dex Otaku

Wtih MP3, it's not just a matter of having the correct number of frames. The frames must also all be the correct length so that CD's 75fps can fit inside them perfectly, which they usually aren't. You will usually still end up with padded frames at the end of the encoded track.

There are currently no means for recovering lost data from HiMD, even using backed up HMA files. If you happen to get this working.. a lot of people would appreciate it, I'm sure. Side-note, though - when connected via USB, HiMDs are powered by USB. The state of your battery should not have any effect on transfers. Or did you mean your computer was battery-powered and its battery died [in which case, disregard this.]

Yes, you can DAO CD-Rs; for tracks to actually be gapless, they still must conform to exact 75fps framelengths; otherwise you'll get padding at the end of the track which creates a glitch [gap] in the recording. CDs and MDs are split up by sectors for tracking where music [data] is. Those sectors are a fixed size; to have two of them meet [remain gapless], the first one has to be "full" in order to meet the second. The sector size limits editing resolution.

Gapless playback relies on following a certain number [the same as CD's] of frames per second [75fps]. The length of a given track has to be an exact number of frames in order to meet seemlessly with the next track. This is not so much a defect of atrac as a "standard" set by MD's several affiliations with CD. Basically, if you can make it work gapless on a CD, it should work as gapless as atrac as well. There are several threads in which advice on how to do this has been offered. I'd suggest searching the fora for "gapless".

Now imagine a rack full of these drives, and HAL sitting beside it.

USB 1.1 [what current Sony HiMDs use] = 12MBps. USB 2.0 = 480MBps [source: USB.org]

This may seem like silly thing to ask, but could you record the sound so we can hear it? A certain amount of noise is normal [and my NH700 is louder than any previous MD I've used], though. The head seeking in particular is the biggest source of noise, and if you've recorded tracks out-of-sequence on a disc [i.e. rearranged the track order on done any editing] it will have to seek out the next section when it's near the end of the one already playing. I've made recordings where the seek motor was obvious, but it's usually more like a quiet whine than anything really loud. So - if you can record the noise, we can probably tell you if it's normal or not.

NRen2k5 means bit depth in the same way that I generally do - linear bit depth, as applies only to PCM. Transform-based codecs do not have a bit depth in this sense.

My question is - who the heck is Buffalo?

A hardware codec can't be either optimal or sub-optimal; it either works or it doesn't. The constraints for the hardware codec are basically that they have to do the job in realtime. This isn't about processing power, it's simply about function. You create a dedicated LSI that does the job, period. A software codec is expected to run faster than realtime, and for most people that means as fast as they can tweak it to go, because they want to rip their CDs faster. This -is- about processing power, since the speed of the codec [perceived by most as higher efficiency, when it's actually not in many if not most cases] scales to the speed of the computer being used. The software codec can be tuned to either do the most accurate job possible, or the fastest job possible, or for that matter anywhere in between. It's likely that SS's codecs are tuned to compromise between the two.

The difference is that when transferring via USB, the SonicStage codec gets used. When copying via optical, the hardware encoder in the recorder gets used. It's possible for the SS encoder to surpass the hardware one in quality [as the software one can be upgraded easily and the hardware one can't] but the general concensus is that Sony probably optimised the software codecs for speed, not for quality. It's been sadi a number of times before, but I'll say it again: Sony - it would be nice to pick quality over speed as an option.

I've purchased from them and found them to be good to deal with. They even phoned me to correct a mistake I made in the order before it was sent out. Their prices tend to be among the lowest [actually, the lowest I've ever found online] in the Canadian market for MD/HiMD equipment, as well.

It sounds a bit like your n420D is having problems with writing data to the disc. I take it it still plays just fine?

Side issue to the mics: The ADCs on HiMDs at least don't appear to have much of an ultrasonic filter on them [i.e. cutting off response above about 20kHz]. Pipe organs can produce fairly "loud" ultrasonics; those ultrasonics create harmonic distortion at the same amplitude but in the audible domain [with a meeting point at ADC's Nyquist frequency]. If you're using mics that have decent response well into the ultrasonic you need to use a lowpass filter before the ADCs or you will get audible distortion that will also interfere with lossy compression schemes. Sometimes this can be heard as distortion or artifacting that you can't otherwise place a reason for the existence of.

Larger headphones tend to be lower in impedance and sensitivity. Larger drivers generally require more power. This doesn't mean that all headphones are impractical for MD / HiMD use, though. I have a pair of 16ohm Sennheiser HD330s [c.1995] and straight from my NH700 [only 5mW/channel] most music becomes ear-splittingly loud at only 22/30 volume. If you're serious about using your portable for full-sized headphone listening, I'd recommend using a dedicated heaphone amp [at home, at least]. The headphone amps in some older MD models are much higher in power than current models, too. Your manual will say what the rated power output of its amp is. If you don't have a manual, there's a section with downloadable PDFs for many models on minidisc.org . Having to use higher volume levels [read: higher power] will affect battery life, but this will depend on the efficiency of your 'phones [sPL at a certain power level]. It doesn't hurt to try it. It might be really quiet, in whch case you obviously need a headphone amp.

And I wish that I could talk theory without sounding adversarial, which I usually seem to. Also - my opinion in this case is not based on personal experience, as yours is. This has the effect of making my opinion questionable until I do experience the same or similar myself. Still, I'm not an audio engineer, I just research things at length because they interest me. Occasionally I can actually remember where I found my information, too. I'm not much of a scientist in that regard. But I must digress - I've wandered us away from the actual thread topic. Feel free to contact me directly if you'd like to discuss things.

Many modern CD players and home receivers don't actually use PCM - same as recorders don't have linear PCM ADCs - they use 1-bit systems that are actually quite similar to how DSD works [using variants of PWM, W = width, not PCM]. The performance of a true PCM ADC / DAC chain compared to most PWM chains is, at least theoretically, technically superiour on many grounds, especially if you're going to be doing any signal processing [such as post processing for recordings, and digital EQ/crossovers for playback]. DSD and PWM variants have advantages of their own, but in terms of recording and processing, they tend to fall short without using large racks of extremely expensive equipment for something that is, ultimately, converted to PCM in the end anyway [for sale to consumers], then containing some of the inherent nastiness of the DSD process as well. If you do a google search for DSD, you'll come up with several research papers about how bad it actually is as a format. These papers tend to have been written by AES members and the like, real audio engineers who understand what they're talking about [far better than I do, I'll add, I just consume their opinions and scientific data and spout little bits back out]. Here's the problem: how do you get a non-timebased, nonlinear compressed stream to convert directly to an analogue timebased linear stream? You don't. You have to convert the frequency-domain info contained in the lossy stream back to something linear in order to convert that to analogue for amplification and listening. You could take MP3 and convert it directly to DSD, incidentally. Thing is - "those nasties of the PCM output circuits" is a bit of a misnomer. As I said, and you can look around for proof of this [especially with consumer equipment], the DACs of most CD players and receivers [and sound cards, incidentally] now use 1-bit systems, based on variants of PWM, which are remarkably similar to how DSD works. The problems are from the inexpensive [less than 1/10th the cost of true linear PCM DACs] 1-bit DACs, not the PCM format or anything inherent to do with how PCM is converted to analogue. If we were all using actual PCM DACs, everything would probably sound a lot better. If engineers, studios, and record companies would also stop bitpushing recordings [the same as running the entire recording through a brickwall limiter] everything would also sound a lot better. Forgive me, but I refuse to put even a smidge of faith in that opinion. GIGO.. if you store PCM at 44.1kHz/stereo/16-bit, then you can't exceed that resolution on the output. You can interpolate, you can noise shape, you can smooth things over and alter the timbre from the original to something you prefer, but you can't exceed the resolution of what came in, and you can't really improve the sound beyond the original recording in any sense other than tailoring it [i.e. adding distortion] to suit your tastes. Rephrased: you can't get out more than is there in the first place. By the same token, anything, regardless of whether it's been mutilated by conversion to a different stream format, is likely to sound better in some way when played back through multi-thousand-dollar systems with very expensive precision DACs and preamps and amps. I'd be willing to be that if you took a Mark Levinson DAC [considered the reference for many years for linear PCM], and a decent CD player, and hooked it up to the same equipment, you'd either get results that were basically equal in quality or possibly even better. Mind you, that's no more than conjecture. I must admit to having a prejudice against DSD. This is based on the fact that I do recording, and DSD is difficult to manipulate, as well as the fact that DSD as a playback medium is that and that only - if you actually want to do anything with the stream, you have to convert it to something usable [i.e. PCM] first. As impressive as it might be to hear Dark Side of the Moon in DSD 5.1 [i own the disc, but have yet to see a single SACD player for sale where I live] DSD is little more to me than a method of copy-protection.

Is a variation of SBR not also used with mp3plus?

I make ambient recordings with some regularity, using an NH700. The principle differences I'm aware of between the models are on the output side. It would be helpful to get the service manuals/schematics for the other units, to verify exactly what the differences are on the input side, if any.

Remember too though - while this may be my experience, YMMV. It's one man's opinion.

I know you were teasing. The above is what loosely substitutes for a sense of humour, in my case.

Does the RH10 support date/timestamping? Thank you to both of you.

I guess I'll wait for the local futureshop to open [it's being built at the moment].

I started a new thread asking owners of the RH10 and RH910 to please scan and upload/attach the spec pages from their manuals. Perhaps soon we'll have more definitive answers to your questions. See http://forums.minidisc.org/index.php?showtopic=9418 Apparently the RH10 does not have speed control.

Anyone with an RH10 or RH910 with an English instruction manual - Could you please scan the spec/feature pages in the manual and attach them to this thread. Thank you!