dex Otaku

See this thread: http://forums.minidisc.org/index.php?showtopic=9344&hl= While it applies to SS 3.0 and the controls are somewhat different, there are equivalents in SS 2.3. What you're looking for is the way to manually set the bitrate that gets put on the HiMD itself, which should be in those settings, in the same place where the choice between "standard mode" and not is, actually. Unfortunately I no longer use SS 2.3, so I can't give precise details to you.

Listed specs won't really be finalised until those of us who are really fussy about making sure the info is correct get the units in our hands. It's hard to do this before they're actually released, and still hard to do this after they've been released and we don't have them ourselves. Commercial sites that sell the equipment often take a great deal of time to update specs with what are actually listed in product manuals. Sony themselves still don't have full specs listed, from what I can see, which is rather telling. As far as whether gen1 or gen2 units are better, I'd say the only important differences are: * gen2 has native mp3 playback * gen2 no longer records in netMD mode [i.e. from live, realtime sources] Given another few weeks the specs on most sites online [including the minidisc.org equipment browser] will likely be updated one item at a time to reflect reality. I don't think you made a mistake in buying an NH900, indcidentally. The only reason I'd upgrade to a 2005 model is to get the OLED display on the RH10. I could personally care less about fluff features like speed control, virtual acoustic engine, and such - things I would never in a billion years even think to use, let alone actually need. I do find it annoying that we can't even get the details of what the units do until Sony get around to, say, posting PDFs of the product manuals [which are occasionally wrong].

That's what community is all about - being smartasses with each other. Er, no. Wait, I meant .. helping each other out. Why is correcting me such a big thing? By no means am I always right [read the thread that diverted into omni microphones and wind noise, for example], nor do I pretend to be [although I do often say things in a way that makes them sound definitive when they're not, or occasionally, as with the microphone thread, say things that I think are right but are just plain wrong]. I'm just as fallible as the next person, and I'd rather be corrected than to have someone walk away with my having misinformed them.

It depends on what you're encoding as well as what you're listening through and where you're listening. Regardless of what codec is being used, I find that certain material suffers less with low-bitrate encoding. This will tend to contradict the purists [i'm fussy but I'm not a purest] but I've found that jazz music, chamber music, classical soloists and quartets, and anything that is relatively sparse [in a harmonic sense] will be handled well by LP2 or, say, MP3 at 128kbps. Pure tones and silence are pretty easy to encode without too much artifacting. But wait - what about all those essential overtones made by acoustic instruments? Well, they disappear, yes. The thing is, you're far less likely to notice it in an obvious way unless there are either many instruments playing at the same time [i.e. jazz band or full orchestra] or only one instrument with a really complex timbre playing [such as solo sax or muted trumpet]. In these cases, however, what you're likely to notice is not distinctive swishing or underwater-sound type artifacting, but a general loss of timbre or obvious colouration of the instruments. Hard to encode material tends to be sonically and harmonically dense, with loud, fast transients [example: electronic drumbeats] and instruments with complex harmonic structures where the harmonics are extremely obvious [example: distorted guitars]. Other sources that are harder to encode include recordings originally made on analogue equipment that have audible tape hiss. Tape hiss drives lossy encoders mad, generally speaking. More specific examples of relatively easy to encode material: [Note that this is not to say there's no audible loss, just that the loss is not likely to be gratingly annoying.] * string quartets * small jazz improv groups * folk music with only acoustic guitar and vocals * electronic music featuring mostly pure tones * film soundtracks that lack complex ambience More specific examples of hard to encode material: * rock/metal/punk/industrial &c. that feature large amounts of distortion * anything with lots of effects such as flange, phase, and chorus * complex ambient sound * horn sections * full orchestras * techno and anything with loud hard drumbeats [some codecs are more forgiving than others, of course; ATRAC in all its incarnations has trouble with transients, though] * pipe organ [the only musical instrument to cover, audibly, the full range of human hearing and beyond] * cymbals in combination with virtually anything else The determining factor is more what I refer to as sonic density than anything else. Music or material made up of instruments or simply sounds with many complex overtones, or with many instruments playing at once, or with lots of effects [including simple reverb] will tend to push any encoder to its limits in trying to decide what the most important parts to allocate bits for are. The denser the material is, the harder it is to encode without audible artifacts. The sparser it is, the easier it is to encode without audible artifacts. LP2 rides the line of being just not quite enough for a lot of what I listen to. HiSP is probably slight overkill for portable use; I've found in the past that MP3 at 192kbps tends to be the minimum [for MP3] that is generally usable [meaning for almost all material you throw at it]. This is part of why there have been a lot of complaints about there being no available atrac3plus bitrate -between- LP2's 132kbps and HiSP's 256kbps. In terms of listening, I find that LP2's artifacting is plainly obvious when listening through pretty much any headphones/earphones. Generally speaking, head/earphones are the easiest way to detect artifacting. Over speakers is another matter entirely; LP2 would suffice for listening in the car or for background music at a party, say - where the ambient noise level of the environment assists in masking the artifacts. Generally speaking, playback through speakers in any other than a very quiet environment will be more forgiving. I have an album here that I use for torture-testing. It's Strange Free World by the Welsh band Kitchens of Distinction [1990]. The high-end on this album is almost brutally engineered; on a good system, it sounds smooth and makes for a good listen. Unfortunately, or fortunately in the case of having become one of my torture-test albums, it also glaringly demonstrates several things, including: the gross read error-rate of a CD player [it actually becomes audible]; DAC defects and problems; jitter problems; encoder limitations and/or deficiencies; amplifier and crossover problems; elusive high-frequency feedback issues for stage PAs; and a number of other things. This is one of my top reference albums for testing lossy encoders. LP2 makes a total unlistenable mess of it, as I find it also does with Radiohead, Nirvana, Smashing Pumpkins, Placebo, Pink Floyd, Ministry, Skinny Puppy, Peter Gabriel, The Cure, Oasis, Nine Inch Nails, Blur, Coldplay, and many other bands/artists I listen to on a fairly regular basis. The only material I have encoded and kept at LP2 was 1960s pop music that was originally recorded in mono, which it seems to do fairly well with. This is kind of interesting too, since LP2 is a true stereo encoding format, not joint-stereo, which means dual-mono material should be no easier to encode than true stereo material. Somehow it still does okay, though.

If you're using an optical out from your computer, check to make sure: * that the master volume and "Wave" controls [as tourist mentioned] are up all the way * that your player [the software you're using] has its volume turned all the way up * that your player isn't using any form of compression or volume-levelling that might cause an overall drop in levels Also, in the system volume control, go to the menu: Options -> Properties, and check to see if there's a separate control for the optical out listed under "Show the following volume controls". If there is, make sure it's enabled, then hit OK and make sure its level is turned up as well. Aside from this, not all MD recorders actually have recording level control when using the optical input. It would be in the 610's manual whether it does or not.

This might not be a VBR issue. There are some MP3 files that, regardless of what system codec I've had installed, and regardless of the fact that SS can -play- them on te computer, SS converts as nothing but silence. There appears to be no consistency to it whatsoever. I have had 320kbps CBR files made by the same encoder both work and not work when converting with SS. The usual way I use around this is to make a CD layout in Nero, write it to a disc image, mount the image, and then copy it using Simple Burner. This may seem like a lot of work, but in fact only takes a couple of minutes. I always do this with whole albums, too, which means SB finds the tags on CDDB without and difficulty. It doesn't maintain gapless playback though, as the source were MP3 files.

Yes, you can upload optical recordings. SS doesn't permit converting them to WAV, but HiMDRenderer does, so you can still do all-digital transfers.

Apologies and digressions - you're right about this. Auto time mark does override it.

I'm with Michael1980 on this, actually. People have commented on how much the SS / SB LP2 codec has improved between SS 2.1 and 3.0, but I don't hear a difference - it still sounds absolutely horrible to me.

Try the links again. They were improperly pasted; I repasted them from the original post they were in, and they now work from here.

Cute, but useless. Now available as an add-on for 3.0, apparently. I've been doing this since SS 2.1 - how is this a new feature? While this is improved, it looks the same as 3.0's [a huge step up from 2.x, I'll add].

Apologies - this is a consequence of skimming the post. In the future, it would help to separate different points on lines of their own [i.e. use paragraphs rather than one big block of text] for readability's sake. Still.. Aha! I should have noted the part about Win98SE before .. Win98 itself lacks support for USB mass storage devices, which is why SS installs drivers for the HiMD [both in mass-storage mode and the "Personal Audio Drivers"]. It sounds like the personal audio drivers might be missing on your system. You should also check what version of DirectX you have installed. I believe SS 3.0 requires DX 9.0c at the least. The only other advice I can really offer is for you to remove and reinstall SS, unfortunately - unless you happen to have a copy of the SS 3.0 disc handy to try and manually install just the HiMD device+audio drivers. You can try installing the personal audio drivers from your original CD, but this will downgrade your driver version and would likely cause more problems than it might solve. I'll see if I can find a link to just the drivers somewhere. As a curiosity note, have you tried waiting to plug the HiMD in -after- you've opened SS to see if it makes any difference? edit: The personal audio driver from SS 2.3 can be found here: http://www.sonydigital-link.com/DNA/common...p?l=en&v=&m=pad I am still looking for the version included with 3.0. edit: Sony's support sites are useless shite. I'm surprised that I even found the above [on Sony Europe]; the US site lacks any links to PAD at all - only links to the SS 3.0 installer. I'm still looking. edit: I should have looked closer to home first. From this post: http://forums.minidisc.org/index.php?showtopic=9058 I believe this is the download you should try [note the install instruction with it]: [Driver] version = 1.0.09.11300 path = http://qw-cx-ssd.sonypictures.com/download...mmon/driver.zip execution = setup.exe /NoAMI

Simple Burner is limited to HiMD modes [PCM, HiSP, HiLP] when writing to HiMD-formatted discs. Don't ask me why Sony did this, as it makes no sense.

First: have you tried SS's help? Second: are you in the transfers window, and do you have the HiMD selected? If you do, the right-hand side of the SS screen will have the contents of the HiMD listed.

It really depends on what you're encoding, but I find that anything below HiSP is basically unacceptable for general use. For very specific types of recordings, LP2 is OK [for music in general I find its artifacting glaringly obvious], and for mono recordings [such as voice only] even HiLP is fine. I'll note that I'm extremely picky about what is acceptable as a base-level when encoding - far more so than the average person.

There are conflicting reports on the RH10 and timestamping. The specs say one way on some sites and the other on others, and some don't say at all. As the 2nd-gen "top of the line" model it would make sense for it to inherit this from the NH1, though. Perhaps some of the RH10 owners out there could confirm or deny this for us.

What does it have to do with computation power? So yes - you have to use SS to put them on your player. Thing is, if you have a large collection of MP3s [like I do] you now no longer have to transcode them. SS just shunts the tracks to the player, albeit now wrapped in DRM, but this doesn't require any processing by your computer, and it means no further quality loss. If that isn't the entire purpose of adding native MP3 playback - not having to transcode - then I don't know what is. So yes, you still have to use SS, but there are a multitude of DAPs other than netMD / HiMD out there that also don't have drag & drop support. The only time this truly becomes a concern is if your primary goal in using MP3s with your player is to be able to also copy them to other people's computers from the same player.

Each pass of compression is lossy, so each pass loses something. This is true whether you're going from 256kbps mp3 -> 256kbps atrac3plus or even 64kbps mp3 -> 256kbps atrac3plus for that matter. Lossy compression means there's a loss for each pass, period. The higher the bitrate of a given pass, the less noticeable loss there is.

If you're recording using the line-in, then no, there's no way around this.

The reason why it caps out at 320kbps is that the standard for layer-III audio caps out at 320kbps. Anything above that is nonstandard, noncompliant, and can't be expected to play on hardware decoders that do follow the standard.

New canalphones and an indigo velvet frock coat.

HiSP - downloadable from SS, 256kbps, 2:23 on a single MD80 formatted as HiMD; SP - realtime transfers only via analogue or optical, 292kbps, 80 minutes on an MD80, no compatibility with 1GB discs. If you require backwards-compatibility, go with SP and realtime recording. If you don't, HiSP is quite decent.

TRS = Tip Ring Sleeve. This is neither inherently mono, nor stereo, nor balanced, nor unbalanced. It's just a connector with three contacts - TRS. The most common use of this connection in professional equipment is balanced mono [almost identical to bantam connectors]. The most common use of this connection in consumer equipment is unbalanced stereo [l + r + common ground]. And yes, TS plugs are commonly used in consumer equipment for unbalanced mono connections. The mic input of your recorder, as with the mic inputs on virtually all camcorders, minidiscs, stereo cassette recorders, and the like - is an unbalanced stereo TRS connection with the same basic layout as any stereo 3.5mm plug/jack. Just like I already said.

I would suggest reporting this to Sony via their online tech support forms, then. This is clearly an issue with software, whether it be SS/SB or a conflict caused by something else on your system.

If SS / SB are the only programs having difficulty with it, I'm not sure what to suggest. It's possible that the modules shared between both simply don't like whatever your drive is.