sfbp

Not true. Please re-read his results. I would have said your compulsive need to attack me is proof of your ego's needs. Of course I misread it, sorry for that. I was more concerned with the board chewing up my posts and spitting them out TBH. Actually that's effectively exactly what it does. By allowing different portions of the spectrum to use up more (or less) bits in any given sample, depending on the composition of the sounds making up the sample. The difference between ATRAC and VBR MP3 is that the latter is variable in the time domain; the former is variable in the frequency domain. I think (feel free to explain it better than I have; my grasp of the real maths is tangential at best - probably OP will explain it better than you or I). Let's (to use your phrase) give the ad hominem stuff a rest. Please.

That's my boy! CD->WAV->LP2 really really doesn't work. Clearly SP is the winner. But I and others already made the judgement that 80m often isn't enough and the compromise we do not object to is to get double the space on the same disk at a slight loss of quality. (and as u mention, LP2 on a HiMD formatted disk gets almost double THAT amount of space, because LP2 is already wasting about 20% of the data on disk for reasons of compatibility with machines that cannot even play it!!!). Also SP cannot go via computer, so is time-consuming to make. I completely concur that AAL is the storage of choice and that (from my limited experience) it converts well to all useful (and useable) formats. Interestingly, did you use AAL/256 or AAL 132? When I found that AAL/256 gave good LP2 output, I stopped worrying about AAL/132, on the grounds that IF i needed to generate Hi-SP (256K) this would be minimal conversion from AAL/256 (just throw away the lossless part). Stephen PS no need to ask about 24-bits, because I will go on telling about it until someone contradicts me The reason that MD sounds so good at low bit rates is PRECISELY that it is a 24-bit format. This means (I think) that very loud and very quiet sounds are properly rendered without any mathematical errors from fixed-point rounding (remember CD is a fixed point format, whereas ATRAC is a floating point format).

I agree that AAL ripping is very good indeed. What I am not so sure about is CD->WAV direct ripping.

No difference from what? You've only told the screen is blank. But I have no idea if that means the whole unit is non-functioning. I originally misread it, perhaps - does ANYTHING happen when you insert a disk? (the usual noises of disk turning) Can you play a disk?

OK, I've looked at the RH10 service manual and I am not much wiser than before. The only thing that occurs to me is that perhaps putting it into service mode may have a fail-safe action which turns the display on. Maybe some RH10 owner out there has run into this. I know my RH1 went black on one occasion and eventually revived for no obvious reason. Another thing to try is to connect up to USB and see if that wakes it up at all. Good luck! Stephen

I think you have to do "resume clear" from service mode. Let me research a little bit.

Hold "display" button for a few seconds. Presumably you also get into the menu when you hold the menu button on the unit down, also. Just you don't see the screen display, right? Oops, the menu option for Contrast does not show up when entering the menu from the remote. If you can figure out how to get into Option (use the remote to enter it and then cancel), the "Contrast" (I suspect this is your problem) is 3 clicks "north" (upwards) of the little wheel on the front of the RH10 (mine is a 910 but so what?). Then FF to increase contrast from blank screen.

Can you use the menu on a stick remote such as Rm-MC38EL?

The menu.... or Service Mode? In the latter case there are some default settings it will make if you do "resume clear" (same for most MD units).

It's free when you've paid for it. See post #11 in this thread.

That's the one I found.

Erik, if you're saying that Sony does (or did) the first when doing a normal Rip of CD->WAV with SonicStage (read fast and don't care about the error correction), I'd be inclined to believe you. If, OTOH paperclip is saying that SS now does the second way, and can prove that the rip is correct and matches EAC, then I will believe him. Could it be that when I originally got turned off to LP2 that the ripper in SS was defective and they later fixed it? If so, I'm willing to eat humble pie and publish a retraction. But I saw what I saw, and there has to be an explanation, even if we haven't (yet) found it. And others have declared themselves similarly fed up with MDLP. It's even possible, isn't it. that this was the reason that Sony "lost" the MP3/Ipud comparison wars 8-10 years back - because someone with left-brain thinking decided (just as I did, to be fair) that the "level playing field" way was to start by ripping the CD directly to WAV file and then rip to different rates of ATRAC compression. Of course that explanation only works if the testers used SS ripper to rip for the sound files being used to test to convert to ATRAC, but some other ripper for the files being converted to MP3. What I *do* recall was that originally the ripping speed was really fast. I *was* subsequently puzzled by slower ripping on a later version of SS. So it is indeed possible that this artifact only existed in 4.0 and/or 4.2 SonicStage (or versions even earlier than that), and is now gone. I think I recall one of those two, but I don't remember which. Stephen

Take a look at the options for ripping CD's in SonicStage - perhaps you will see what I mean. I've explained my logic, I don't think there's anything else I can say to convince you.

I'm not sure you are right. Otherwise why would Sony specify different speeds of ripping ("fast" vs "high-quality") for some of its codecs? That word "just" is always a red rag to this particular bull. And why should one not think that oversampling on a CD player/reader is not matched by the same thing in the digital domain?

You can try Google. At this stage in the product's lifetime, you will probably have to pay for the service manual, however, since it is relatively new. I found one for $25. Maybe you can do better......

time to look for the service manual

I did it on the 754. Why don't you follow those instructions, first? You don't explain where it goes wrong....

Before we get to name-calling, please stop and consider why as early as 1985 CD player manufacturers were advertising 20-bit oversampling......... (well, actually 16x oversampling, i.e 16 + 4 = 20 bits) My impression of the oversampling technique (and it may well apply to different techniques for ripping the music off a CD) is that you end up with proper dithering of those values which lie "between" two digitally-encodable values, and that the effect is to "anti-alias" the sounds, so there are no interfering harmonics. Musicians will tell you that all kinds of notes that are never played make up the timbre of an instrument. CD's start to sound natural (and, incidentally to compress well) when the anti-aliasing (which is incidentally used in graphics processing systems ubiquitously) is applied, and the false harmonics cancel out. As I said, my $.02 - I can't prove most of this, and I don't have a proper background in sound engineering. But I definitely know it's possible to get a perfectly good-sounding wav file which absolutely refuses to compress properly, using SonicStage as the initial ripper. I presume this is what Sony is referring to when they talk about "fast" or "high quality" ripping speeds for their other codecs. Interestingly they don't offer that option (analogous to the oversampling CD players) for Sonic Stage's WAV extraction. Stephen

I take 100% responsibility. I (like, it seems many others here) tried LP2 and LP4 very early on in the game and found the results were awful. I did so by importing a WAV from a CD, and then downconverting it. In the end I concluded it must be the import step which was at fault, for whatever reason. There's a good reason that I believe this may be a defective process. I think that CD playback, and also ripping direct to ATRAC actually manage to recover more than the nominal 16 bits on an audio CD. We have read for 25 years about oversampling which effectively gives you 18, 20 or even more bits of resolution with a good player, or an effective ripper. I totally agree that if one has GOOD wav files (such as those generated by EAC, when properly configured) then they can be downconverted quite well. But I think you can see that a 16-bit file on disk might not downconvert to a low rate 24-bit format (ATRAC) as well as direct ripping to that same format. YMMV.

I agree 100% that this noise sounds like an artifact from the PC card. Sorry for doubting you, but as I never had such a problem, I was attempting to imagine some other simple explanation. You could try a much simpler and cheaper card. I use a. Terratec Aureon Fun 5.1 b. Sony PCLK-MN10. Both seem to give really clean optical out with no constant noise like you are seeing. I'm wondering about some other INPUT to your sound card that needs to be turned completely OFF. Eg microphone input....?

I would try Simple Burner, for a start. And remember to use Type-S playback I also initially rejected LP2 but that was because I went about it the wrong way. I ripped to WAV and then converted. This tends to produce garbage. Sorry I wasn't clear. I mean that lp2 is the most compatible playback format. But I rip to AAL on the hard disk, which if need be can be converted to LP2 or any other format. The PC never plays back the lossless part. The only point of it is for when you want to convert to some other format, or maybe edit it (I am not certain, but I assume Sound Forge WILL use the lossless part to rebuild the internal wave when you edit one of these files). It's a standalone program from Sony, though I have a strong suspicion SonicStage needs to be installed before you run it. Once it is run, of course the MP3 files are totally standard and can be used anywhere. The real "secret" of ATRAC is that it is in many ways BETTER than standard 16bit 44.1Khz PCM (CD format) because it actually has 24-bit resolution on all samples, even at the lowest data rate. It does this by making the magnitude a floating (instead of fixed) point number. This will tend to avoid many of the weird artifacts in other formats (have you ever tried converting MP3 back to ATRAC?).

just to inject my $0.02 worth of advice, which you can easily disregard. The most compatible format of all is LP2. This works on all NetMD players, and can be used on HiMD (you get 15 or 16 hours on a single HiMD 1GB disk). It is still 24-bit ATRAC so you get most of the benefits of SP with the additional enhancements of Type-S on all HiMD units (and some NetMD units) for playback, which IMHO improves the sound so much you really don't notice a lot of difference (yes there is SOME difference) from ordinary SP. Except for LIVE recordings I would choose this format. For HiMD conversion from CD's I typically convert to AAL 256kbps) which can convert to anything with little degradation, but usually for simplicity I simply transfer to HiMD - without further bitrate converting. This leaves the :lossless: part behind (on the PC) and gives you ordinary 256kbps (Hi-SP) on the MD. All the formats are 24-bit, except when you convert to WAV on the PC from one of the atrac formats. For SP, this conversion is OK. For the other formats, I feel it tends to be a bit noticeable in the final result. Finally, Sony has somehow figured out MP3 conversion (with their MP3CONVERSION.EXE tool) that as long as you always choose 256kbps for the MP3, the results are quite decent. I just converted a bunch of files for my dad, and was pleasantly surprised how good the MP3 results were. Most MP3s however are not nearly that good. The key is to keep a high bitrate, even for 66 and 132 kbps ATRAC sources.

I just had EXACTLY that problem; RCA into 3.5mm jack you are at the mercy of whoever built the cable. Sony does some tricks on its car headunits when connected to satellite changer(s) by not using the ground on the audio lead. Something about the double ground at the 2 RCA jacks causing a ground loop....?

WhenEVER I see the word "just" i rise about 3 inches off the ground, floating, in anticipation of a knockout punch But I think you already answered your own question (Philippe is right), there is no coax in for an MD portable (some decks have them). So unless you have the right OPTICAL cable, you won't get a very good sound on the analog output into MD - mainly because the 3.5mm jack will pick up lots and lots of noise, especially from a PC. I am wondering if you tried to connect coax out directly to the analog jack (same as optical jack) socket of the MD........?

what exactly do you mean by "PCM encoded music"? All digital sound is PCM encoded in some form. If you mean 1411kbps (CD standard quality) 44.1Khz "LPCM" then you are right, something is off on your system. But you do not need NetMD driver for anything uploadable. In fact NetMD can never upload anything, it simply isn't designed to. Your setup SHOULD be able to transfer HiMD tracks without difficulty. But you canNOT upload standard ("legacy") MD tracks, at 292, 132, or 66 kbps tracks to a computer using the NH900. What bit rate does Sonic Stage show you? (you may have to turn the display of bitrate on, try rightclicking the column heads).