dex Otaku

1. Portable PCM recording that runs off 1 AA battery and fits in your pocket 2. Relatively inexpensive removable media 3. USB direct digital transfers to your computer [even with the limitations of SS, this is worth it] 4. Better-than-expected quality mic preamps and AGC for a portable 5. Acceptable-quality lossy compression [though I wish ATRAC SP were usable in HiMD mode] That's it for me.

This depends on what you mean by "lossless." If, by lossless, you mean "the data that came in is exactly the data that comes out" then PCM is completely lossless. If you mean, however, that there is a loss incurred elsewhere in the recording chain, such as colouring imposed by cheap, low-power analogue to digital conversion, then you're dead-on correct. On the other hand, keep in mind that even if PCM mode is limited to 16-bit, and the ATRAC/3/plus modes encode >16-bit, the first stage before data reduction is still converting that analogue signal to PCM. i.e. if you're recording from analogue sources, you're depending 100% on the accuracy of the unit's ADCs; the compressed signal comes from the same source as the PCM signal. On one hand, this could indicate that your preference might be based on liking how ATRAC colours the sound with its artifacts, assuming that the encoder is fed the 16-bit stream. On the other hand, if the encoder is fed a >16-bit stream, that might explain your preference in terms of, despite ATRAC's being lossy, its actually being capable of recording, in some sense, a higher resolution signal than PCM mode is. Note that these are merely theories of mine. I have no way of testing them, since I lack real measurement equipment &c. It would be nice if Sony's engineering dept. would answer questions like this. I know of only one studio in this province that still uses analogue multitrack tape. Most have moved to hard-disc based systems or have been using ADAT [and Tascam DA-X88 systems, which are what CBC use in their studios as well as in their $500k 56-track mobile van] and the like for at least the past 7-10 years. By the same token, many of the digital studios I've seen keep equipment like 1/2" open-reel 6-track Dolby SR analogue recorders on hand to run the final master through an analogue phase just to get the "phat" sound of slight tape saturation in their recordings. Analogue equipment [thinking such as Otari 2" 24-track machines] are extremely expensive and finicky to maintain, and most of the people I've met who own multitrack units have gone digital to avoid the maintenance issues alone.

SQAM is a reasonable place to start. ATRAC in all its incarnations has known weaknesses which mainly revolve around the paramaters which make it editable [such as fixed encoding frame lengths and a fixed frame overlap length]. The most obvious result is the known problems with both ringing and pre-echo. Basically -any- source material that contains fast/hard transients should be able to trip up the ATRAC/3/plus encoders, including te infamous castanets sample and basically any electronic music with loud, hard beats. High-frequency transients should be the most obvious to cause artifacts.

All MD and HiMD recordings are sampled at [or actually, resampled to] 44.1kHz. PCM mode is 16-bit stereo only, same as CD. If you throw your tracks at SS or your HiMD via optical in using 24-bit/48kHz, it will all be resampled to 44.1kHz. In fact, all streams, even those which are 44.1kHz fs already, are resampled to 44.1kHz when using the optical in. ATRAC at the very least [sP mode] and likely all of the ATRAC/3/plus formats are capable of directly encoding >16-bit material, though I'm honestly not sure whether the SS codecs are limited to 16-bit [in which case everything would be requantised before encoding]. It is highly likely that the SS decoder, at the very least, is limited to 16-bit PCM decoding [as evidenced by copying its output using Total Recorder, and its lack of bit-depth settings regardless of what your sound system supports]. The information re: sampling rate &c. is in your manual, btw.

Actually, ATRAC is a discrete-transform encoding system, and has no bit depth in the sense that PCM does. ATRAC is capable of encoding 20+bit PCM and of decoding to 20+bit PCM. As are basically any lossy formats whose codecs have been made to natively convert >16bit PCM, and including MP3, OGG, AC3, DTS pro [a hybrid codec], and many if not most other recent codecs. Funny point, though: PCM on HiMD is 16-bit, 44.1kHz stereo. If the portables contain a 1-bit ADC [as most equipment do] then they should be capable of converting to >16-bit PCM directly, and encoding to ATRAC/3/plus formats from >16-bit PCM - meaning, in theory, that the lossy formats, albeit with compression artifacting, might actually be capable of recording with a higher dynamic range than PCM mode can.

Ah. See, the question was about recording formats. I assumed that meant for recordings, not for downloads, since they are not the same thing.

For important things - PCM For most things - HiSP is sufficient I never record anything even remotely important [including speech] with less than HiSP. The only thing I've used HiLP for is testing purposes. Were ATRAC SP usable in HiMD mode, I would probably choose it over HiSP, simply because of SP's maturity. And, were I ever to record live castanets, I'd stick with PCM.

Different masters have traditionally been made for [this is not always the case any more]: * radio * cassette * LP * CD &c. As far as differences between countries, I have heard of this but never really noted it in action. It would make sense under circumstances, as the equipment [especially for broadcasting] in different regions use different loading/EQ curves [loading being more or less dynamic range compression]. My guess is that this would be more for standards compliance than to suit the tastes of listeners in a particular region, though I may likely be wrong on this. Terms to look up if you're interested include AES/EBU and SMPTE.

Yes, actually. True. I can't say what the actual voltage drop is, but it is negligible enough for things to work properly. The only probable artifact of this [as per the discussion between A440 and I on this] is that you might lose a few dB of distortion-free SPL. [if anyone here remembers their impedance math, you could figure out what the drop is provided we know the actual resistance/impendance of the RSVC]. This would not be related to your problem, though. It still sounds more like a cable problem, though if it's caused by the RSVC it might be simply a dirty potentiometer or something, as A440 already said.

This is a built-in version of the Wave Converter that was necessary before SS 3.0. i.e. there is no actual new functionality here, it just means one fewer program to open to do exactly the same process I've been doing from the start. The original reason for my asserting that backups are basically necessary was because SS 2.x was botching tracks during upload, meaning the only safe way to get them backed up was before the upload to SS takes place. So far I have tested SS 3.0 with a couple of hundred HiLP and HiSP uploads, with an average system load [leaving lots of other progs open], and none have been lost.

There's only one real answer to this: I use whatever cabling the equipment dictates I must use.

Compensating levels for compression loss would have one effect and one effect only: increasing the audible artifacts. Okay, two effects: it would also utterly destroy the original timbre [tonal balance] of the recording. An example - One of the most common masking principles used by lossy compression algorithms is that of louder adjacent tones masking quieter ones, i.e. a tone at 1,000Hz at -6dBfs will mask out one at 1,050Hz and -12dBfs [grossly simplified, yes]. What you're suggesting is that the quieter tone be boosted so that the encoder is forced to include its information, correct? The result of this would be to: 1) destroy the timbre of the instrument that adjacent tone/harmonic comes from 2) overstress the encoder, which now has more data that it wants to encode, which then makes priority decisions almost impossible, and most likely ends up with far worse artifacting, since the encoder no whas too few bits to store the "desired" information. There is only one true way, IMO, to compensate for a lossy encoder's artifacting, and that is to increase the bitrate until the artifacting is made impossible by the actual presence of the data whose loss causes the artifacts. i.e. record uncompressed.

I'd said you're somewhere on the fence in terms of right/wrong here, bug80. Traditionally, the mastering process has had two 'layers': the first is to make the stereo master, the 2nd is to optimise a duplication master for each medium the recording will be distribuuted on. These days this is generally for CD, and most mastering engineers end up "bit pushing" their recordings to make them sound as loud as possible on a 16-bit medium. The end result is basically the same as back when they made separate distro masters for cassette and LP: each one is compressed slightly differently, and probably has a slightly different EQ curve as well. The process of bitpushing can both add to and subtract from normal artifacting, for example. On one hand, it can add distortion and artifacting caused by clipped peaks that PCM can play out much as limited analogue signals do, but most forms of discrete transforms will make only into distortion. On another hand, compression and limiting [what bitpushing really is] can both exaggerate [under certain circumstances] and lessen the effects that extremely hard transients have on discrete transform encoding [such as ringing and pre-echo]. Basically, your option [2] there is pretty much what they've always done, and what they still do, though to my knowledge they don't master things to compensate for artifacting with compression, exactly. Most of the time now they appear to just try and squeeze tha maximum volume possible out of the medium, at the expense of dynamic range.

That would be the NHF-800, with the tuner remote. Not in the slightest. USB mass-storage mode works with any Hi-MD formatted disc, whether it be 1GB or reformatted legacy MD. Please note that write/read speeds with reformatted legacy MDs are much slower than with 1GB discs According to the current FAQ, "Data Transfer Rate" is: Reformatted Legacy MD: 4.37Mbps 1GB disc: 9.83Mbps I'm assuming these are peak read speeds; write speed should be just under half of this. You will most likely find yourself running back to your Zip250 as it is probably still a great deal faster. True, but they're stilll painfully slow. USB mass storage mode works with any OS that supports it, including OS X and linux.

2.0 would upload and store as OMG, no? So shouldn't the library automatically update OMG tracks to OMA? I never ran into this, myself, so I don't know the solution. It's possible that your tracks are still sitting there in the SS library, but haven't been converted yet. You can look in the SS storage folder, there will be a subdir titled "Hi-MD" that should contain your uploaded tracks. Otherwise, sorry, but they're probably just gone.

I'll forgoe the philosophical debate on analogue vs. digital. This is well-known in the audio world as a personal preference based on perception [an inherently individual experience] and nothing more. Lossless encoding means: bits in = bits out The end stream is identical to the source stream [assuming perfect error correction and whatnot]. In the end, with this as with all recording methods whether analogue or digital, the rule is simply GIGO; Garbage In, Garbage Out. A simpler answer might be to say that FLAC sounds identical to PCM, since you're listening, bit for bit, to the same thing. The only difference is in the manner of storage.

Thanks, though I would note that PS-2 is not a common term, it is a specific product name. And yes, that should work, as it's exactly what I described back there [a P48 stereo mic preamp]. PS-2>AD-20? This seems a might redundant, considering the fact that the AD-20 has a gain-controllable mic preamp built in.

You're welcome.

Please define what "PS-2" is in these terms, since using a Playstation 2 as a mic preamp makes no sense. There's need to use external A/D though I would consider it if you have access to something of higher quality than the HiMD has built-in. Your chain should be something like: SMK-H8K ..via XLR cables to P48 [48V phantom power] stereo microphone preamp ..via line cable [whatever the preamp demands to 3.5mm TRS] to Hi-MD line-in

The unit's manual will have more information than I do, but I'd guess that the deck, as a component unit with full-sized power supply &c. would have a slight edge over the portable's ADCs. Keep in mind that if you transfer your recordings to HiMD digitally via optical line, you CAN NOT simply convert them to Wave using Sony's utilities after uploading them to SS. You can only convert optically-copied tracks using marcnet's HiMDRenderer.

See here: http://forums.minidisc.org/index.php?showtopic=7989 Cheers, and welcome to the forum.

You could sort by genre? As far as I know, SS is a single-library system.

that orange one really catches my eye.

Well, since you're basically making a stereo-compatible mono recording by doing so - none, really. You're just accepting that your recording is in mono, really.

I am saying that this might solve the problem, but I don't know as I haven't checked myself, yet. Please note that the service in question is marked as "manual" for starting, though it appears to be running after a reboot on my machine, despite no other Sony software running on boot. As long as you're not trying to burn CDs in SS, closing it shouldn't have any ill effect. Still - no, I'm not saying everyone should do this.