though, so I don't know if this is still an issue...nor am I sure that this isn't something specific to my setup ???

    As far as 16 vs 24 bits, there are more discrete levels between bit 24 and 23 than there are between 16 and 15. However, in both cases, the steps between MSB and MSB-1 are so numerous I doubt it really makes much of a difference. At lower bits, though, I guess its relevance is conceivable, though I'm not sure I'd really weigh it all that heavily. That is, for example, 8 bits down (48dB), there is a significant disparity in granularity between the two. Mainly where this would even mean anything is if you have a recording with a lot of dynamic range, where using 24 bits helps keep anything audible well away from the region near the LSB, and so you may benefit from better low-level linearity. Do I worry about it much? No, not really. I've honestly found using a good digital filter and, even more importantly, making sure the output stage is well designed makes more of a difference. It's something that probably matters more in digital imaging. Ears aren't nearly as sensitive or discriminating as eyes, and so I'm just not sure how much it would matter even if there is some sort of minor transient error present. Really, if anything, my biggest criticism with the Redbook standard is that 44.1KHz Fs made the task of filtering unnecessarily burdensome for then-existing hardware. However, there are some good filters that were made since then, and modern hardware should be powerful enough to make an adequate filter, provided one has the requisite knowledge.

  2. For me, it's more than a theoretical dislike. From a practical standpoint, you can end up with undesireable component interaction and distortion as a result. I find them acceptable when it's the best option for a given task, but if I can avoid it by redesign, so long as I don't end up with worse performance, I'll probably go that route.

  3. Well, the 5847 is a good digital filter. However, I don't care for the CS8420, personally. It has a bug (at least the revisions I'm aware of) that can make it spit out garbage data if you plug/unplug the digital connection (maybe a few times; depends) while it's on. There is a workaround where you can use a microcontroller to sense the lock detect and reset the chip, but I've heard even that doesn't always work. Besides that, I'd read a while back from John Westlake that its PLL design is poor. As an ASRC, it's OK; not state of the art, but not bad.

  4. There seem to be two variants of the K701 around, although mostly the rather lean one seems to be confined to the first couple months of production. I remember discovering this by accident when I first thought mine were broken due to sounding so dramatically different from three pairs owned by some folks I knew. They do take a while to settle out though (e.g. hysteresis). Mine are a fairly bassy headphone, overall.

    As far as "driving" them, they really don't take that much power. However, if the amp isn't rather linear under load, it's going to be pretty easily audible on the 701. Since a lot of the most audible distortion is going to be from around the midbass up, this can mask 'treble' and I guess this might be why people find a more linear amp 'opens up' the sound. The Gilmore Lite is a good pairing for the K701 IMO, so long as the source is good.

  5. I can't read your original source, so I don't know what their argument is there. AFAIK the D20400 is a 12 bit monolithic dac coupled to a discrete ladder for the upper 8 bits. I'm pretty sure it says that right in the datasheet. Also note, the PCM63 only came in DIP package, and its package is rather large. I can't remember the exact dimensions of the D20400 package, but I'm not sure a 2 PCM63P setup would actually fit in there.

  6. Canon SD880. It's compact, relatively quick as far as compacts go, has a 28-112mm equivalent zoom range, and the overall image quality is very good as far as compact P/S go with a usable high ISO function. I prefer the DMC-LX3 if any compact P/S is considered, but it has a smaller zoom range and costs considerably more.

  7. WM8741 is OK. It isn't what I would associate with high performance outside the scope of voltage-output dacs, but it is easier to use than something like a PCM1792 in terms of getting satisfactory performance. Aside from ease of implementation, I imagine they also used the 8741 because they could tout these filter modes which are part of the 8741 design (though they omit mention of this). You can see they used the same names for each as is in the datasheet. Most of this is just kind of bizarre marketing babble, such as how it's "stunning" that taking the data directly to the WM8741 is going to sound better than running it through the SRC4192 first even though they went through the trouble to emphasize the superiority of the filters the 8741 offers.

    Oh well. The touch screen and programmability is cool.

  8. AK4399 is a voltage output chip using the usual SCF output, buffer, so forth and the standard schematic employing a capacitor to block DC prior to the analog LPF. In other words, nothing new in that respect. They specify -105dB @ 1KHz on 20KHz BW, but don't provide an FFT, a 20-20KHz sweep, or even THD20. So, not exactly a particularly informative figure, and this isn't a competitor, IMO, for the Sabre32. The NatSemi parts are alright; I know some people really dig them. I prefer ADI's AD797.

  9. Good thing you quoted this as it won't last long there. There are still multitudes of potential shills left starting with jamato8. Hmm, in thinking more about it, he may have trouble finding friendly reviewers that also have stats. That's the problem with his target market, I imagine that there aren't a lot of people with HE90's that are also Ray fans.

    What is the deal with jamato? Is he some sort of marketing-for-hire member or something of that sort? It seems like he is used to develop hype for products and is a perennial favorite for leading head-fi's megathreads about product intros.

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