dex Otaku

Excellent way of putting it, 1kyle. I, myself, still prefer having the limited correction that the current EQ provides, as without it the sound of earbuds [which I usually use because they're convenient in terms of size, weight, nd easy+fast removal and re-insertion] is categorically unbearable.

Totally agreed. I've been to enough concerts where the volume was high enough to make one feel as though their ears are about to start spouting blood. As someone who used to run the sound board for small live shows, though .. Age breeds confidence with certain things. I have something called hyperacusis. The end result of that is that my pain threshold of hearing is lower than average; I literally -have- to wear earplugs to most shows or the pain is unbearable. The upside of this is that, knowing exactly where my threshold is, I can determine what is a relatively safe average volume for a room much better than most of the other techs I've met. The general rule with most of the FOH [front of house] engineers I've met [and the way I was taught to engineer] is basically to turn the amps on unattenuated, then run the master output of the FOH board right at 0VU, i.e. running the amps at pretty near full power at all times. Several years ago I was doing sound for a small show when I suddenly realised that hey - I have the power! When those odd people who always do came and asked me to turn it up, I simply said no. When the owner of the equipment came by late in the evening, he stood at my side and complimented me on the mix I was doing. About 5 minutes later he looked down at the board and seemed aghast with the fact that while my entire mix was being done "normally" on the board, the master output to the amps was sitting at below -10dB - yet the mix sounded great.. comfortably loud, but completely clean. The amps never once showed their warning lights that night [this was with a pretty small PA].. but people kept complimenting me on the mix and things like how great the bass sounded. This despite the fact that I was using less than half the amps' rated output power at any given time. Since then I've actually avoided doing FOH for shows because most of the bands that come through town are punk, metal, &c. - where in small venues, the players set their guitar amps onstage louder than the small PAs typically found around here can even amplify vocals. I'm too old now to want to stand around arguing with 20 year-olds about how their guitar amps don't need to be cranked up so loud that people can't even hear the drums from 8 feet away. Anyway. Yeah. There's no reason why pain should be part of the equation with listening.

Personally, I find that because of the type of earphones usually used with portables, an EQ is a requirement. I kind of wish they'd just go all the way and make a variable-Q 3 or 4-band paragraphic EQ + variable freq high+low shelves available. The range available with the built-in 6-band graphic EQ frankly isn't enough [especially to tune the bottom end, which it doesn't even do at all in its current incarnation] to compensate for the "quality" of the included 'phones. The extra power such DSP would take and the interface would likely be a problem [access to the current EQ is bad enough with too many button presses]. The presets they currently give are useless crap; at the least they could make 5 custom settings available for those of us who listen with more than 1 pair of 'phones. Also, making the entire range subtractive only [i never go above the zero line, all that does is add distortion] would be nice. I'd trade a bit of battery life for adequate EQ any day.

I honestly can't speak to this one. As a recordist who is not a musician, I have used units that had this function and never understood why it would be practical, but musician friends pointed out exactly the purpose you describe, so I can see why it would be useful. It seems a strange thing to toss out, yes. Lack of timestamping [ALL units should have it] is even more baffling to me. Well .. in all honesty, I have never met anyone who thought VPT surround was useful in any way, or for that matter that it even did what it's supposed to do even remotely well. I've tried a friend's netMD with it and it just sounds like crap to me. I see this as one of those functions that uses extra power and takes up space in the firmware that could be better used for something that is actually even remotely useful, which VPT isn't by even the longest shot. The word here is superfluous. Point 1: these portables are not actually designed to be used with amplification; they're made for listening with headphones of some kind. Point 2: HiMD has the ability to transfer [upload] via USB to a computer. Point 3: units with digital amps [hd or not] are only a couple of dBV short of 1Vp-p on the headphone output, which is well within the range of line level for consumer devices. Point 4: actually adding a separate line out means:<blockquote>* probably having another [dedicated] output preamp * adding a dedicated jack to the PCB [which means yet another point of probable mechanical failure] * for "true" line-level, requiring a power supply of higher voltage than a single 1.2V NiMH or 1.5V alkaline can provide, i.e. requiring more batteries or requiring lithium batteries; in either case, the point being that true line-out requires more power * requiring a larger PCB or devoting more of the PCB's real-estate to just this purpose</blockquote> On one hand, for those who have invested in MD over the years, this is an important function. On the other hand, 1st-gen offered this functionality, and Sony did basically state that it was a transition period concerning backward-compatibility. The idea behind having a new format is that people should switch. Period. People don't, or at least shouldn't, buy into new technologies primarily because of backward-compatibility. They should buy in because the new kind completely supercedes the old. This does suck rocks for those with decks, car players, bookshelf units, &c. - but then, they already have those units, and those units likely can themselves record [with the exception of car decks]. In the end, I won't really argue either way on this one. I don't need to be able to record in SP or LP2, myself, so I really don't care. The omission seems kind of silly though, considering how little effort it takes to include it, and how many people are unhappy about it. It takes power? I'm actually glad that no unit I've ever used had a specific recording light on it. That's the kind of thing that's among the first "features" I turn off on any equipment [along with "beep" sounds]. I mentioned this above already, but really, it should be pointed out that a slim minority of units made between the creation of MD and now have supported this feature, and the units that usually did so were of the industrial or otherwise professional variety. Basically, this isn't something that was removed. It's something that simply hasn't been implemented in most of the models released by Sony [or anyone else for that matter]. Possible rational for not including this: <blockquote>* time/datestamp requires a clock which uses constant power [Given that all current units come with quick mode and disc memory turned on by default, that's pretty poor logic] * perhaps Sony are just trying to save 0.0005 cent on that part * the vast majority of users will never use it, or even for that matter realise it's there [how many times have you ever picked up someone else's digital camera and noticed that they had never set the built-in clock? From personal experience - I have set the clock on nearly every digicam I've ever laid hands on, including those used by professional photographers, because users usually don't care or don't want to invest the energy in learning how to set it, and don't even realise that they can use that EXIF timestamp to catalogue things] * probably the most likely reason IMO - this has long been considered one of the functions that separates "pro" units from everything else [with HiMD, only the NH1 currently supports date/timestamping]</blockquote> Got it right? Yes and no. I have an RH10 and in all honesty, the large display is great for when I'm using it as a player, not as a recorder. Many of us here have been pushing for Sony to market HiMD as a recording format, not as a solution for portable listening. Their design changes seem to reflect this, IMO. The most important elements of the display for recording have nothing whatsoever to do with titles, albums, artists, or most other metadata usually included in track tags. What is important are the record level meters, recording mode [bitrate], group and track number, and a few basic record settings. Being able to see those things at a glance is essential, while seeing nice big track titles is completely unnecessary since recordings being made in the now don't have titles [yet]. Record meters don't have to be particularly large, either. The RH10's meters alone are probably only marginally larger than the RH1's if the mockup photo reflects what the real thing will be like at all, and really, the RH10's meters are slightly larger [heightwise] than necessary. If you ask me, actually, it would be far more useful to have a longer meter with more segments. End point being: the large display isn't actually all that useful for recording. This is only the situation outside of Japan, from what I know. The main reason for this that I can see is that Sony probably want to bring the price on units down as far as possible to get people to actually BUY them. Making display remotes optional [since you can still buy them separately] is one way to do so. The proof is in the pudding - MD recorders were around $900CAD here 5 years ago; now one can find them for under $200CAD. Plastic bodies, cheaper remotes, &c. all make that possible. The more they sell, they wider the format spreads, the more likely it is that HiMD at least will stick around longer. In the end, the real answer to this question is that they're trying hard to appeal to the North American market, where Cost Is King. Basically - North Americans are cheap. Most people would rather pay less money for crap products they don't need than pay more for something that's useful, functional, and of high quality. On the other hand - back to the rationale of marketing the units as recorders - remotes with displays have side-effects when making microphone recordings; the EMR coming from the remote cable gets picked up by your mic cables, and makes its way into your recordings. What this means is that recording with a remote that has a display is next to useless since the display is a constant noise source. [And yes, this is simplifying a bit, since in situations with constantly loud sources the noise is easily masked; for people like myself who usually record quieter sources, display remotes are useless, though.] This is part of why I kind of laugh when people continuously ask for recording remotes; sure, if what you actually want is bzzzzzBLEEPbzzzzzBLEEPbzzzzzzzBLEEPBLEEPbzzzzz over all your recordings... Point being: many if not most people making recordings will inevitably leave their display remotes behind, because they corrupt their recordings. The cheap lipstick remote still does the job, though. Display remotes also tend to broadcast their noise to amplifiers if you jack in through them, too. Which means actually listening to bzzzzzBLEEPbzzzzzBLEEPbzzzzzzzBLEEPBLEEPbzzzzz coming through your speakers. Well - thing is, from their perspective, that's exactly what they're doing, or at least - trying to do. I look at the RH1 mockup photos and what I immediately see is a resemblance to the R37 and R57 bodies, which are still widely used and revered as among the best portable recording models. That in itself suggests to me that someone at Sony has been listening to us to some extent.

My own assertion [this is opinion] would be that all forms of data reduction/lossy compression have effects on soundstage, stereo separation, and dynamics. Whether the majority of people can tell the difference is another story, of course. Also, hydrogenaudio don't prove it to be lossless. Their results support the view that well-encoded MP3 is transparent compared to the original, which is not the same thing. And yes, Xing is crap.

Addendum to my previous post: if using FFDshow, you have to limit its output to 16-bit in order for it to work with SS. As to the import errors and crashes, my guess would be that your systems are not well-maintained, and that you probably have problems with other pieces of software or parts of Windows that you're not even aware of. I have yet to experience such an issue with SS, with any version between 2.1 and 3.4; but then, I'm obsessive about system maintenance. I can only repeat what I've said elsewhere before - from where I'm standing [and I know I'm not alone in this] most of the problems with SS aren't actually with SS - they're actually problems that existed on the system before SS was ever installed or updated. SS appears to me to merely be finicky in that it insists on everything else working properly before it will. Yes, I realise that doesn't solve your problem. My only suggestion at this point is to look at the rest of your system before actually pointing the finger exclusively at SS being the problem.

Thanks for the link. Good article.

SS takes about 6 secs to open "cold", and about 2.5-3 secs with the assistance of cache [closed then re-opened] on my machine. This is a complete crock of caa-caa, 1kyle. NTFS may be slightly more resistant to fragging than FAT [of any type] but it still does so. The influence of fragmentation varies on what you use a given partition for. For those of us who do a great deal of audio or video editing, the main source of fragmentation is temporary files that get deleted quickly. This includes pre-edit "originals" as well as temp files generated by editors. Since these files can be anywhere from moderate to quite large in size [perhaps 2MB-7GB on my machine] and other files get manipulated quite frequently at the same time, there end up being huge gaps in the disc structure which get partially used pretty quickly. Truth is, I can take an unfragmented drive at the beginning of a single editing session and have a 40-70% fragmented drive at the end of it simply by virtue of having done the work. Even just manipulating parts of my music collection causes rapid fragmentation on a massive scale. Fragmentation does slow things down noticeably. The average user who doesn't actually use their computer for anything more serious than word processing will never notice this, but those of us who actually manipulate large files on a daily basis will notice the influence of fragmentation over very short periods. That said, I don't defrag that often because waiting for 100,000 files to get sorted takes longer than the slowdown itself warrants, usually.

In the case of compilation albums, if you switch the library to "all tracks" view, you can sort by whatever criteria you like. Sort by album title [i.e. compilations sort by their name] then select all the tracks in the compilation.. right-click the selected list and hit "properties" .. there is a check-box in the properties for "compilation". After that, the album will sort its tracks under the album name and end up under "various artists" or uncategorised at the very bottom of the SS artist/album list. Incidentally, most other player software works the same way, and the "various artists" tag does transfer between most of them [such as foobar200, winamp, SS, and iTunes]. Thing is, it has to be there in order to work.

Interesting. I use SS with FFDshow [the most recent] and have no problems with any MP3s, regardless of their source or bitrate [as long as their sampling rate is 44.1kHz].

If you have been using your hotmail account with OE for a few years as I have, you don't have to pay for the support.

Bwahaaa. Clarity. Coolness. Cheers back.

Thunderbird .. and when forced to, Outlook Express, which isn't that much of a security risk as long as the user has more than a pea for a brain. I am big on clients. I loathe webmail, and only use it when forced to.

How about ads showing what HiMD is actually best used for? Such as: * someone recording a performance of live music for their friend in a small bar [and have the friend thanking them for the recording!] * someone recording a location interview of some kind * someone recording the language of some relatively-unknown society out in the jungle * someone recording - whatever! What would I make my ads for HiMD about? Recording, recording, recording, recording, and recording.

By wasting power .. one way is to first convert the DC input voltage to AC .. then you can step it up using a transformer. There are other ways, but this is usually the simplest. Oops, forgot the part about converting the AC back to DC again.

Phono cartridges require phono preamps. I personaly have never seen a standalone recording device with a phono preamp built-in. The preamp not only matches the impedance of the signal from the cartridge to an amp or recorder by amplifying the signal to line level, it also de-emphasises [equalises] the signal from the cartridge. Without de-emphasis, the frequency response of the signal from the cartridge is completely whacked. You did mention this in your post, but then seemed to completely forget about it immediately after. Jerry-rigging connections directly from the cartridge to the recorder simply will not work. Also, while they are more linear response-wise, my experience is that ceramic cartridges are fairly rare except in exceedingly cheap [read: absolute crap-quality] turntables [the kind that come with cartridge/stylii combos that are heavy and large enough to actually plow grooves and almost completely destroy an LP in a few plays]. I could be wrong about this, but I'll note that my opinion is based on having actually serviced turntables for a few years, and having followed audiophile magazines for a number of years; I've yet to see a single decent turntable come with a ceramic cartridge, or ceramic cartridges being sold separately for decent turntables. The only places I've ever seen ceramic cartridges commonly used were on school A/V turntables [not built for fidelity by any means] and extremely cheap home sets such as the kind that often had 8-track decks built-in [also of dubious fidelity at best]. As for the hum from turntable recordings, the mainly problem is usually electromagnetic interference with the cartridge and cables. This is why most turntables and phono preamps use a cheap version of balanced connections [which functions rather poorly compared to professional balanced connections]. In the end some of the best solutions I know of are:<ul><li> keep the turntable as far from all EMR sources as possible</li><li>build a faraday cage around the turntable, including the lid, and ground the cage</li><li>make sure the polarity of the cartridge cabling is correct [matched for both sides]</li><li>make sure the grounding wire is actually connected to your preamp</li><li>try replacing your cartridge wiring with a shielded/balanced cable, turntable cabling can be purchased from electronic retailers and is made small enough to run inside your tonearm. </li></ul> And lastly: <ul><li>buy a better cartridge/stylus, and turntable, one with proper balanced wiring out to a preamp or with a built-in preamp</li><li>buy a better preamp [noting that the preamps built into even decent home A/V amps are often absolute garbage, as are the kind built into the cheap "plug & play" turntables sold these days, as are the inexpensive standalones sold at places like Radioshack]</li></ul> I somehow missed this on the first pass. My apologies, 1kyle. I'll add, though: the recording outputs of a preamp/switcher/home AV amp almost always bypass any built-in EQ or tone control.

The simplest way I've ever found to remove narrowband resonances is with a narrowband filter. Use a 4-band parametric or paragraphic EQ with a very high Q value; set all 4 filters to maximum attenuation and to centre at 60Hz. If necessary, run the same filter again. It may also be necessary to filter harmonics of 60Hz, i.e. 120, 240, 480Hz. Pattern-based noise reduction algorithms can work with material like this, but I personally prefer to avoid them because of the audible artifacting they cause. If the noise you need to get rid of is a single resonance or specific range of harmonics, narrrowband notch filters almost always work better in the end.

Straight for the purveyors of the USB standard. Semantically speaking, this would imply that we're both incorrect, and also both correct. While USB 1.1 may indeed be deprecated, and USB 2.0 may have backward compatibility, the stated spec of USB 1.1, which is mistakable as nothing else, is what I said. As indicated above, again semantically, we're both using the wrong nomenclature in describing USB at all. Funny how that works. I tend to also [incorrectly] equate 2.0 with High Speed, simply because the 480Mbps rate was the reason for ratifying a new version [and version number]. But yes. MD80 read at something like just over 1Mbps .. they write at between 400-600kbps .. HiMD writes at around 4.8Mbps and reads at about 9 peak. Real world [measured] performance though has PCM audio uploading from the unit at about 6Mbps, average.

Well, to begin with, USB 1.1 = 12Mbps, USB 2.0 = 480Mbps. Secondly, since the highest officially stated spec for the read speed of Hi-MD is between 9-10Mbps [a physical limitation of the DWDD format itself] with a write speed between .5-.6x that .. it seems pretty unlikely that HiMD is going to suddenly jump from a stated maximum speed below even that of USB 1.1 to anywhere even remotely near 480Mbps. HiMD currently already uses USB 1.1.

Mention of the clock has my interest piqued. Damn, I wish I had a tax return this year.

As much as many wil disparage behringer products, there's nothing really that wrong with them. [i'd take behringer over peavey any day, actually] Whether it will work for you really depends on how you're going to use it. Most small mic preamps like those from behringer have either XLR or 1/4" inputs .. if you want to use miniature mics like many MD/HiMD recordists do, those preamps will likely lack any inputs that will work with them [requiring 1-10V bias power for most mini electrets, for instance]. Basically, match your materiel to each other, i.e. decide what mic you want, get and use a preamp that is appropriate for that/those mics, and get appropriate cabling to run from the preamp to your recorder. And on the performance side - even a $100 behringer mini 2-channel mixer has better specs now than quite expensive equipment had 10-15 years ago. Unless you're recording extremely quiet sources [which tend to be problematic with virtually any equipment] the performance of even rather modest preamps is likely to exceed your expectations, regardless of whether they'd satisfy purists or not. Why is it that everyone always jumps for the SM57? Honestly? I've been using them since I was a child, and have ALWAYS considered them to be one of the worst-sounding microphones ever created. I spent all last week having to convince my friend who's setting up a home studio that stage vocal mics are basically inappropriate for about 90% of the jobs people try to use them for [unless of course they're all you have]. If you want a general-purpose mic, i.e. one that you want not only to do vocal recordings with, but also to do ambient recordings, close-mic instruments, &c. - then get a general-purpose mic. Buying and using a stage vocal microphone means that every single recording you make will be severely coloured by the response of the mic. SM57s for example [which are anything but "generic mics", they are quite specifically vocal mics] have a response plot that looks like someone wanted to make the most painful range in human hearing [3.5-7kHz] as loud as possible with no regard whatsoever for things like low or high end. There are many rather-inexpensive condensor mics available which would likely suit your needs. There are also dynamics which are likely to be slightly more expensive if they have sensitivities comparable to the condensors in that range. On the other hand, you can also do what most of us here do and buy a pair of relatively inexpensive [$30-80USD] electret condensors for portable recording. Take a look around some musicians' or recordists' sites for ideas about mics. Some may be out of your range or unavailable where you are [a problem here in rural Canada, I can tell you] but doing a bit of research will likely help you make a more informed decision. Suggestions: http://www.dpamicrophones.com/ [there's a link there for "microphone university" with some useful info] http://www.soundprofessionals.com/ .. an example of a commercial company that makes/sells microphones http://www.m-audio.com/ .. another company that makes less-expensive home-studio-ish equipment including mics http://en.wikipedia.org/wiki/Microphone for reference

1kyle - I don't know where you get your info from, but from what I can recall [and look up on multiple sources right now] - MP3 was designed [proprietarily, by a corporation's research division] for digital broadcast use [i.e. radio and TV]. It was not open source or public domain, and much of the most significant research into psychoacoustically adaptive data reduction methods were done by corporate interests. Universities and other academia were involved in the same [and have been for a long time] but MP3 itself came out of the effort to produce a usable relatively low-bitrate reduction algorithm for digital broadcast - something that has and had nothing to do with modems [and I'll note that at the time that MP3 and ATRAC were both being developed, 56kbps modems as we know them now didn't exist yet; most people were maxing out at 14.4kbps or less back then]. Further, while people still seem to assert that MP3 and ATRAC were developed with different aims, I still think they're all wrong - both were developed specifically for audio's "full" bandwidth [20Hz-20kHz], i.e. they are both general-purpose codecs; MP3 is also tunable to a certain extent, and can actually have superior transient response to ATRAC's. If anything, MP3, since it was designed specifically for lower bitrate encoding than ATRAC, and as a general-purpose codec, is probably more versatile. My point? That MP3 was designed to encode audio, and that ATRAC was designed to encode audio; they don't do things exactly the same way, and in some specific ways each is more efficient than the other, so really - this difference, given intentionally good encoding conditions [i.e. using close to identical output bitrates of both], is precious little. Anyway. Enough niggling points on that. Part of the problem with any discrete transform coding method [fft, dct, &c.] is that the resolution of the time-based data [samples] to be analysed determines the highest resolution of the frequency-based data it can produce [part of the reason for packet-overlap, too]. This actually makes a good argument for using high-resolution audio. 16-bit, 44.1kHz stereo audio limits how finely [timewise] you can convert to packets in the frequency domain; using higher sampling rates means potentially increasing the size of the packets to be analysed [using the same length of time per packet] or using the same packet length for much shorter periods of actual time, either of which results in higher transform resolution/accuracy. There's a tradeoff involved in the accuracy/packet vs. packet length; longer packets of PCM to be analysed means higher transform accuracy, but longer packets also mean poorer transient response [among other things]. This is one of the complains about ATRAC/3/plus, that it's packet length is so long that transient response is poor [causing pre-echo, &c.]. Until the source data [PCM for example] moves to high-res [read: specifically, higher sampling rates], everyone will be limited by what's in use. You can't pull extra resolution out of data that simply isn't there to begin with. Anyway.

This is a crock of poop. SS uses your system codec, for one, so it's not SS that is influencing playback or conversion, it's simply whatever you have installed. Also, while encoders can be optimised to give dramatically different results [from one to the next], decoders should basically always decode "identically" [unless they're severely broken, which is actually rare, and within the limits of decoders supporting different bit-depths and dither algorithms]. Actually, it's a high-shelf filter [a high pass passes only highs, exactly the opposite of what you meant].

Yes, if I had the money, I'd buy a 3rd-gen unit. And to add to. the side-debate on photography: The only reason I see left for using film is long exposure photography [night photography, for example]. Show me a DSLR that can properly expose for 5-10 minutes. Also, to back up 1kyle's points about film resolution, I have made 2700dpi scans of modern 400 and 800ISO 35mm film and that clearly showed the dye clouds in the film itself as being larger than the scanner's resolution. 26Mp? Maybe with 64ISO transparency and the finest prime lenses that exist. I think Kodak were stretching it a bit, there.

I noticed .. was the first thing I noticed .. they have the -12dB mark at the bottom of the scale [which on my RH10 and NH700 is about -38dB] .. very strange. The unit looks nice, though. P.S. .. hey Sony - TIMESTAMPING! TIMESTAMPING! TIMESTAMPING! TIMESTAMPING! TIMESTAMPING! TIMESTAMPING! TIMESTAMPING! Yep.